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Extended Fourier analysis of signals
Abstract. The extended summary of Dr.Sc.Comp. thesis [6] is created to emphasis the tight
connection of the proposed spectral analysis method with the Discrete Fourier Transform
(DFT ! the most extensively studied and fre"uently used approach in the history of signal
processing. #t is shown that in a typical application case$ where uniform data readings are
transformed to the same num%er of uniformly spaced fre"uencies$ the results of the classical
DFT and proposed approach coincide. The difference in performance appears when the length
of the DFT is selected greater than the length of the data. The DFT solves the un&nown data
pro%lem %y padding readings with 'eros up to the length of the DFT$ while the proposed
(xtended DFT ((DFT deals with this situation in a different way$ it uses the Fourier integral
transform as a target and optimi'es the transform %asis in the extended fre"uency range
without putting such restrictions on the time domain. Thus$ the #nverse DFT (#DFT applied to
the result of (DFT returns not only &nown readings %ut also the extrapolated data$ where
classical DFT is a%le to give %ac& )ust 'eros. The (DFT significantly extends the usa%ility of
the DFT %ased methods$ where previously these approaches were considered inapplica%le [8!32].
The (DFT founds the solution in an iterative way and re"uires repeated calculations to get the
adaptive %asis$ and this ma&es its numerical complexity much higher compared to DFT. This
disadvantage was a serious pro%lem in *++,s$ when the method has %een proposed.
Fortunately$ since then the power of computers has increased so much that nowadays (DFT
application could %e considered as a real alternative.
Table of Contents
(xtended Fourier analysis of signals...............................................................................*
* #ntroduction......................................................................................................-
- .ro%lem formulation........................................................................................-
-.* /asic expressions of classical Fourier analysis.................................-
-.- /asic expressions of extended Fourier analysis................................0
0 .ro%lem solution..............................................................................................1
0.* (xtended Fourier transform of continuous time signals...................1
0.- (xtended Discrete Time Fourier Transform......................................2
0.-.* 3 particular solution for discrete time signals....................2
0.-.- 4enerali'ed solution for discrete time signals...................5
0.0.0 #terative (DTFT algorithm.................................................6
7 (xtended DFT algorithm..................................................................................6
1 (DFT and other nonparametric approaches...................................................*,
1.* Capon filter approach......................................................................*,
1.- 489S solution................................................................................*-
1.0 :igh!;esolution DFT......................................................................*0
2 Computer simulations....................................................................................*0
5 (DFT algorithm in <3T93/ code...............................................................-,
6 ;eferences......................................................................................................-2
Extended summary of Dr.Sc.Comp. thesis 1
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
1 Introduction
3 Fourier transform is a powerful tool of signal analysis and representation of a real or complex!
valued function of time x(t (hereinafter referred to as the signal in the fre"uency domain
}
= dt e t x F
t je
e ( ( $ (*.*
e e
t
e
d
e
F = t x
t j
(
-
*
(
}
. (*.-
The Fourier transforms orthogonality property
( -
,
,
e e to
e e
=
}
dt e e
t j t j
(-
providing a %asis for the signal selective fre"uency analysis$ where e, e
,
are cyclic fre"uencies
and o(ee
,
is the Dirac delta function. =nfortunately$ the Fourier transforms calculation
according to (*.* re"uiring &nowledge of the signal x(t as well as performing of integration
operation in infinite time interval. Therefore$ for practical evaluation of (*.* numerically$ the
signal o%servation period and the interval of integration is always limited %y some finite value O,
-O>-?t?O>-. The same applies to the Fourier analysis of the signal x(t sampled versions:
nonuniformly sampled signal x(t
and x(!$ are derived from the %and!limited in O signal x(t. 9et write the %asic expressions of
the classical and the proposed extended Fourier analysis of continuous time signal x(t and its
sampled versions x(t
and x(!.
2 Problem formulation
E!he formulation of a pro&lem is often more essential than its solution 'hich may &e merely a
matter of mathematical or experimental sill.( Albert Einstein
2.1 Basic expressions of classical Fourier analysis
The classical Fourier analysis dealing with the following finite time Fourier transforms
}
O
O
O
=
- >
- >
( ( dt e t x F
t je
e $ (0.*a
O
*
,
( (
"
t j
e t x = F
e
e
$ (0.*%
O
*
,
( (
"
! j
e ! x = F
e
e
$ (0.*c
e e
t
e
d
e
F = t x
t j
(
-
*
(
O
O
O
O
}
. (0.-
where (0.- is the inverse Fourier transform o%tained from (*.- for %and!limited in O signal.
