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AN ADAPTIVE RECURSIVE FILTER
Soon-Ju Ko
NAVAL POSTGRADUATE SCHOOL
Monterey, California
THESISAN ADAPTIVE RE-CURSIVE FILTER
by
Soon-Ju Ko
December 1977
Thesis Advisor: Sydney R. Parker
Approved for public release; distribution unlimited
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An Adaptive Recursive Filter5. TYPE OF REPORT 4 PERIOD COVERED
Electrical EngineerDecember, 197/
8. PERFORMING ORG. REPORT NUMBER
7. AUTHORf*;
Soon-Ju Ko
8. CONTRACT OR GRANT NLMBERf«J
9. PERFORMING ORGANIZATION NAME AND ADDRESS 10. PROGRAM ELEMENT. PROJECT. TASKAREA 8 WORK UNIT NUMBERS
II. CONTROLLING OFFICE NAME AND ADDRESS
Naval Postgraduate SchoolMonterey, California 93940
12. REPORT DATE
December 197713. NUMBER OF PAGES109
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Naval Postgraduate SchoolMonterey, California 93940
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20. ABSTRACT (Continue on tavaraa alda II nacaaaary and Idantlty by block number;
An adaptive recursive digital filter is presented in whichfeedback and feedforward gains are adjusted adaptively to mini-mize a least square performance function on a sliding windowaveraging process. A two-dimensional version of the adaptivefilter is developed and its performance compared with the optimalWiener filter. The filter is shown to be effective in separatingthree diagonal trajectory streaks from a background of correlated
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noise added to white noise. Although the recursive adaptive filterapproaches the optimal Wiener filter in performance, it does notrequire a priori statistical knowledge as does the Wiener filterto which it is compared. The results indicate that the recursiveadaptive filter "learns" the statistics and adapts.
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Approved for public release; distribution unlimited
AN ADAPTIVE RECURSIVE FILTER
by
Soon-Ju KoLieutenant, Republic or" Korea Navy
B.S.E.E., Seoul National University, 1975
Submitted in partial fulfillment of therequirements for the degree of
ELECTRICAL ENGINEER
from theNAVAL POSTGPADUATE SCHOOL
December 1977
ABSTRACT
An adaptive recursive digital filter is presented in
which feedback and feedforward gains are adjusted adaptively
to minimize a least square performance function on a sliding win
dow averaging process. A two-dimensional version of the adap-
tive filter is developed and its performance compared with
the optimal Wiener filter. The filter is shown to be effec-
tive in separating three diagonal trajectory streaks from a
background of correlated noise added to ivhite noise. Although
the recursive adaptive filter approaches the optimal Wiener
filter in performance, it does not require a priori statistical
knowledge as does the Wiener filter to which it is compared.
The results indicate that the recursive adaptive filter "learns"
the statistics and adapts.
TABLE OF CONTENTS
I. INTRODUCTION 7
II. LINEAR STOCHASTIC PROCESSES 10
A. INTRODUCTION 10
B. DEFINITION AND PROPERTIES OF LINEAR
STOCHASTIC PROCESSES 11
C. MODELING OF LINEAR STOCHASTIC PROCESS -22
1. Introduction 22
2. Method of Least Squares 25
3. Method of Filter Response 34
III. ADAPTIVE FILTERS 43
A. NONRECURSIVE FILTER 43
1. Introduction 43
2. Performance Surface, Gradient and
Wiener Solution 45
3. LMS Algorithm for One-Dimensional Filter 48
4. Two-Dimensional Adaptive Filter 52
a. Adaptive Filter Structure 52
b. Wiener Solution 54
c. LMS Algorithm 56
B
.
RECURSIVE FILTER 58
1. Introduction 58
2. One-Dimensional Adaptive Recursive
Filter 59
a. Structure 59
b. Problem in Wiener Solution 61
c. LMS Algorithm 64
3. Two-Dimensional Recursive Adaptive Filter
IV. ADAPTIVE NOISE CANCELLER 77
A. THE CONCEPT OF ADAPTIVE NOISE CANCELLING 7 7
B. NOISE CANCELLING WITHOUT AN EXTERNAL
REFERENCE INPUT 82
v - EXPERIMENT AND RESULTS 85
V. CONCLUSIONS 105
BIBLIOGRAPHY 108
INITIAL DISTRIBUTION LIST 109
I. INTRODUCTION
The term "filter" is often applied to any device or
system that processes incoming signals or other data in such a
way as to eliminate noise, to smooth signals, to identify each
signal as belonging to a particular class, or to predict the
input signal from instant to instant. There is an abundance
of literature covering the theories involved under the head-
ings of estimation, identification, modeling, prediction, etc.
The usual method of estimating a signal corrupted by noise is
to pass it through a filter that tends to suppress the noise
while leaving the signal relatively unchanged. The design of
such filters falls in the domain of optimal filtering, which
originated with the pioneering work of Wiener [8] and was ex-
tended and enhanced by the work of Kalman-Bucy [9] and others.
Filters used for the above purpose can be fixed or adap-
tive. The design of a fixed filter is based on a priori know-
ledge of both signal and noise statistics. On the other hand,
adaptive filters have the ability to adjust their own para-
meters automatically, and their design requires little or
no a priori knowledge of signal or noise characteristics.
This work presents an approach to signal filtering using an
adaptive filter that is in some sense self-designing.
The adaptive filter described here bases its own "design" (its
adjustment of internal parameters) upon estimated (measured) sta-
tistical characteristics of input and output signals.
The statistics are not measured explicitly and then used
to design the filter; rather, the filter design is accomplished
in a single process by a recursive algorithm that automatically
updates the system parameters with the arrival of each new data
sample. It is assumed that the input and the output at the
sampling instants are the only measurable quantities of the
system. It is also assumed that the unknown filter coefficients
parameters) to be designed enter linearly in the difference
equations which describes the self-designing process.
The steepest descent method is employed in which the pre-
vailing filter parameter vectors are perturbed at each itera-
tion in a manner so as to decrease a prescribed functional
(error criterion or cost function) to be minimized. The steep-
est descent method is one of the well known gradient based
algorithms.
For the case where the functional being minimized is the
mean square error, where the error is the difference between
filter output signal and the desired signal, the filter is
called the least mean square filter (LMS filter) . Various
adaptive algorithms are currently available depending upon
the cost function and the method used to minimize cost function.
The popularly used performance criteria are the least mean
square criterion, the maximum likelihood ratio (MLR) criterion,
and the maximum signal to noise ratio (SNR) criterion. Here
the LMS criterion only is studied and the steepest descent
method is employed. Inevitable errors in the estimation of
the statistics prevent the adaptive filter from delivering
optimal performance since the adaptive filter is not based on
the a priori knowledge of statistics. In Chapter II, the con-
cept of linear stochastic processes is reviewed as a preliminary
study for this thesis, and the modeling of stochastic processes
is studied. These can be considered as background material
for the following chapters.
In Chapter III, the concept of adaptive filters is intro-
duced and the structure of the signal and the mathematical
model of the processor is delivered. The algorithm for the non-
recursive adaptive filter by Widrow [1] is reviewed and the new
algorithm for the recursive adaptive filter is developed as is
the two-dimensional adaptive filtering process.
In Chapter IV, the adaptive noise cancelling concept is
analyzed rather qualitatively and its application to the special
case in which no desired signal is available is analyzed. In
Chapter V an experiment is performed through computer simulation
to check the feasibility of algorithms developed in the pre-
vious chapter and a comparison with the optimal Wiener solution
is made. In Chapter VI, the conclusions are presented together
with a summary of the experimental results and suggestions for
further research.
II. LINEAR STOCHASTIC PROCESSES
A. INTRODUCTION
The problem of defining a random process is of considerable
importance in the analysis of systems subject to noise distur-
bance. Often a partial definition of the process will suffice
as in the case of linear least mean square error filtering,
where only a knowledge of the correlation function is required.
For other problems, such as those involving nonlinear filtering,
more complete information will generally be needed. A complete
description of a random process requires a knowledge of the
distribution functions of all orders. But in practice few
processes apart from the normal and Markov are defined in this
manner. For the purpose of analysis, a model to generate the
random process is desirable and for a model to give a complete
description of the process, the distribution functions should be
derivable from the model. While both continuous and discrete-
time linear process may be defined, only the discrete -time case
will be considered here. The discrete -time linear process
can be considered to be the result of the digital filtering of
a sequence of independent and identically distributed (IID)
random variables.
The linear processes are important since they are inherently
simple in terms of physical considerations and form a class
which includes many discrete time normal random processes.
In the following section the definition and properties of the
linear processes are summarized.
B. DEFINITION AND PROPERTIES OF LINEAR STOCHASTIC PROCESSES
It has been found useful in the theory of stochastic processes
to divide stochastic processes into two broad classes: sta-
tionary and nonstationary . Intuitively, a stationary process
is one whose distribution remains the same as time progresses,
because the random mechanism producing the process is not chang-
ing as time progresses. A nonstationary process is one which
is not stationary.
Let {x(i), ieT} be a stochastic process with finite second
moments. Its mean value sequence , denoted by m(i) , is defined
for all i in T by
m(i) = E [x(i)] (2-1)
and its covariance kernel, denoted by K (j.i), is defined
for all j and i in T by
K (j.i) = Cov [ x (j), x (i)] (2-2)
An index set T is said to be a linear index set if it has the
property that the sum i+-j of any two numbers i and j of T also
belongs to T. Examples of such index sets are T= {1,2 , . . .},
T = {o,±l, ±2, ....}, T={i;i>o} and T={i:-°° < i < » }
A stochastic process (x(i), ieT }, whose index set T is linear,
is said to be
i) strictly stationary of order k, wherek is a given positive
integer, if any k points i, i+1, . . . i + k in T+ , where
T+ - (x (i) , i>_o}, and any j in T+ , the k dimensional random
vectors (x(i) , x (i+1). . x (i+k) }and{ x(i+j),. . . x(i+j+k)}
are identically distributed.
1
1
ii) strictly stationary if for any integer k, it is
strictly stationary of order k.
iii) widesense stationary (covariance stationary) if it
possesses finite second moments, if its index set T is linear,
and if its covariance kernel K (j , i) is a function only of the
absolute difference |j-i|, in the sense that there exists a
function R(n) such that for all j and i in T
K(j,i) = Cov[x(j), x(i)] = Rxx (j-i) (2-3)
or more precisely, Rj,,,^) has the property for every i and j in Z+
Cov [x(i) ,x (i+m) ] = E [x (i) x (i+m)
]
= Rxx (m ) (2-4)
We call RxX (m ) the covariance function (autocorrelation function)
of wide sense stationary time series (x(i), ieT+ } .
The second problem concerns the concepts of ergodicity
and the strong law of large numbers in terms of linear processes
To present a complete discussion of this question is not rea-
sonable for review purposes, but it is interesting to con-
sider certain aspects of it. For strict sense stationary pro-
cesses, the ergodic theorem is the strong law of large numbers
and states that
if{x(i),ieT } is a strict sense stationary, ergodic
random process and E[ |x (o)
|] <°°,
mthen l^m
_J_Z x (i) =E [x (o) ] with probability 1. (2-5)
m i=l
12
In general, a stochastic process is said to be ergodic if it
has the property that sample (or time) averages formed from
an observed sequence of the process may be used as an approxima-
tion to the corresponding ensemble average.
Stationarity and ergodicity concepts are readily extended to
two-dimensional random fields (for two dimensional signal, the
term random field is preferred to random process)
.
The assumption that the field is stationary means that
the statistics of a point in the field are not dependent on the
location of the point. Then, a stationary, two-dimensional
field has an autocorrelation function defined as:
Rxx (m,n)A E{x(k,£)x(k+m,£+n)
}
(2-6)
and it is also said to be ergodic if the statistical (ensemble)
average of random field x(k,l) is equal to the spatial averag-
ing of all points. That means
E[x(k,i)] = <x> (2-7)
where <x> by definition represents spatial averaging
<x >=lim -i- _LMiMi .
Mi-00 m m ^ 5 ' (2-8)M Mi M 2 o o2-M»
Now, consider a stationary sequence of random variables
{x(i) , ieT}. The correlation function of the sequence may
be written in the form
I^x(n^ e
1WndF(w) (2-9)
L13
where F (w) is a nondecreasing function, called the spectral
distribution. If F(w) is absolutely continuous with F (w) =
f (w) almost everywhere, we may write
it iwn
-7T
Rxx (n) =| e f(w) dw7 (2-10)
Under certain conditions, (e.g. £ |R(n)|<<», finite), then=o
correlation function may be inverted to yield the spectral
density as
f (w)= _1_ i Rxx (n)e~inw
(2-11)2tt -°°
A sequence of random variables(x (i) , ieT} is said to be a
process of moving average if it admits the mean square repre-
sentation
x (i) =_E h(i-j) u(j) (2-12)
where(u (j), jeT} is a collection of orthornomal random
variables (sequence of white noise) . If the sequence (h(i)
,
ieT} is one sided (ie, h(i) = 0, i<0), then {x(i), ieT} is
called a one sided moving average process.
A process is said to be regular if the error for prediction
one time unit ahead is nonzero. It is known that a process is
one of moving averages if and only if its spectral distribution
function is absolutely continuous. Furthermore, a process with
an absolutely continuous spectral distribution function is
regular if its moving average representation is one sided. These
facts serve to motivate the following definition of a linear
process
.
l A
DEFINITION [5]
A linear process {x(i) / ieT} is one having the structure
j
x(j) = Eh(i)u(j-i) = Zh(j-i)u(i) , (2-13)o -°°
where {u(i), ieT} is a sequence of independent and identically
distributed (IID) random variables. The set of real constants00
{h(i) , ieT+} is such that E|h(i)| <<», and the functiono
00
H(z) = Z h(i) z~L where z is a complex variable, is analytic
o
and has no poles outside the unit circle in the z plane. The
correlation function of this process is given by
5£n) = g h(j)h(j+n) (2-14)
and the corresponding spectral density f(w) is
f(w) = - |? h(j)e^ w|
2 = i|H(e iw )
|
2 (2-15)2
TT ° 2 *
It is assumed further that the process (2-13) has a rational
spectral density, that is
f(w) = iI H(elw )
I
2 = l B(e )
A(e lw )
2, (2-16)
where both A(e ) and B(elw ) are polynomials in elw of
finite order with all their poles inside the unit circle.
The process (2-13) may be generated by passing the sequence
{u(i), ieT} through the digital filter H(z). By the assumption
in the definition of the linear process and by the restrictions
on the spectrum, there exists an inversion D(z) where
D (z) = -MeL = _1
—
B(Z) H(z)
In general, D(z) and H(z) may be infinite polynomial in z.
D(z) may be written as the one-sided sequence
D(z) = g d^" 1 (2-17)
Passing the x(i) sequences through the digital filter will
recover the generating sequence u(i), that is
J
u(j) =_E dtj-j)x(ij= § dli)x(j-i) (2-18)
This is called an operation inversion. In general, we will have,
assuming a = 1
n _iEb . z oo
H(z) = -2-=- r- = E h(i)z_1
(2-19)1 +
y a . z o
and the process{x (i) , ieT} may be represented in two ways
i) x(j) = E h(i)u(j-i)(2-20)
m . nn) x(j) = i biu(j-i)- laix(j-i)
o 1
The second representation indicates that if H(z) is an all-
pole function, then we have
miii) E a^x(j-i) = u(j). (2-21)
o
In this case, since inversion uses only a finite number of past
samples, the process is called "finitely invertable." It is
clear that finitely invertable linear processes form a subclass
of autoregressive schemes for which case the set {u(i)} in
(2-21) would be orthonomal rather than independent. The con-
cepts and definitions above can be readily extended to a
two-dimensional linear process. A two-dimensional linear
process {x(m,n), TTiezj, nez2) could have the structure
x(m,n) = ZZh(k,g )u(k-m,£-n)oo
m n= ZZ h(k-m,ft-n)u(k,H) , (2-22)
where (u(k,§L), kez-]_, £.£Z2} is a two-dimensional sequence of
IID random variables with zero mean and unit variance. The set
of real constant {h(k.SL), kez, +, isz2+ } is such that
0O0O , n0000
1 n 1 (1 — lc — V
ZZ I h(k,j)) |<°° , and the function H(Z,,Z ) = EZh(k,X)Z, Z S00 J. z 00 ± z
where Z, and Z2 are complex variables, is analytic. The correla-
tion function of this process is given by
Rw<m ' n ) = ??h(k,Jl)h(k+m,!+n) (2-23)•*•* 00
and the corresponding spectral density f(w,,W2) is
f(w1#w2 ) =(^)
2I? h(k,l)eiwlk e
iw2^
I ^ 2
2TT.H(eiwl, e
iw2) (2-24)
With the same reasoning as in the one-dimensional case, we have
i n general
,
H(ZX , Z
2 )
Nl
N2
Z Z bij 1 2
Z-1
Z ^
M, M 91+Z
1Z aij Z,
1Z2_:i
1=0 j=o(i, j)^(o,o)
00 00 n —V —
u
5 g h(k.J)z1
kz2
l. (2-25)
and the process (x(k,l), keZ,, leZ^Jmay be represented in two
ways
i) x(k,H) = ??h(i.j)u(k-i,l-j)oo
N1N2 Mi M 2
ii) x(k,£) = ZIb. .u(k-i,j2.-j) - E I a, ,x (k-i, jL-j) (2-26)00 1 J 00 J-j
(i rj)^(0,0)
It should be noted that the moving average scheme would be ac-
complished by passing the orthonomal IID random variables through
the nonrecursive filter and the autoregressive schemes through
the recursive filter for both one-dimensional and two-dimensional
random process.