Transforms (0.*% and (0.*c are &nown as Discrete Time Fourier Transforms (DTFT of
Extended summary of Dr.Sc.Comp. thesis 2
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
nonuniformly and uniformly sampled signals. The values of reconstructed signal x
O
(t outside
the o%servation period O are 'eros or vanishes depending on whether (0.- applies to the results
(0.*a or (0.*% and (0.*c.
The signal amplitude spectrum is the Fourier transform (0.* divided %y the o%servation
period O$
(
*
( e e
O O
O
F = S . (7
The fre"uency resolution of the classical Fourier analysis is inversely proportional to the signal
o%servation period O.
C%viously$ one can get the formula (0.*a %y truncation of infinite integration limits in (*.* and
the DTFT (0.*a and (0.*% as result of replacement of infinite sums %y finite ones. This mean$
the classical Fourier analysis supposed that the signal outside O is 'eros. #n other words$ the
Fourier transform calculation %y formulas (0.* is well )ustified if applied to time!limited within
O signals. Cn the other hand$ a %and!limited in O signal cannot %e also time!limited and
o%viously have non'ero values outside O. 4enerally$ the Fourier analysis results o%tained %y
using the exponential %asis tend to the Fourier transform$ if O, while in any finite O there
may exist another transform %asis providing a more accurate estimation of (*.*.
2.2 Basic expressions of extended Fourier analysis
The idea of extended Fourier analysis is finding the transform %asis$ applica%le for a %and!limited
signals registered in finite time interval O and providing the results as close as possi%le to the
Fourier transform (*.* defined in infinite time interval. The formulas for proposed extended
Fourier analysis could %e written as
dt t t x = F $ ( ( (
- >
- >
e o e
o
}
O
O
$ (1.*a
=
*
,
$ ( ( (
"
t t x = F e o e
o
$ (1.*%
=
*
,
$ ( ( (
"
! ! x = F e o e
o
$ (1.*c
e e
t
e
o o
d e F = t x
t j
(
-
*
(
}
O
O
$ (1.-
where in general case the transform %asis o(e$t$ o(e$t
}
S dt
e
t x
t j
. (6
Extended summary of Dr.Sc.Comp. thesis 3
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
Dow$ letFs use (5 as a signal model with &nown amplitude spectrum S(e
,
for fre"uencies in
range -O?e
,
?O and$ in the minimum least s"uare expression (2$ su%stitute F(e %y the signal
model Fourier transform (6 and the signals x(t$ x(t
}
=
O
O
A
*
,
,
-
, , ,
$ ( ( ( ( -
,
"
t j
d t e S S =
e e o e e e o e t
e
$ (+%
}
=
O
O
A
*
,
,
-
, , ,
$ ( ( ( ( -
,
"
! j
d ! e S S = e e o e e e o e t
e
. (+c
The solutions of (+ for a definite signal model (5 provide the %asis o(e$t$ o(e$t
and o(e$!
for the extended Fourier transforms (1.*. To control how close the selected signal model
amplitudes S(e
,
are to the signals x(t$ x(t
and o(e$!.
The formula (6 is showing the connection %etween the signal model Fourier transform and its
amplitude spectrum$ from where S(e
,
could %e expressed as signal model Fourier transform
divided %y -to(ee
,
. Ta&ing (6 into account$ S
*
(e is calculated as the transforms (1.*
divided %y the estimate of -to(ee
,
in the extended Fourier %asis$ which is determined from
(+ in the case A=0 and e
,
=e$
dt t
e
dt t t x
= S
t j
$ (
$ ( (
(
- >
- >
- >
- >
e o
e o
e
e
o
}
}
O
O
O
O
$ (*,a
=
*
,
*
,
$ (
$ ( (
(
"
t j
"
t e
t t x
= S
e o
e o
e
e
o
$ (*,%
=
*
,
*
,
$ (
$ ( (
(
"
! j
"
! e
! ! x
= S
e o
e o
e
e
o
$ (*,c
and showing that the amplitude spectrum on the fre"uency e is estimated as ratio of the signal
extended Fourier transform to the transform of exponent with a unit amplitude in the same %asis.
This is true also for classical Fourier transform. For example$ after su%stituting exponential %asis
t j
e = t
e
e o
$ ( in (*,a$ the denominator %ecomes e"ual to O as in formula (7 for the classical
Fourier analysis.
Galues of the denominator in formulas (*, are in inverse ratio to the fre"uency resolution of the
extended Fourier transform.