In the study of systems subject to the random signals, the
concept of power spectral density function is of importance.
For a given transfer function of a linear filter, the cross power
spectrum between input and output of the filter, and the output
power spectrum is of primary concern.
Consider a (continuous) system subjected to a random input
signal. Given a linear system with a transfer function
H(jw), the input to the filter a stationary process x.(t)
with an autocorrelation function R (z) and a power spectral
density function G (w) , then the following relationships
are obtained
Gxy (w) - Gxx (w)H*(jw)
G„„(w) = G (w)H(jw) (2-27)yy *\Y
Combining above two equations,
Gyy
(w) = Gxx (w)|H(jw)| 2, (2-28)
18
where G (w) is the cross power spec trum and G (w) is the
output power spectrum, the cross correlation would be cal-
culated by using the inverse Fourier transform of G (w)
.
For the continuous two-dimensional linear system of
H (Jwi/Jw2)/ subjected to a stationary two-dimensional input
signal of power spectrum Gxx (w, , w2
)
.
Bxy (wl' w2*
= Gxx (wl'
W2)H ^W1'^ W2^
Gyy
(wl'
w2 )
= Gxy (wl'
w2
} H ( jwl'
jw2
) (2-29)
Again combining above two relationships, the output power
spectral density function is
G (w1# w2
) = Gxx (w1
, w2)|H(jw1# jw
2 ) | (2-30)
Consider a discrete linear system to which a random input se-
quence is applied with a transfer function H(z). The input
sequence is a stationary process x(i) with an autocorrelation
function Rvv (m) and its?-transform G (z), where G (z) is equi-xx J xx ' xx ^
valent to the" power spectral density in the continuous case,i.e
Gxx< 2>
4 3»Wm>] (2-3D
Then,
G (z) = G (z) H(z)xy xx K'
v
Gw (z) = G (z) H(z)H(z_1
) (2-32)
For the two-dimensional discrete case,
Gxy
(zl'
z2
}= G
xx(z
l'Z2)H^1' z
2}
and Gy^ 1
^2
)= G
xx(z
l'z2
} H (zl'
z2
) H ^l^ Z ~2~] ( 2_33 )
19
As a summary of this section, a discrete-time linear process can be
considered to be the result of digital filtering an indepen-
dent identical random sequence having zeromean and unit variance,
The moving average scheme is the result of filtering through the
nonrecursive filter and the autoregressive scheme is that of
filtering through the recursive filter.
And for a linear system, the relations between transfer
function, power spectrum, and auto correlation are given by
G = G |h|2
yy xx ''
G = G Hxy xx
R (autocorrelation function) - 7[ transform of power
spectral density function.
The Figure 2-1 shows the block diagram which describes
the various relations and concepts.
20
ORTHONORMALI . I . D . RANDOMSEQUENCE
WHITE NOISE \
OF ZERO MEAN\AND UNIT'VARIANCE ;
LINEAR SYSTEM
H
LINEARCORRELATEDRANDOMPROCESS.
G (Z)xx
x(n)RANDOM INPUT
G (Z)=G (Z)H(Z)H(Z1
)VV XX
y(n)
GxxV Z2
)
x(m,n)
H(Z1
, Z2
)
h (m,n)
JGyy(Z lf Z
2)=GXX (Z
1,Z
2)H(Z
1,Z
2)
H(Zjlz2l)
WHITE NOISE
G (Z,, Z )=lxx 1 ' 2
G (Z, ,
yy iz2
) H(Z 1/ Z2)H(Z
1
" 1/ Z
2
~ 1)
FIGURE 2 -1STATISTICAL
PROPERTIES OF LINEAR SYSTEM
21
C. MODELING OF LINEAR STOCHASTIC PROCESS
1. Introduction
A more active concern at this time is that of system
modeling. It has been shown in a previous section that a linear
stochastic process (or field) can be generated by filtering
white noise through a linear filter. The problem can be stated
as follows. What is the filter equation (difference equation)
that produces a typical random process with a specified autocorrelation
function? That is, with the knowledge of second -order statistics,
determine the filter coefficient a's and b's in equations (2-20)
and (2-26) . It is clear that if one is successful in developing
a parametric model for the behavior of some random process,
then the model can be used for different applications, such as
prediction, estimation, smoothing, etc.. As far as the general
modeling problem goes, one of the most powerful models currently
in use is that where a signal x(n) is considered to be the out-
put of some system (filter) with unknown input u(n) such that
the following relationship holdsP q
x(n) = - I clkx(n-k) + gZ
Qb£u(n-£), b =l (2-34)
JC*- J.
where a^, 1<_ k<_ p , b£ 1 <_5,<_q
andthe gain G are the parameters of the hypothesized system.
Equation (2-23) says that the "output" x(n) is a linear function
of past outputs and present and past inputs. That is, the
signal x(n) is predictable from linear combinations of past
22
outputs and inputs . For 1he two-dimensional case, the difference
equation corresponding to equation (2-34) would be
Mx
M2
Lj L 2
x(k,£) = - Z Z ai-x(k-i / £-j)+G Z Z b ,
2u (k-^, £-£
2 )
i=o j=o -1 £,=u £
2=u '
where (i,j)^(0, 0) (2-35)
Equations (2-34)ard(2-35) can also be specified in the frequency
domain by taking the z transform of both sides of Eq (2-34) and Eq
(2-35) .
q -£
H(z) = jL<f> = G l+ *i,
1'*' (2 "36)U(Z) p ,
1+ Z a. ZK
k=lk
oo
where X(Z) = Z x(n)Z~n is the Z transform of x(n), and u(Z)n= —°°
is the z transform of u(n)
.
For the two-dimensional case,
L, L 9-£.. -£
s i \ *2
zi
z2
(2 -37)
X(Z,,Z~) £,=0£ 9=0
U<W Z
1Z2
a Z"1
Z"3
x A1 +.
L L aij
zl
z2i=o j=o J
(i. j)^(o.o)
where X(Z1,Z
2) = 3[x(k,£)]
U(Z1,Z
2) = 3[u(k,£)]
H(Z)an<}I(Z lf Z2 ) in Eqs (2-36)arri (2-37) are the general pole zero
models.
23
The roots of the numerator and denominator polynomials are the
zeros and poles of the model, respectively. There are two
special cases of the model that are of interest,
i) all- zero model ak = 0, a-ij =
ii) all- pole model b = 0, bJU£2
=
1 £ l 1 3
<_ £1
<_ L1
<_ l2< L
2
But (£1
, £2)^ (0,0)
As mentioned in section II-B, the all- zero model is known
in the statistical literature as the moving average (MA) model
and the all -pole model is then known as the autoregressive (AR)
model. The pole-zero model is then known as the autoregressive
moving average (ARMA) model. It should be recalled here that we
are interested in the case where u(n) or u(kTi) is white noise, and
this will be treated as a special case in the following. The
modeling problem can be stated as"given signal x(n) or x(k,2j,
find the filter coefficients (a's and b's)and the gain, G, in
Equation (2-34) in some manner." Two approaches will be given for
a solution of the above problem. The first is the method of least
squares which is based on the optimal estimation concept, and the
second isthefilter response method in which linear system pro-
perties are used. The one-dimensional case will be treated
first, then two-dimensional case including some examples. For
example, a lowpass correlated random process (field) and band
24
limited random process (field) are chosen since they represent
practical examples.
2. The Method of Least Squares
Athoughthe following can be applied to the deterministic
signal and stochastic processes, stationary or nonstationary
,
it is emphasized for only a stationary random process and only the
all-pole model is considered [6] . In the all- pole model, we
assume that the signal x(n) is given by as a linear combination
of past values and some input u(n)
:
Px(n) = - Z A vx(n-k) + Gu(n) (2-38)
k=l K
where G is a gain factor.
Here, it is assumed that the input u(n) is totally unknown,
which is the case in many applications. Therefore, the
signal x(n) can be predicted only approximately from a linearly
weighted summation of past samples.
Let this approximation of x(n) be x(n), where
x(n) = - I Avx ( n"k ) / (2-39)k=l
K
then the error between the actual value x(n) and the predicted
value x(n) is given by
AP
e(n) = x(n) - x(n) = x(n) + E A vx(n-k). (2-40)k=l
K
e(n) is also known as the residual. In the method of least
squares, the parameter ^'s are obtained as a result of the
minimization of the mean square error with respect to all of
the parameters
.
25
If the signal x(n) is assumed to be a sample of a random
process, then the error in equation (2-40) is also a sample
of a random process. In the least square method, we minimize
the expected value of the square of the error.
2P
E[e^(n)] =E{[x(n) + E Avx(n-k)]z
} (2-41)k=l
K
2E[e (n) ] is minimized by setting
8E[e2(n)l . ^ 1
< . <(2 _ 42)
3A£
From (2-41) and (2-42) we obtain the set of equations
PI Ak BKx(n-k)x(n-i)] = -E [x (n)x (n-i) ] 1-i-P (2-43)
k=l
Then the minimum average error is given by
E = E[x 2(n)] + F AkE(x(n)x(n-k) ) (2-44)
k=l
For a stationary process x(n), we have
E[x(n-k)x(n-i)] = Rxx (i~k)
where P^x(i) is the autocorrelation of the process.
Note that equations (2-42) and (2-44) lead to the well known
orthogonality principle [7] . Since in the least squares
method, we assumed that the input is unknown, it doesn't make
much sense to determine a value for the gain G. However,
there are certain interesting points that can be made.
26
Equation (2-39) can be written as
x(n) = - ? A,x(n-k) + e(n) (2-45)k=l
K
Comparing (2-45) and (2-38) , it is seen that the only input
signal u(n) that will result in the signal x(n) as output is
that where Gu(n) = e(n). (2-46)
That is, the input signal is proportional to the error signal.
e(n) can be also considered as the modeling error. The error
variance can be calculated by Equation (2-41) and the filter
coefficient A, (k=l . . . p) would be calculated by equation
(2-43) if the correlation function of process x(n) is available
At this moment, it should be recalled that a linear random
process is generated by linear filtering of white noise. Therefore,
we are interested in white noise inputs for the purpose of
modeling a given linear random process. That is, the
input u(n) is assumed to be a sequence of uncorrelated samples
(white noise) with zero mean and unit variance .
E[u(n)] = 0, E[u(n)u(m)] = <5mm
Then the output x(n) forms a stationary random process
x(n) = - ? Avx(n-k) + Gu(n) (2-47)k=l
K
Multiplying equation (2-34) by x(n-i) and taking the expectation,
E[x(n)x(n-j) ] =- I AVE [x (n-k) x (n-i)
]
k=lK
+E[Gu(n)x(n-i)
]
(2-48)
Noting that u(n) and x(n-i) are uncorrelated for i>0 and re-
calling that for stationary process, E [x (n) x (n-i) ] =R (i)
,
s
Equation (2-35) turns out to be
PRxx
(i) = - 2 AkRxx (i-k)for p > i> o (2-49)
k=l
and R(o) can be obtained by plugging x(n) of Equation
(2-3 8) into E quation (2-48)
p2R
xx (o) = ~ l A&ik)+G (2-50)k=l
Therefore , the gain can be given by
°2 =R- <0)+ V A
k* xx <k> (2-51)
k=l
It is noted that through Equation (2-46), that is Gu(n)=e(n) /
the white noise input of zero mean and unit variance generates
the random process e(n) , which is again white with zero mean
2and variance of G . Therefore, from Equation (2-4 9) , the re-
cursive filter coefficients A, , (k=l, ....p) can be calculated
and using these values the gain G would be calculated by Equation
(2-51) with the knowledge of autocorrelation function of a given
class of linear random process. So far, modeling of one-
dimensional stochastic process has been considered. Similar
reasoning can readily be extended to the modeling of two-
dimensional random fields. Again, the two-dimensional all- pole
model is considered.
Mi M2x(k,&)=-£ z A..x(k-i,£-j)+Gu(k / £) (2-52)
i=o j =o J
(i,j)^(o,o)
28
Let's define the set ft(k,£) such that for all i,j
(k-i, PL-j ) en(k,£),
all the values of x(k,£) in Q,(k,i) will be used to estimate
the point x (k, £)
.
K-Ml
^2a » •
V
•
• 4 *
•
#
• » •
i• x(k,£-M )
4 •
• #
•J»
fi(k,£)
x(k,£)# #
• • • •
FIGURE 2-2
DEFINITION OF fi(k,£)
29
Again the coefficients A- • will be determined such that the
mean squared error is minimized. The estimate of x(k,&)
,
Ax(k«£), is given by a linear combination of the previous values
Ax(k,M = - Z
nEx(k-i f Jl-j) (2-53)
The mean squared error is
E[e 2 (k,£)]=E{[x(k,£)- x(k,£)]2
} (2-54)
If E[e (k r £)] is minimized, x(k,£) is the "linear
least squares estimate" of x(k,£).
Going through the same procedure as in the one-dimensional
case, that is, substituting (2-53) in (2-54) and differentiating
with respect to each A. ., setting derivatives equal to zero, we
obtain the following set of simultaneous equations for the
unknown A.
.
E([x(k; t)4(k,Jl)]x(i,j)} = o for allfij)efi, (2-55)
which says that the coefficients A. . must be such that the
estimation error [x (k,£) -X (k,£) ] is statistically orthogonal to
each x(i,j) that is used to form the linear estimate. This
is known as the orthogonality principle in linear least square
estimation.
The modeling error is the difference between the true
value x(k,£) and the estimate x(k,£). By definition,A
e(k,£)=x(k,£)-x(k,£)=Gu(k,£) (2-56)
from the equation (2-52) and (2-53)
.
Again, we are interested jntbevhite noise field of zero
mean andunit variance. Then the modeling error is also a
30
random field.
Rewriting the Equation (2-52) in terms of the error e(k,£) gives
x(k,£)=-rfiZ A
j_.x(k-i / £-j)+e(k,£) (2-57)
To calculate the error variance, multiply (2-57) by x(k-m,£-n)
and take the expectations
E[x(k,£)x(k-m,£-n) ]=- ZfiZA
i-E [x (k-i 7 £-j ) x (k-m f £-n)
]
+E[e(k,£)x(k-m,£-n) ] (2-58)
For the stationary process,
E[x(k-i,£-j)x(k-m,£-n) ] -F^lm-in-j ] , then (2-58) will be
RxxCm1n)=-E^ZA
i.Rxxfrn-if n-j) for allfm, n)efi. (2-59)
The second term on the rLghtside of (2-58) is zero, because of the
orthogonality principle and R(0,0) can be obtained by the
following equation:
Fj) 7 Q)=-Z QL A
ij Rxx(i,j)+G2
(2-60)
Therefore the modeling error variance,
E [e2 (k,£) ]=E{ [ (Gu(k,£)
]
2} = G 2 is given by the equatiom
G2=E[e 2 (k,£) ]=R^(D,0)+E I A
i-j^x ^'^ and the mean of error
e(k,£) is E[e(k,£) ]=E[Gu(k,£) ]=0
Again with the knowledge of autocorrelation function of the
two-dimensional stationary random field, the filter coefficient
A. . can be obtained from the Equation (2-59) and using these
values of A. • , the gain G in Equation (2-52) can be calculated
by Equation (2-60)
.
Example 1 Consider a one-dimensional stationary band limited
random process forwhichtheautocorrelation function is given by
31
R^x (m) = />,m| cos(w m)
Find a model which describes this process,
all-pole model is chosen such that
An
x(n) = - i A,x(n-k)+Gu(n)k=l
K
where E [e(n) ] =
E [u(n)u(m) ] = <Smn
One has to choose the order of the difference equation; here,asecond
order model (p=2) is chosen. Then
x(n)=-A1x(n-l) -A
2x (n-2) +Gu (n)
The problem shrinks down to that of finding A,, A2 / G with the
given R (m)
.
3 xx
From Equation (2-4 9)
k=l
R^l) = - A-^o) - A2^-l)
W ] " " Al^ 1} " A2%0)
Putting this in matrix form
-%D - m 0)
From Rxx(m) = Piml cos (w m)
o '
-1 - p COS w
- p COS w -1) \
A2 J
cos w<
P^ cos 2 w,
Therefore At n ,-i cO <> \1 =__ £ cox w„ (1-/°^ cox 2 wq)
n 2 21 - P COS Wo
32
A2
= P2 sin 2w
1 - p2cos 2 w
From Equation (2- 51)
G = R (0) +xx
? A,
k=l XX (k)
= 1 + A,R (1)+ A 9 P (2)1 XX ^ XX
= 1 + A,p cos w + A2P
cos 2 Wo
, _ p2cos 2wo (l-ft2cos 2 w°) p 4 sin 2w cos 2 w
1 -/
>2cos 2w 1 - p2cos 2 w
Example 2 Given a stationary two-dimensional random field with
Rxx(m,n) = p|m|p|n| ,
find the autoregressive model of this random field (most mono-
chromatic images can be assumed to have this form of auto-
correlation function). A first-order model (M = 1, M2
= 1) is
chosen. Then equation (2-52) can be written as follows:
x(k,A) = -7 A10x(k-1,£)-A11x(k-1 / £-l)-A01x(k,£" 1
) +Gu(k,£)
Where u(k,£) is white noise with zero mean and unit variance.-
Then, using Equation (2-59)
1 1
iym,n)=-i^Q £ Q
AijR
y|m-i,n-j) for all(m / n
) £ fi .