/efore finding the the extended %asis functions for ar%itrary S(e
,
$ it is reasona%le to consider a
simple signal model having a rectangular form$ S(e
,
@* for -O?e
,
?O and 'eros outside. Then
the estimators (+ reduces to
Extended summary of Dr.Sc.Comp. thesis 4
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
,
- >
- >
-
,
( ( -
,
e e o e e to
e
d dt t )
e
=
t j
} }
O
O
O
O
A
$ (**a
}
=
O
O
A
*
,
,
-
,
$ ( ( -
,
"
t j
#
d t e =
e e o e e to
e
$ (**%
}
=
O
O
A
*
,
,
-
,
$ ( ( -
,
"
! j
#
d ! e = e e o e e to
e
. (**c
The solution of (** allows to esta%lish relationship %etween the classical and extended Fourier
analysis.
Problem solution
#n this section the integral least s"uare error estimators (+ and (** are solved and su%se"uent
analysis of the o%tained results are performed to find out the only those solutions that can lead to
practically reali'a%le algorithms.
.1 Extended Fourier transform of continuous time signals
The solution of (**a for continuous time signal x(t is found as a partial derivation
- - $ ,
$ (
+ + = O s s O
c
A c
t
t e o
$ and leads to the linear integral e"uation
( )
et
e o
t t
t
j
e dt t )
t
t
O
O
}
=
O
- >
- >
(
(
( sin
. (*-
Step %y step solution of (*- is given in [2]. Finally$ the %asis o(e$t are o%tained %y applying a
specific functions system ! a prolate spheroidal wave functions
=
, @
(
(
$ (
t
,
t
e
e o
. (*0
The extended Fourier Transform of continuous time signal x(t are given %y
O s s O =
e e e
o
! $ ( (
, @
a , F
$ (*7.*
< < =
t a t t x
! $ ( (
, @
o
$ (*7.-
-
,
,
(
(
(
e
e
e
o
,
a
,
S
=
=
$ (*7.0
where
t t t
d x =
a
+
+
( (
*
-
-
}
O
O
$
dt t =
+
+
(
-
-
-
}
O
O
and
j =
,
(
-
(
|
.
|
\
|
O
O
O
O
e
t
e .
The extended Fourier transform in accordance with (*7.* re"uesting a calculations of infinite
sums$ this mean$ an infinite "uantity of mathematical operations$ therefore itFs impossi%le for real
world applications. Theoretically$ the value of denominator
-
,
(e
,
"
=
in amplitude spectrum
formula (*7.0 tends to infinite for "$ and the extended Fourier transform (*7.* provide a
supper!resolution ! an a%ility to determine the Fourier transform for the sum of sinusoids or
Extended summary of Dr.Sc.Comp. thesis 5
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
complex exponents$ if fre"uencies of them differ %y ar%itrary small finite value.
.2 Extended !iscrete Time Fourier Transform
#n this su%section the minimum least s"uare error estimators (+%$c and (**%$c are solved and the
extended Fourier transforms for uniformly and nonuniformly sampled complex!valued signals
are o%tained. The proposed approaches have %een developed in articles [3] and [4]$ where the
derivations for real!valued discrete signals are given.
.lease note that the following notations are used in the matrix e"uations:
superscripts X
!*
$X
!
$X
H
$X
-
denote inverse$ transpose$ complex con)ugate$ :ermitian
(complex con)ugate transpose of the matrix XI
.> represents element!%y!element division of two matrices with the same si'eI
sum(X means addition of all matrix X elementsI
dia.(X forms the row vector %y extracting the main diagonal elements from "uadratic
matrix X or it puts the elements of vector X on the main diagonal to form a diagonal
matrix.
.2.1 A particular solution for discrete time signals
The solutions of (**%$c can %e o%tained similarly to (**a as partial derivatives of
,
$ (
=
t
l
e o c
A c
and
,
$ (
=
l! e o c
A c
$ l@,$*$-$..."!*$ and leads to the systems of linear e"uations
( )
l
t j
"
l
l
e t )
t t
t t
e
e o
t
=
=
*
,
(
(
( sin
$ (*1.*
( )
l! j
"
e ! )
! l
! l
e
e o
t
=
=
*
,
(
(
( sin
. (*1.-
The solution of (*1 in the matrix form is expressed as
e e
E R A
*
= $ (*2
where A
e
("x* and E
e
("x* are the extended Fourier and the exponential %asis.
The formulas of (xtended Discrete Time Fourier Transform ((DTFT for signal model S(e
,
@*$
-O?e
,
?O, are derived %y su%stituting of transform %asis (*2 into expressions (1 and (*,
$ $ (
*
O s s O =
e e
e o
E xR F (*5.*
$ $ (
*
< < =
t t x
t
E xR
o
(*5.-
e e
e
o
e
E R E
E xR
*
*
(
-
= S
. (*5.0
The matrices for nonuniformly sampled signal case are composed as follows
x (*x" !
(
t x
$ E
e
("x* !
l
t j
e
e
$ R ("x" !
(
( sin
$
l
l
l
t t
t t
r
O
=
t
and E
t
("x* !