(i f j)^(o.O)
Rxx(l„0) = - A1Q
Rxx (0,0) - A
11Rxx
(0 -1) -A^Rxx (1 ,-1)
Rxx(l^l) = - A.. _R (0,1) - Ain R (0.0) -An1 Rxx(l# 0)10 xx 11 xx 01
Rxx(0 v l) = - A,.R (-1,1)- A...R (-l,0)-A n ,R (0, 0).10 xx 11 xx 01 xx
Putting thisin matrix form gives
-Rxx.(0,0) -Rxx(0,-1) -Rxx(l,-1)
-Rxx(0,-1) -Rxx(0, 0) -Rxx(l f 0)
-Rxx(-1 T 1) -Rxx(-l t 0) -Rxx(0 f 0)
N
10
11
L
0lJ
Rxx(l r 0);
Rxx(l T l)
Rxx(0»l)^
33
For thegiven auto correlation function above
-1-f> -f
z
-P -1 -7°
fi2 _
;/
o _! /
10
11
l
01
which yields
L
01
L
ll
L10
= ~P
= P:
The modeling error variance or the square of the gainG can be cal-
culated by Equation (2-60)
21 X
G =Rxx(°'°>
+i=
Z
Q ^Aij R
xx(i '^
(ir j)^(0,0
= 1 + A01
Rxx(0 ' 1)+ All
Rxx (1^ + ^O^x (1 ' 0)
= (1-p2
)
2
D I ml pt n I
Therefore, the complete model for Rxx = r f is
x(k,£) = px(k,£-l) + /3 x(k, £) + px(k-l,£) + (l-:f )u(k,£)
where E (u(k.£) ) =
Eiu(k.A) u(k-p,£-q)] = O.pq.
3 . Method of Filter Response
Another method of modeling a linear stationary random
process is based on the concept thatalinear random process is a
result of filtering white noise through a linear filter. In
Section B. of this chapter, the properties of linear systems
have been discussed.
34
For the discrete system, it is known that the filter output
power spectral density is the Z transformation of the output correla-
tion function. That is,
Gyy(Z) = 3 [R.yy
(m)]
Gyy(Zi, Z 2 ) = 3 [R yy(m,n)]
and also it is noted that
Gyy (Z) = Gxx(Z) H (Z) H (Z~ )
Gyy(Z , Z ) = Gxx(Z ,Z)H(Z,Z)H(Z , Z ),12 12 12 1 2'
where Gxx(Z), Gxx(Zi, Z2)is the input power spectral density
function and H(Z), H (Zi, Z2) is the transfer function of the
filter. Denoting the white noise input as u(n)oru(k,£) and
the output of filter, which is a linear stationary random pro-
cess we are going to model as x (n)or X(k,£)r
the problem
can be stated as follows: For a given autocorrelation function
ofalinear random process (field) R(m) , (R (m,n)), find a linear
system such that when the input is white noise, the output of the
filter gives a given autocorrelation function R(m) , (R(m,n)).
u(n)
u(k,*)
H(Z)
H(Z X , z 2 )
x(n)—+-
FIGURE 2-3
THE MODELING PROBLEM IN THE DISCRETE CASE
x(k,4)
Given ^(m), Rxx (m ' n)
35
If the input is white noise, then
GUU (Z) = Const,
Guu (Zl'
Z2 }
= Const.
Therefore, the solution for the required filter is to find a
function H(Z) or H(Z-,, Zo) which satisfies
G (Z) = Const-H(Z)H(Z_1
)X^V
GXX (Z1# Z2
) = Const-H^, Z2)H (Z£
!
, Z2
_1) (2-62)
where G„„(Z), G (Z, , Z ) is known through the relationXX XX X ^
Gxx< Z) = 3 I Rxx<m >l
Gxx< Zl'
Z 2> =3[Rxx<m ' n 'l
since R (m) , R (m,n) is given.xx xx 3
Butforthe two-dimensional case, there is an inherent difficulty
in factorization of G (Zj, Z2
) to H(Z-,, Z2)H(z7 , Z
2~
).
Therefore, only separable functions can be modeled by this
technique.
Example 1 For a given stationary linear random process with
R (n) = S 2p,mI cos(w m) , m=o, 1, 2 . . .
Find a difference equation which will give a random process
with autocorrelation above.
G (Z) = MR (m) ]xx ^ xx
_ g* 2 (l-|°2
) . [-Z /°cos w +(l4-^2 )-Z~
1cox w Q ] (2-51)
(l-2f>Zcos w +p2Z2
) (l-2/
PZ~ 1cos w +f2z~
2)
The second step is to find a factored expression for G (Z) ,i.e.
Gxx Z) = G1(Z)
* Gl^ 2" 1
)
36
Assuming that G (Z) has the form
GXX (Z) = 3 2 (l-p2
) az + b az + b (2-52)
l-2/°Z cos w +/^2z~
2l-2pz cos w +p
2z
Comparing (2-52) and (2-51) , a and b can be obtained as
a = j ( 1-p cos w +p2 + l+y° cos w +p
1 2b = j 1-p cos Wo+p 1+ P COS W +jO
From equation (2-50), if we choose H(Z)= az 1 + b
l-2pz'1cos w +p
2z~
2
2 2then the term & (l-j° ) in Equation (2-52) can be considered
as G (Z)uu
Therefore
(1-p2
) d2
if m=0
Ruu (m)
if m^O
and the complete model is drawn block diagram form in Figure 2-4.
u(n)
Hm _ az +b
x(n) has givenautocorrelation
White Noisewithvariance= (1-/32) 2
l-2pz cos wo +p
2 z-2
x(z)
FIGURE 2-4
FILTER FOR ONE-DIMENSIONAL BAND PASS PROCESS
37
lb put the system's input-output relation into state equation
form, it is defined for convenience as
u(z) = z~ w (z)
where w(n) is also white noise with same autocorrelation
function
R(m) = (1-P2)tf
2if m=0
WW '
if m^O
then the transfer function is
X(Z)H(Z) =
D ef ining
Xx(z)
az 1 + bU(Z)
-p2z
1X2
(z)
-1 2 -2l-2pZ cos w +p z
X2(z) U(z)
-1 2 -21-2'P cos woz +p z
X(Z)-1
= (az +b)X2(z)
then
and
Xx(n) = -p x
2(n-l)
X2(n) = x
1(n-l) + 2p cos w x
2(n-l) + w(n-l)
X(n) = ax2(n-l)+bx
2(n) = -P ax(n)+bx
2(n)
Putting the state and output equations in matrix form gives
( i \\ ( „ ~2 V . . .> f - \x-^n)
x9 (n)
- p
1 2 cosw
-2
x1(n-l)
>
i x2(n-1
w(n-l)
x(n) = (-p a b) x-^n)
x (n)
Example 2 The two-dimensional "band limited" discrete
Markov process is defined by the autocorrelation function
t, / \ ,2 r , (m) , \ir^(m ) / \i m=0.1.2...R^On,!!) -<*•
[frv cos (w
fm)] [p 2
cos(w2n)] nmQ
'
fl'
t2mmm then
the discrete power spectral density
Gxx (zl'
Z 2> =6J2 (1-Pi2)d-P2
2)
A(Z1 / 2
2}
B(ZX
, Z2 )
where
2-1 2-1A(Z
X, Z
2) = [-Z
1Cosw
1+(l+y0
1)-Z
1CosW-jJ [-Z
2CosW
2+(l+j?
2)-Z
2Cos w
2]
B(?l'
z2
)= [(1" 2 r
3
izicos w
i+/
]
izi
) d-2p1 zj[1cos w
1+p2z^
2)]
[(l-2^2z2cos w2+^
2z2
) (l-^z^cos w2+(G2 z
2
2)]
Putting A(Z,, Z2
) in the following form
A(ZX
, Z2
) = [(alfz^
1+b1
) (a2z2+b
1) ] [(a
2z2
1+b
2) (a
2z2+b
2) ]
and comparing this equation and above A(Z,Z~)t a,, a2 , b, , b
2
obtains
al j( Jl-p^os w
1+
p1+ V 1 + p-|_cos v^+p^)
a2=7 ( 7 3 "P2cos w
2+Pi
+ V : +f°2
cos w2+l°2
}
|( Jl-p^cos wx
+p x
2- yi +
(
o1cos w1+px
2)
b2=
* (n/
1^003 w2
+,
p 2
2- v/
X +f3
2cos W
2+P2
2)
Let H(Z1
, Z2
) = (a1z~1+b
J) (a
2z~ 1+b
2)
(l-2^1z~ 1cos w
1+/°2z
1
2) d-2/0
2z2
1cos W2+P2
Z2
2
39
then from (2-50)
R (m,n) =uu
f
O)2(l-P1
2) d-P2
2) if n=o, m=0
o if n^O, m^O
u(z 1# z2
)
White Noiseof variance=^2 (l-p1
2)(l-f>
H(Z1;Z2 ) —>-
FL<(m,n)
^'"cos^mXpJwSTiJ.n)
FIGURE 2-5
FILTER TO GENERATE BAND PASS RANDOM FIELD
It is convenient to define
.
U(ZX , Z
2) = Z
1
-1Z2
" 1W(z 1/ z2 )
W(k,£) is also white noise with the same statistics as w(k,£)
Rww (n 'm)=\ ^(l-f^
2) d-f2
2) if n=0 and m=0
if n^O or m^O
Then, the filter has the form
X(Z lf Z2
) = (a1z~ 1+b) ( a
2z2
' 1+ k>2
) Z1
~ 1Z2
~ 1
w(z lf z2 )
(l-2piz~ 1cos
-2 -1Wl+Pl
Zl ^
^ 1 ~ 2P2
Z2
COS W ->"1"P-> Z '> ^
2-2;2T2 Z
2
The following definitions are made:
40
N1(Zlf Z
2)
N2(Z ir Z
2)
-piWv z2
'
Zo-lw(Z , Z )
2 l 2
cos w, +2—^2"
1-2^.2:, """"cos w, + 3, ~,
(a^ 1 + b1)N
2(Z
1 , Z2 )1*1
From the last three definitions one can write the set of
difference equations
\N-^k,*.)
N 9 (k,£)
/
\
\
-Pi'
N1(k /
£-l)
1 If.cos w1
N2(k,£-1)
/SN
1
w(k,£-l)
x-^k,^ = -^>, a1N1 (k,£)+b1N2 (k,il)
Now, additional definitions are made.
M1(Z
1' V = -PIZ2~\ {Z
1'Z2
}
M2(Z
1 , Z2
) =Zl
Xl
(Z 1' Z2 }
l-2^2z2
1 cos w2+/>
2
2z2
-2
From these definitions it follows that
-1X(Zlf Z
2) = (a
2z2
+ b2
) M2
(Z 1# Z2
)
(2 _ 55)
Then
rMx(k,£)
, M (k,£)
(-f/
1 2f> COS w2 J
M^k'-l,*)
M (k-l,Jl)
' (P
x1(k-l,£)
(2-56)X(k,£) = -/
o2^a
2M1(k / I) + b
2M2(k,£)
Combining these all together , the following form is obtained
41
^(k+1,4)^
M2 (k+1,*)
N1(k,£+1)
N2(k / £+l)J
1
lo
"Pi
2p2cos w2
-p,a, b.
-Pi
2plCos^
M^kffcf
M2(k,£)
N1(k,£)
N2(k,£)
f
'"1
UJ
w(k,£)
and
(k,fc) = [-p2
a2
b ]
r
M1(k,4)
M2(k,£)
N^kfJl)
N2 (k,£)J
TVo methodsof modeling have been discussed. Some of the above
examples will be used in later chapters. It should be noted
again that the filter response method cannot be used for the
case where the autocorrelation function is nonseparable.
42
III. ADAPTIVE FILTERS
A. NONRECURSIVE FILTER
1. Introduction
Many forms of adaptive filters have been described in
the literature, some of which have been shown to be optimal
(or suboptimal) in certain applications. The special form
of an adaptive nonrecursive filter developed by Widrow [1]
is reviewed here to give some insights to the recursive adap-
tive filter developed in next section.
The filter to be considered here consists of a tapped
delay line, variable weights whose input signals are the signals
at the delay line taps, a summer to add the weighted signals,
and machinery to adjust the weights automatically. The impulse
response of such a discrete system is completely controlled
by the weights. The adaptation process automatically seeks an
optimal filter impulse response by adjusting the weights.
Two kinds of processes take place in the adaptive filter:
training and operating. The training (adaptation) process is
concerned with adjusting the weights, and the operating process
consists in forming output signals by weighting the delay line
tap signals, using the weights resulting from the training
process. During the training process, an additional input
signal, "the reference (or desired) response," must be supplied
to the adaptive filter along with the usual input signals.
43
This requirement may in some case restrict the use of
this particular form of adaptive filter. An example illustrat-
ing the use of the desired signal is the case of modelling an
unknown system by a discrete adaptive filter as shown in Figure
3-1. Here a discrete input signal x(n) is applied to an un-
known system to be modeled. The discrete adaptive model is
supplied with an input x(n). The output of the unknown system
d(n) is compared with the output y(n) of the adaptive system.
This system can self-adapt to minimize the mean square error,
(throughout this thesis, the mean square error is chosen
as the performance measure), where the error is defined as the
difference between the output of the adaptive model and the
desired signal (for this problem the desired signal is the output
of the unknown system to be modeled)
.
GHHadjustableweights
y(n) FilterfcOutput^
Input
Unknown System
To be modeled
systemoutput
FIGURE 3-1 MODELLING OF UNKNOWN SYSTEM
44
then Ny(n) =
i| w±x (n-i) (3-1)
e(n) = y(n) - d(n) (3-2)
Noting that equation (3-1) is the convolution summation, the
sequence of weights (wi) can be seen asthe impulse response
of the adaptive system.
2The weights w- are adjusted to minimize E(e ) .
It will be shown that if the input and output signals of the
system being modeled are stationary, the error signal has a
mean square value which is a quadratic function of the weight
settings
.
For the minimization of mean square error, the steepest
descent method is used. Throughout this thesis , the terms
"filter coefficient updating process" and "adaptation process" are
used interchangeably and it is assumed that the input to the
adaptive system and the desired signal are stationary random
processes (or random fields for the two-dimensional case) .
2 . Performance Surface, Gradient and the Wiener Solution .
The input signals are weighted and summed to form an
output signal [Equation (3-1) ] . Introducing the vector notation
such that w( n )
=^w '
wl '
w2 * * * W
N^
and X* = [x(n), x(n-l), x(n-2) . . . x(n-*l)] (3-3)
45
Then E quation (3-1), which describes the linear combination
(operating process), can be written in matrix form.
y(n) = "^x = x% (3-4)
and e(n) = y(n) - d(n)
= WTx - d(n)
T.he square of this error is
e2(n) = TFx"x%~ - 2d(n) t\T
Tx + d 2(n)
(3-5)
(3-6)
the mean square error, the expected value of e (n) , is
2,
_
% , -2,_ x ^T«. ^T w (3-7)E[e (n)] = d*(n) - 2W^
d + W- Rxx
where the vector of cross-correlation between the input signals
and the desired response is defined as
E[d(n) X (n) ] = E|d(n)x(n)
d(n)x(n-l) xd
d(n)x(n-N) (3-8)
and where the correlation matrix of the input signal is defined
as
E[X(n)XT
(n) ] = E X(n)X(n) X(n)X(n-l) . . . X(n)X(n-N)
X(n-l)x(n)X(n-l)X(n-l) . . .X(n-2)X (n-N)
X(n-N)X(n-N)
A= R.xx
(3-9)
46
It may be observed from (3-7) that for stationary input signals,
the mean-square error is precisely a second order function of
the weights. The mean square error performance function may be
visualized as a bowl shaped surface, a quadratic function of the
weight variables. Then the adaptive process has the job of con-
tinually seeking the "bottom of the bowl." A means of accomplish-
ing this by the well-known method of steepest descent is dis-
cussed below.
In the non- stationary case, the bottom of the bowl may be
moving, and the orientation and curvature of the bowl may be
changing. The adaptive process has to track the bottom of the
bowl when inputs are nonstationary . The method of steepest
descent uses the gradient of the performance surface in seeking its
minimum. The gradient at any point on the performance surface
may be obtained by differentiating the mean-square error function
of Equation (3-5) with respect to the weight vector. The
gradient is
V{E(e 2(n))}= "2 Rxd + 2 Rxx W (3-10)
To find the "optimal" weight vector WTMq , .i.e. the one that yields the
least mean square error, set the gradient to zero. Accordingly,
Rxd= Rxx WLMS
W = R_1
R (3_11)LMS Rxx Rxd
Equation (3-11) is known as the Wiener-Hopf equation in
matrix form.