(
( sin
l
l
t t
t t
O
t
.
=niformly sampled se"uence x(! can %e considered as a special case of nonuniform
sampling at time moments t
O
=
t
$ E
t
("x* !
(
( sin
l! t
l! t
O
t
.
#n particular$ if sampling of signal x(! is done with Dy"uist rate$ !=t/O$ the matrix R
%ecomes a unit matrix I and the formula (*5.* coincide with classical DTFT (0.*c$ %ut the
formula (*5.0 reduces to well &nown relationship %etween discrete signal Fourier transform and
Extended summary of Dr.Sc.Comp. thesis 6
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
its amplitude spectrum
e o
e e xE = =
O
( ( F F
$ (*6.*
e o
e xE
"
S
*
( = . (*6.-
8hereas for nonuniformly sampled signal x(t
e e
E R E
*
and the fre"uency resolution should increase proportionally to the num%er of
samples in the signal o%servation period O. #n the %order!case$ if num%er of samples within O
increasing infinitely$ "$ and the discrete time signal tends to the continuous time signal x(t$
the (DTFT (*5.* gives the same results as (*7.*.
.2.2 "enerali#ed solution for discrete time signals
Dow$ let consider the solution of the minimum least s"uare error estimators (+%$c for ar%itrary
selected signal model S(e
,
. The derivation formulas for %oth estimators are similar to ones
given in previous section. For example$ a partial derivation of (+% %y %asis functions
$ * $...$ - $ * $ , for $ ,
$ (
=
c
A c
" l =
t
l
e o
provide the least s"uare solution
, ( $ ( ( ( ( -
, ,
H
*
,
, , ,
, ,
= |
.
|
\
|
=
O
O
}
e e e o e e e o e t
e e
d e S t e S S
l
t j
"
t j
$ (*+
("uation (*+ can %e rewritten as
, ,
-
,
*
,
,
(
-
,
( ( - $ ( (
, ,
e e e o e t e o e e
e e
d e S t d e S
l l
t j
"
t t j
=
|
|
.
|
\
|
O
O
O
O
}
}
. (-,
The filtering feature of Dirac delta function ( ( (
, ,
x f dx x x x f =
}
o applied to the right part
of (-, gives the final form of the system of linear e"uations
l l
t j
"
t t j
e S t d e S
e e
e e o e e
t
O
O
=
|
|
.
|
\
|
}
-
*
,
,
(
-
,
( $ ( (
-
*
,
$ (-*.*
l! j
"
! l j
e S ! d e S
e e
e e o e e
t
O
O
=
|
|
.
|
\
|
}
-
*
,
,
(
-
,
( $ ( (
-
*
,
$ (-*.-
for l@,$*$-$...$"!*$ where
-
(e S is the signal model power at e
,
=e. The e"uations (-*.- are
applica%le for uniformly sampled signal x(! and can %e derived from (+c in a similar way as
(-*.*. The (DTFT %asis A
e
("x* ! o(e$t
t = t x
t
E xR
o
(-0.-
Extended summary of Dr.Sc.Comp. thesis 7
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
e e
e
e e
e
e e
e
o
e
e
e
E R E
E xR
E R E
E R x
A E
xA
*
*
*
-
*
-
(
(
(
= =
-
-
-
S
S
= S
(-0.0
The elements of matrix R ("x" in the formulas (--$ -0 are expressed %y integrals
}
O
O
=
,
(
-
, $
,
(
-
*
e e
t
e
d e S r
l
t t j
l
$ (-7.*
}
O
O
=
,
(
-
, $
,
(
-
*
e e
t
e
d e S r
! l j
l
$ (-7.-
for nonuniformly and uniformly sampled signal x (*x" cases$ correspondingly. #f the signal and
its model power spectra are close$
- -
( ( e e
o
S S ~ $ then the matrix R elements (-7 are also an
estimate of the autocorrelation function for the se"uence x. Similarly$ the elements of matrix E
t
("x* in (-0.- ac"uire integral form
}
O
O
= e e
t
e
d e S e
l
t t j
l
(
-
(
-
*
or
}
O
O
= e e
t
e
d e S e
l! t j
l
(
-
(
-
*
.
The inverse transform (-0.- calculated on time moments t=t
or t=!$ @,$*$-$A$"!*$ returns
%ac& the input se"uence x undistorted. Case signal model S(e
,
@* the formulas (-- and (-0
reduces to (*2 and (*5.
The fre"uency resolution of the (DTFT is in inverse ration to
e e
e E R E
*
-
(
-
S and varied in
the fre"uency range -O?e?O.
.. Iterati$e E!TFT algorit%m
Calculation of the (DTFT %y formulas (-0 re"uires &nowledge of the signal model spectrum
which generally is not &nown. 3t the same time$ the amplitude spectrum o%tained in the previous
section %y the formula (*5.0 can %e used as a source of such information. This suggests the
following iterative algorithm$ where the signal model spectrum S(e
,
tends to the signal
spectrum S
*
(e:
/teration $. Calculate (
* (
e
a
S (*5.0 applying default signal model S(e
,
@*.
/teration 0. Calculate (
- (
e
a
S (-0.0 %y using the signal model (
,
* (
e
a
S .
/teration 1. Calculate (
0 (
e
a
S (-0.0 %y using the signal model (
,
- (
e
a
S .
2
/teration i. Calculate (
(
e
i
a
S (-0.0 %y using the signal model (
,
* (
e
i
a
S .
The iterations are repeated until the given maximum iteration num%er is reached or the power
spectrum do not alter from iteration to iteration$
-
* (
-
(
( ( e e
~
i
a
i
a
S S .
The (DTFT output F
*
(e (-0.* is calculated for the last performed iteration /.
/y default the signal model S(e
,
@* is used as input of the (DTFT algorithm. :owever$
additional information a%out the signal to %e analy'ed can %e applied to create a more realistic
signal model for the (DTFT input and to reduce the num%er of iterations re"uired to reach the
stopping iteration criteria.
& Extended !FT algorit%m
The (DTFT considered in the previous section is a function of continuous fre"uency(-O?e?O$
while descri%ed %elow the (DFT algorithm calculate the (DTFT on a discrete fre"uency set
-Ose
n
<O for n@,$*$-$A$3!*. The num%er of fre"uency points 3>" and it should %e selected
sufficiently great to su%stitute the integrals (-7 used for calculation of matrix R ("x" in the
Extended summary of Dr.Sc.Comp. thesis 8
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
expressions (--$ -0 %y the finite sums
}
=
O
O
O
~ =
*
,
(
-
,
(
-
, $
( (
-
*
,
3
n
t t j
n
t t j
l
l n l
e S
3
d e S r
e e
e
t
e e
t
$ (-1.*
}
=
O
O
O
~ =
*
,
(
-
,
(
-
, $
( (
-
*
,
3
n
! l j
n
! l j
l
n
e S
3
d e S r
e e
e
t
e e
t
$ (-1.-
l$@,$*$-$A$"!*. The matrix composed of (-1.* and (-1.-$
(
(
(
(
(
(
=
, ( ... ( ( (
... ... ... ... ...
( ... , ( ( (
( ... ( , ( (
( ... ( ( , (
* $ * * - - $ * * * * $ * * , , $ *
- * * $ - - $ - - * * $ - - , , $ -
* * * $ * * - - $ * * $ * * , , $ *
, * * $ , , - - $ , , * * $ , , $ ,
" " " " " " " "
" "
" "
" "
r t t r t t r t t r
t t r r t t r t t r
t t r t t r r t t r
t t r t t r t t r r
R $ (-2.*
( )
( )
( )
( ) ( ) ( )
(
(
(
(
(
(
, ( ... 0 ( - ( * (
... ... ... ... ...
0 ( ... , ( ( - (
- ( ... ( , ( (
* ( ... - ( ( , (
* $ * - $ * * $ * , $ *
* $ - - $ - * $ - , $ -
* $ * - $ * * $ * , $ *
* $ , - $ , * $ , , $ ,
" " " " "
"
"
"
r ! " r ! " r ! " r
! " r r ! r ! r
! " r ! r r ! r
! " r ! r ! r r
R
$ (-2.-
possesses :ermitian symmetry$
H
$ $ l l
r r =
$ %ut (-2.- for uniformly sampled signal has also a
Toeplit' structure. The matrix elements r
l)
representing the autocorrelation function of the
selected signal model and can %e calculated %y applying the #DFT to the signal model power
spectrum
-
(
n
S e . The fre"uency O/t=2f
u
=f
3
in (-1$ where f
u
is the signal upper fre"uency and
f
3
is the Dy"uist rate of a %and!limited signal$ and it is assumed to %e normali'ed (e"ual to * in
DFT calculations. The choice of the fre"uencies Ke
n
L@K-tf
n
L depends on the num%er of
fre"uencies needed for accurate estimation of (-1 as well as for detailed signal spectrum
representation$ and the limitations on the total amount of computation. (ventually$ the uniform
set of fre"uencies is prefera%le in most application cases.
The (DFT can %e expressed %y the following iterative algorithm
- i i
3
E EW R
( (
*
=
$ (-5.*
( * ( ( (
(
i i i i
EW R x xA F
= = $ (-5.-
( (
. (
* (
* (
(
E R E
E R x
S
=
i -
i
i
dia.