47
Then the minimum mean square error may be obtained by
substituting (3-11) into (3-7)
E [e2(n)] min = a"
2(n) - w£
MS Rxd (3-12)
3. LMS Algorithm
In seeking the minimum mean-square error by the
method of steepest descent , one begins with an initial
guess as to where the minimum point of the mean-square error
surface may be. This means that one begins with a set of initial
conditions for the weights. The gradient vector is then mea-
sured, and the next guess is obtained from the present guess
by making a change in the weight vector in the direction of the
negative of the gradient vector. The method of steepest descent
can thus be described by the following relation
W(n+1) = W(n) + kV[E(e 2(n))] (3-13)
2The expression for V[E(e (n) )] is obtained by using
Equation (3-10)
.
W(n+1) = W(n) + 2kRxxW - 2k Rxd (3-14)
2The gradient vector V[E(e (n))] is the gradient of the expectation
of the squared error function when the weight vector is W(n)
.
When the performance function is quadratic, the gradient
is a linear function of the weights. The advantage of working
with the quadratic performance surface lies both in this linear
relation and in the fact that such a surface has a unique-
minimum .
48
The purpose of the adaptation process is to find an exact or
an approximate solution to the Wiener -Hoff equation (3-11)
.
One way of finding the optimum weight vector is simply to solve
(3-10) . Although this solution is generally straight forward,
it could present serious computational problems when the num-
ber of weights N is large and when input data rates are high.
In addition to the necessity of inverting an H x N matrix, this
method may require as many as j\l(N+l)/2 autocorrelation and cross
correlation measurements to be made to obtain the elements of
^x' Rxd*
No perfect solution of equation (3-11) is possible in prac-
tice to estimate perfectly the elements of the correlation
matrices.
A method for finding approximation solutions to (3-11) is
presented below. The accuracy of this method is limited by
statistical sample size, since weight values are found that
are based on finite-time measurements of input-data signals.
This method does not- require explicit measurements of
correlation functions, nor does it require matrix inversion.
It is the "LMS" algorithm based on the steepest descent method.
This algorithm does not even require squaring, averaging, or
differentiation in order to make use of gradients of mean-
square error functions.
When using the LMS algorithm, changes in the weight vector
are made along the direction of the estimated gradient vector.
49
Accordingly
_ Ar 2
W(n+1) = W(n) + kV{ [E(e^(h) )]
}
(3-13)
Where
W(n) = weight vector before adaptation
W(n+1) = weight vector after adaptation
k = scalar constant controlling rate of convergence
A 2,_,„ ^ ,. c „ r _2
and stability (k<o)
V [E (e^ (n) )] = estimate of gradient of Ete^Cn)] with respect
to W with W = W(n)
One method for obtaining the estimated gradient of the
mean square error function is to take the gradient of a single
time sample of the squared error; that is
V[E(e 2(n)] = V[e 2
(n)] = 2e(n) V[e(n)] (3-14)
From Equation (3-4)
V[e(n)] = V[y(n)-d(n)] =V [WT(n) X (n) -d (n)
]
= X(n) (3-15)
Thus,A 2V[E(e z
(n))] = 2e(n)X(n)
= 2[WT(n)X(n)-d(n)]X(n) (3-16)
The gradient estimate of (3-16) is unbiased, as will be shown
by the following argument: For a given weight vector W(n) , the
expected value of the gradient estimate is
:
E{V [E (e2(n))]}= 2E{ [W
T(n) X(n) -d (n) ] X (n)
}
" 2V" 2Rxd < 3 - 17 >
50
Comparing (3-17) and (3-10) , we see that
E{ V[E(e 2(n))] } = V E [e
2(n)
]
(3-18)
and therefore for a given weight vector, the gradient estimateA 2V[E(e (n) ) ] is unbiased.
Using the gradient estimation formula (3-16) , the weight
iteration rule Equation (3-13) becomes
W(n+1) = W(n) + 2k e(n) X (n) (3-19)
and the next weight vector is obtained by adding to the present
weight vector scaled by the value of error. This is the LMS
algorithm. Looking at Equation (3-19) , the adaptation process
is a simple first order recursion equation which can be realized
as shown below.
X(n)
2kX(n)W(n+1) UNIT
DELAY
"W7-JWEIGHTSETTING
e(n)
FIGURE - 3-2 FILTER COEFFICIENT UPDATING
51
Tothis point, the basic concept of LMS nonrecursive adaptive filter
has been introduced and reviewed. More details can be seen in [1] .
Widrow [1] showed that the weight vector mean converges to the
Wiener solution aidthatthe boundsof the step size k should be in the
region such that
1
Amax< k < o for the stability and convergence,
where X is the maximum eigenvalue of Rmax xx
4 . Two-dimensional adaptive filter .
a. Adaptive filter structure
The input-output relation of the two-dimensional
filter is given by two-dimensional convolution.
P *3
y(k,Jt) l E • wi
. x(k-i, £-j), (3-20)i=o j=o J
where y (k, I) is the filter output
and W. . is the fiiiteimpulse response of filter.
Here, it is assumed that the input. X(k,£) is a stationary random
field.
In Equation (3-20) , a set of two-dimensional
stationary input signals is weighted and summed to form an out-
put signal and the filter output is intended to match a desired
(reference) signal in accordance with the minimization of mean
squared error, where the error is the difference between filter
output and desired signal.
Ae(k,£) = y(k,£ ) - d(k,£). (3-21)
52
Introducing the vector notation such that
and
oo ol
WT
% W10
Wll -• W
lqW20
,W ]
pq
X ± = [x(k,JD, x(k,£-l), . . .x(k,i-q)x(k-l,A)x(k-l f A-l) . . .x(k-l,£-q)
x(k-2,£) x(k-p,£ -q)] (3-22)
then Equation (3-20) can be written in matrix form.
y(k,£) = ^Fx = X^W (3-23)
where W is a weight vector of dimension (p+1) (q+l)xl
X is a input signal vector of dimension (p+1) (q+l)xl
The weight vector of the filter is supposed to be adjusted in
the direction such that performance criterion (mean square error)
is to be minimized. Thus, the linear combinatorial system
in Equation (3-20) will be given with variable weights.
x(k,l)
yOt.DHCOM 3-3 LIHEAB COMBINATORIAI- -SYSTEM itf
TWO-DIMENSIONAL ADAPTIVE flLTER
53
In Figure 3-3, the linear combinatorial structure is given. Then,
the adaptive nonrecursive filter can be drawn as following:
r
x(k,*)
LLINEAR COMBINATORIAL
OF FIGURE 3-3
Input IWITH VARIABLE WEIGHTStationary f
RandomInput 7
y(k,£)filter output
3—
V
e(k,£)
FIGURE 3-4 STRUCTURE OF NONRECURSIVE ADAPTIVEFILTER
b. Wiener solution
From equation (3-22) and (3-23), the error signal
can be written by
e(k,Jt) = W^X - d(k,£)
The square of this error is
(3-24)
e2(k,£) = WTXXTW - 2d(k,£) XTW + d
2(k,£) (3-25)
E [e2(k, £) ] =E [d
2(k, £) ] -SR^W^+W^R^W
The expected value of e (k,£) is
(3-26)
where the vector of cross correlations between the input
signal and desired response is defined as
54
'd(k,£)x(k, £)
d(k, £)x(k, £-1)
E[d(k, £)X] = E
d(k, £)x(k, £-q)
d(k, £)x(k-l, £)
d(k,£)x(k-l,£-q)
V d(k,£)x(k-p, £-q)
=ARxd (3-28)
and where the correlation matrix of the input signals is
defined as
E[X3T] =
/xlxl
x l*2 x f3 xlX (p+q) (q+D
X2X 1X2X2
X2X3
X2X (P+D (q+D
x 3'xl X3X2
X3X3 *3 x(p+D (q+D
>
(3-29)
*P+1 ) q+l)Xtl^p+l) q+2)
x2
X(.p tj) q+])
x(pf]} q+|
=RXX , where x(p4])(q+
^x(k-p / £-q)
55
The gradient at any point on the performance surface can be
obtained by differentiating the mean square error function of
equation (6)
V[E(e 2(k,*) )] = 3 E(e
2(k,t)
3 w
"2Rxd+ 2Rxx™
To find the "optimal" weight vector W MS that yield the least
mean square error, set the gradient to zero. Accordingly,
Rxd " VWLMS Rxx
Rxd (3-30)
Equation (7) is the Wiener Hopf equation in matrix form, again
the minimum mean square error is obtained by substituting
(3-30) into (3-26)
.