$ (-5.0
M (M
- ( * ( i i
dia. S W =
+
$ (-5.7
for the iteration num%er i@*$-$0$A/$ where (-5.* is (-1 expressed in the matrix form. The
exponents matrix E ("x3 has elements
n
t f j
e
t -
or
! f j
n
e
t -
if the sampling is uniform. /y default
the diagonal weight matrix W
(i
(3x3 for the first iteration is a unit matrix W
(*
@I. #f other
diagonal matrix is used as input of the (DFT algorithm then it must have at least " non'ero
elements. For the su%se"uent iterations W
(i
is filled with power spectrum values calculated %y
(-5.7. There could %e additional criteria for stopping the iterations %efore the maximum num%er
of iterations / is reached$ for example$ the iterations could %e interrupted$ if the relative change of
the power spectrum |sum(W
(i
-sum(W
(i#*
|>sum(W
(-
$ for iN-$ is smaller than a given threshold.
The #DFT can %e applied to output F and return %ac& original "#samples of uniform or
nonuniform se"uence
-
3
FE x
*
=
. (-6
Extended summary of Dr.Sc.Comp. thesis 9
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
Since the length of the fre"uency set 3>"$ then (-6 can %e modified to o%tain a se"uence x
*
(*x3 ! x
*
(t
m
$ m@,$*$-$A$3!*$
-
3
3
FE x
*
=
o
$ (-+
where exponents matrix E
3
(3x3 has elements
m n
t f j
e
t -
or
m! f j
n
e
t -
for uniform sampling case.
The reconstructed %y the formula (-+ se"uence is the original se"uence plus forward and
%ac&ward extrapolation of x to length 3 and>or interpolation if there are gaps inside of x. The
maximum fre"uency resolution of the iterative algorithm is limited %y the length 3 of fre"uency
set$ not %y the length " of se"uence as in application of classical DFT. This mean$ the (DFT is
a%le to increase the fre"uency resolution 3>" times in comparison with the classical DFT. This
can %e verified %y comparing the diagonal elements of the product of #DFT and DFT %asis$
* >
*
( < = 3 "
3
dia.
-
E E $ with the relationship$ * > .
*
*
( , s = < S F A E
3 3
dia.
-
$
corresponding to the #DFT and (DFT %asis A (-5.-. 3t the same time there is a restriction on
the fre"uency resolution sum(F.>S@3"$ which is satisfied %y iteration$ and in order to achieve
a high resolution at certain fre"uencies$ the (DFT must decrease the resolution on other
fre"uencies. The deviation |sum(F.>S-3"| also could %e used as an additional criteria for
stopping of the (DFT iterations$ %ecause of indicate the possi%le inaccuracy in the o%tained
result$ mainly caused %y the finite precision in calculations. #f this happens$ the result of the
previous (DFT iteration should %e considered as a final one.
#n a %order!case 3@"$ the iterative algorithm output do not depend on weight matrix W and
the optimal (DFT %asis can %e found in a non!iterative way (as result of the first (DFT
iteration.
' E!FT and ot%er nonparametric approac%es
#n the previous sections$ starting with the Fourier integral (* and using its orthogonality
property (-$ %y esta%lishing and solving the minimum least s"uare error estimators (+$ the
(xtended DFT is o%tained analytically. Dow letFs ma&e comparison with other nonparametric
methods ! Capon filter$ 4enerali'ed (8eighted 9east S"uares (489S solution and :igh!
;esolution Discrete Fourier Transform introduced %y Sacchi$ =lrych and 8al&er in *++6$ and
try to analy'e the ways and opportunities of derivation of an iterative (DFT algorithm %ased
on these approaches.
'.1 Capon filter approac%
The Capon filter also &nown as <inimum Gariance spectrum estimate (see [8$ 9$ 19$ 22] can
%e viewed as the output of a %an& of filters with each filter centered at one of the analysis
fre"uencies
( ) $... - $ * $ , $
O
( ( (
*
,
= = =
=
n ! h ! n x n! y
"
e e e
h x
. (0,
#n the matrix notation ( ) ( ) ( ) | | ! " n x ! n x ! n x n! x * ( $...$ - ( $ * ( $ (
O
+ = x is the filter input
signal and h
e
@[h
e
(,$h
e
(!$h
e
(-!$...$h
e
(("!*!]
!
is the filter coefficients. :ere the su%script
4 indicate a dependence on the filterPs center fre"uency.
The Capon filter is designed to minimi'e the variance on the filter output
{ } { } { } { }
e e e e e e e e e
c c c c o h R h h x x h h x x h
x
- - - - - -
y
n! y n! y n! y = = = = =
O O O O
( ( (
-
-
$ (0*
su%)ect to the constraint that its fre"uency response at the fre"uency of interest 4 has unity
gain
Extended summary of Dr.Sc.Comp. thesis 10
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
* ( (
*
,
= = =
=
e e
e
e
e h E
! ! j
"
e ! h -
$ (0-.*
* ( (
H
*
,
H
= = =
=
e e
e
e
e E h
- ! j
"
e ! h -
$ (0-.-
where {} . c denotes the expectation operator and the matrix E
e
("x* has elements
! j
e
e
. The
constraints (0-.* and (0-.- must %e satisfied %y the filter (0, and %y the :ermitian transpose
filter
- - -
n! y x h
O
(
e e
= $ correspondingly. The matrix { } x x R
O O-
x
c = ("x" is the sample
autocorrelation matrix and it can %e composed from the values of the signal autocorrelation
function. For example$ so called %iased estimate is calculated %y
( ) * $...$ - $ * $ , $ ( (
*
(
*
,
H
= + =
=
" l ! x ! l x
"
l! r
l "
xx
(00
and$ ta&ing into account that ( (
H
l! r l! r
xx xx
= $ the sample autocorrelation matrix is filled as
( )
( )
( )
( ) ( ) ( )
(
(
(
(
(
(
, ( ... 0 ( - ( * (
... ... ... ... ...
0 ( ... , ( ( - (
- ( ... ( , ( (
* ( ... - ( ( , (
* $ * - $ * * $ * , $ *
* $ - - $ - * $ - , $ -
* $ * - $ * * $ * , $ *
* $ , - $ , * $ , , $ ,
" " " " "
"
"
"
x
r ! " r ! " r ! " r
! " r r ! r ! r
! " r ! r r ! r
! " r ! r ! r r
R
. (07
<athematically$ the Capon filter coefficients can %e o%tained %y minimi'ing the variance (0*
under the constrains given %y (0-.* and (0-.-
min * ( * (
H
= =
e e e e e e
E h h E h R h
- !
x
-
5 $ (01
where $ are 9agrange multipliers. The conditions
, =
c
c
e
h
5
and
, =
c
c
-
5
e
h
have to %e fulfilled
to determine the minimum of (01. /oth re"uirements lead to the same solution
H *
H *
e e
e
e
E R E
E R
h
=
x
!
x
. (02
and$ traditionally$ the Capon power spectrum is computed as
H *
*
(
e e
e e
e
E R E
h R h
= =
x
! x
-
Capon
6
. (05
#n order to o%tain an iterative (DFT algorithm from the original Capon filter approach$ the
sample autocorrelation matrix R
x
(07 has to %e su%stituted %y R
!
@E
H
WE
!
. The matrix R
!
("x"
can also %e o%tained as a transpose of the (DFT matrix R defined %y (-2. The elements of
"uadratic diagonal matrix W (3x3 represent an estimate of power at time moment n!@,$
determined from one sample at the output of each Capon filter
( )
( )
-
H
*
H
*
- -
O
O
, (
e e
e
e e
E R E
E R x
h x
= =
! !
!
y $ (06
where the filter input se"uence x
O
(0, is related to the (DFT input se"uence x as
( ) ! " x ! x * ( (
O
+ = or
( (
O
* +
=
"
t x t x
$ @,$-*$--$..$-("-*$ for uniformly or
nonuniformly sampled se"uence cases$ respectively.
Finally$ an iterative algorithm$ with the initial condition for W
(*
@I$ can %e formed as follows
! i i !
E W E R
( H (
=
$ (0+.*
( )
( ) ( )
H
*
(
H
*
(
(
.
O
E R E
E R x
S
=
i ! !
i !
i
Capon
dia.
$ (0+.-
Extended summary of Dr.Sc.Comp. thesis 11
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
M (M
- ( * ( i
Capon
i
dia. S W =
+
$ (0+.0
with the iteration num%er i@*$-$0$A/. The estimate of the power spectrum
-
M M
Capon
S
coincides
with the results of the (DFT$ while the phase spectrum$ definitely$ is different. #t should %e noted
that the calculation of the Capon filter output power %y formula (05 is theoretically well
)ustified$ whereas the derivation of (0+ re"uires ad hoc assumptions and su%stitutions$ and
actually is a measurement of power o%tained from )ust a one sample at the output of filter. This
leads to conclusion that the approach (0+ is simply a filter!%an& interpretation of the (DFT$
similarly to the DFT which can also %e considered as %an& of filters. #n addition$ an iterative
algorithm derived on the %asis of the filter!%an& can not reveal all the (DFT capacity such as the
a%ility to estimate the DFT (-5.- and restore the signal (-6$ -+.
'.2 "()* solution
The 4enerali'ed (8eighted 9east S"uares approach (see [13$ 16, 23, 32] in the fre"uency
analysis is %ased on the following data model
7 89LS
!
S e E x + = (
H
e
e
$ (7,
with e
7
denoting the noise and interference (signals at fre"uencies other than 4 component$
and
(
H
e
e 89LS
S E
representing the signal component on the fre"uency of interest with
un&nown complex amplitude S
89LS
(4. The 489S minimi'es
Q ( R Q ( R
H * H
e e
e e 89LS
! -
89LS
!
S S E x Q E x
$ (7*
which is solved %y
H *
*
(
e e
e
e
E Q E
x Q E
=
!
! !
89LS
S
$ (7-
where Q ("x" is the covariance matrix of the data model component e
7
. There are two
special cases of 489S called 8eighted 9east S"uares (89S and ordinary 9east S"uares
(9S. 89S occurs when all the off!diagonal entries of Q are ,$ while 9S solution is o%tained
from the 489S under assumption that e
7
in (7, is a white noise$ hence Q@I.
The pro%lem of 489S estimator is that$ in general$ the covariance matrix Q is not &nown$
and must %e estimated from the data along with the S
89LS
(4. The initial estimate (the *
st
iteration could %e e"ual to 9S solution$ it is (7- with Q@I. Dext$ to ensure that the 489S
solution wor&s in an iterative way as (DFT do$ covariance matrix Q should %e replaced %y
R
!
@E
H
WE
!
. 3s a result$ 489S solution (7- coincides with the (DTFT formula (-0.0
( )
( )
( (
*
*
H
*
*
e e
o
e e
e
e e
e
S S
-
! !
! ! !
89LS
= = =
E R E
E xR
E R E
x R E
(70
and$ as shown in the Section 0.0.0$ can %e successfully used for calculation of the amplitude
spectrum iteratively. 3lthough su%stitution of a noise matrix %y R
!
would %e easy done$ it is
not supported %y 489S data model (7,$ from where the matrix Q represents the data model
component e
7
only and the signal component (
H
e
e 89LS
S E must %e excluded from it$ whereas
the matrix R
!
is calculated for the entire signal x
!
$ including e
7
and (
H
e
e 89LS
S E . Furthermore$
the derivation of (DFT shows that the signal can %e restored %y formula (-6$ which fits
perfectly to the iterative update of the matrix R. =sing estimates
( ( e e
o
S S
89LS
=
in the
data model (7, leads to a predetermined split of overall energy at the fre"uency 4 in %etween
components (
H
e
e 89LS
S E and e
7
. The conclusion reached is that ma&ing the derivation of the
(xtended DFT algorithm possi%le$ invalidates 489S minimi'ation expression (7* which
re"uire separation of %oth data model components.
Extended summary of Dr.Sc.Comp. thesis 12
Dr.Sc.Comp. Vilnis Liepi Email: vilnislp@gmail.com
'. +ig%,-esolution !FT
The third method considered here is the :igh!;esolution DFT (:;DFT [7]. The authors
presented an iterative nonparametric approach of spectral estimation$ which minimi'es the cost
function deduced from /ayesP theorem and$ as well as the (xtended DFT$ ma&es it possi%le to
o%tain high!resolution Fourier spectrum. The :;DFT algorithm can %e reduced to the following
iterative procedure:
- i i
3
E EW R
( (
*
= $ (77.*
( * ( (
(
i i i
-:DF!
EW R x F
= $ (77.-
|
.
|
\
|
=
+
-
( * (
*
i
-:DF!
i
3
dia. F W
$ (77.0
for iteration num%er i@*$-$0$A/ and with the initial condition W
(*
@I.
The #DFT (-6 applied for any iteration output (77.-$ return %ac& the se"uence x undistorted.
The main difference %etween approaches is that the :;DFT algorithm lac& of formula for
estimate of amplitude spectrum (-5.0. #nstead$ as input for the next iteration$ it uses the Fourier
spectrum estimated in previous iteration (77.0. Therefore$ the results of the :;DFT differ from
output of the (DFT significantly.
. Computer simulations
The computer modeling of the (DFT algorithm are performed for the complex!value test
signal used in [6]. True spectrum of the test signal consisting of a %and!limited noise in fre"uency
range R-,.1...-,.-1Q :'$ a rectangular pulse in range R,...,.-1Q :' and unit power complex
exponent at fre"uency ,.01 :'. The signal upper fre"uency is f
u
@,.1 :'. =niform and
nonuniform test se"uences of length "@27 samples are derived %y simulating *,!%it 3nalog!to!
Digital Converter (3DC. Sampling and mean sampling periods of %oth se"uences are e"ual$
!=!
s
@*s. Sampling time moments for the nonuniform se"uence are generated as$ t
@!Bt
$
@,$-$...$"-*$ where Kt