E[e 2(k,£)] min
= E[d2(k,£)] - W
LMST R
xd (3-31)
c. LMS Algorithm
Consider a two-dimensional field x(k,£) to be processed
(usually two-dimensional filters are used in process-
ingdiscrete two-dimensional image fields) and assume that the
two-dimensional field consists of NxN discrete points (which may be a
sensed signal by NxN pixel elements of a sensor) . The adaptation
processes is that of adjusting the filter coefficient W- • in
56
accordance of minimization of the mean square error as in the one-
dimensional case. The adaptation scheme may be predetermined as
being columnwise scanning or diagonal, or row-wise scanning.
Here the row scanning process is adopted as shown in Figure 3-5.
p • • § m
li •
o •
k-p •*»
Region for jfc^
adaptation.y(k,A) =
p qI Ew
ia(fc-ii*-j)fko o
e(k,i) -
e(1) \ • •
• • •
• • • •
e • •
• •
Region for(j+l)
adaptation
^n
• < »l »
.th
*^-(j+l) th estimation point
estimation point
* •
FIGURE 3-5 ADAPTATION SCHEME
57
Therefore ,theNxNadaptation processes will be required to complete
the filtering of anNxN two-dimensional field. Using e(k,£)
to denote ' the error at the jth iteration (then e(k+l,£) is
the (j+l)th error), then the filter coefficient updating process
can be described by
W(j+1) = W(j) + uV[E(e2(k,£)]
where W(j+1) = coefficient vector after adaptation
W (j) = coefficient vector before adaptation
u = negative scalar constant controlling rate
of convergence and stability.
The gradient of tie mean square error is to be estimated by
V[E(e 2(k,£)] = V[e 2
(k,£)]
where e(k,£) = y(k,£) - d(k,£)
then W(j+1) = W(j) - ue (k, I) V [e(k, I)
]
= W(j) - 2ue(k,£)X
where X is defined by equation (3-22)
.
Along with y(k,l) = W^and e(k.£) = y(k,£) - d{k,l) ,
the LMS algorithm will be completed.
B. RECURSIVE FILTER
1. Introduction
In the previous section, it is shown that adaptive non-
recursive filters have a finite impulse response; that is, they
can produce only zeros with no poles in the filter transfer
function. This limits the capability of transversal adaptive
58
filters in many applications. To overcome this limitation a
new adaptive filter structure is described which is capable of
producing poles in the transfer function. The basic configura-
tion considered here is quite standard; that is, the present
output sample of the filter y(n) is a linear combination of the
present and past samples of the input x(n), x(n-l), . .x(n-M)
and the past samples of the output, y(n-l), y(n-2). . . y(n-N).
The present output sample of the filter is compared against a
reference sample. The resulting error samples are used to adjust
the filter parameters, feed forward gains and feed back gains to
minimize some error function. TheQie-dimensional recursive filter
is developed first, then it is extended to the two-dimensional
adaptive filter.
Recently Feintuck [2] and White [3] have proposed a technique
for making digital filters with zeros and poles adaptive. This
development may enhance the possiblity of obtaining accurate
models for unknown systems. The new approach is developed into an
algorithm. It employs the steepest-descent criterion for para-
meter adjustment but it differs in the estimation of mean
squared error gradient vector from Feintuck [2] and Widrow [1]
.
2 . One-Dimensional Adaptive Recursive Filter
a. Structure
The recursive filter is described by its transfer
function
59
M
Y(Z) i=oZ
bi
Z
X(Z)1 + I aL Z
i=l (3-32)
In the time domain, the input-output relation of the digital
filter is given by
M Ny(n) = I b
±x(n-i)- I a.y(n-i) (3-33)
i=o i=l
where y(n) = nth sample of the filter output
x(n) = nth sample of the filter input
a. = feedback coefficients i = 1/2 f . N
b. = feed forward coefficients i=0,l,2..N
The output samples of the filter are intended to match those of
a reference (or desired) signal d(n). in accordance with
the minimization of some error criterion, the filter parameters
a., b. will be adjusted at every iteration. Thegsneral scheme of the
adaptive recursive structure is given in Figure 3-6. The two
finite length transversal filters areusedinthe forward path and
feedback path to form the recursive filter of Equation (3-33).
60
d(n)
reference signal
Input Signal
x(n)
FIGURE 3-6 ADAPTIVE RECURSIVE LMS FILTER
USING TWO TRANSVERSAL ADAPTIVE
FILTERS
b. Problem in Wiener Solution
Introducing vector notation for the signals and sets
of filter coefficients we have
*-TA - [ a^, a.^, . . . . aN]
ttTB =
[ bo' b
l bM l
X(n) = [ x(n), x(n-l) . ... x(n-M)]
Y(n) T= [ y(n-l) f y(n-2) . . . y(n-N)]
61
Then Equation (3-33) can be written as
y(n) = B^Xdi) -AT
Y(n) (3-34)
where A is the feedback coefficient vector (nxl)
Bis the feed forward coefficient vector [ (M+l) xl]
X (n)is the iiput signal vector at nth iteration [(M+lxl]
Y (n)isths output signal vector at nth iteration (Nxl)
The performance criterion is again minimum mean squared error,
where tte error is the difference between filter output and desired
signal (reference signal)
.
That is, the filter is used to estimate a desired waveform
d(n) in a minimum mean square error sense. Assume that the
observables are stationary and zero mean and let e(n) denote the
error waveform at nth sample, then
e(n) A y(n)-d(n) (3-35)
= BTX(n)-A
TY(n) -d(n)
and the mean square error is
E[e2(n)] = E[(B r x(n)-ATY(n)-d(n) )
2]
= E [BTX (n) X(n)
TB-2BT X(n) Y^n) A+ATY(n)
Y
TA
-2BTd(n)X(n)+2ATd(n)Y(n)+d2(n)
]
= E [d2(n) ] +B
TRxxB+A
TRyyA-2BrRxyA
" 2lTRdx
+2lTRdy < 3 " 36)
where R^ = E[X(n)XT(n)]
Ryy
= E[Y(n)YT (n)]
Rdx= E td ( n )^( n )l
Rdy = E[d(n)Y(n)]
and ^ = E[X(n)YT(n)]
62
The theory of Wiener filtering employs known second-order input
statistics to dictate the impulse response of the linear filter
that minimizes the mean square error; that is, as in the previous
section, the knowledge of second order statistics R, , R is
assumed to calculate the optimum impulse response
(optimum weight vector in nonrecursive adaptive filter) Wn
But here in the recursive algorithm, it is also required that
the autocorrelation of the output, R , and the cross correlation
of the output and the input, R , and the cross correlation of the
output with the desired waveform, R, , should be assumed known.
Thus, the set of statistics mentioned above is assumed to be known
for a moment, and will be used to determine the weights in
the recursive filter. The statistics for the fixed para-
meter filter are not a function of these statistics, but instead
the weights are a function of these statistics. Therefore, R„..,xy
R, and R are to be considered constant when the differentiationax yy
is made with respect to A and B.
The set of weights (filter coefficient vectors) which minimize
the mean square error can be found by getting the gradient vector
with respect to filter parameters equal to zero.
9E(e2(n)]
9 A= 2R A - 2 R 1 + 2 R, =0
yy xy dy
R~ 1
(R, - R "B) (3-37)yy dy xy
and
9 E[e2(n)] A Vr^/ 2
9 B 3[E(e (n))]
= 2R 1 - 2R A - 2 R,xx xy dx
B *xx-l ( Rdx +*xy
X) (3 " 38)
Thus, one can solve for the filter coefficients if all the
second order statistics are known. But without knowing the
impulse response of the filter, the R , R, and R can not ber e xy dy yy
calculated with only input and reference signal statistics.
Noting that we are looking for the impulse response which
minimizes the mean square error in some way, it is clear that
R , Rd , R are not available and so the wiener approach is
not feasible.
C. LMS ALGORITHM.
An iterative gradient search technique (the method of steepest
descent) is revisited. Here, in the recursive algorithm, it
updates the filter coefficients with steps proportional to the
gradient vector. This updating process is
A(n+1) = A(n) + k AA3.
= A(n) + k_VA[E(e
2(n))] (3-39)
B(n+1) = B(n) + kbAB
= B(n) + kbvB [E(e
2(n)]
where
A(n) , B(n) = filter coefficient vectors before adaptation
A(n+1) B(n+1) = filter coefficient vectois after adaptation
k , k, = scalar constants controlling rate of convergence
and stability (k , k\<o)2 a b
gradie
to A and B respectively.
VA [E(e (n) ) ] , V [E(e (n) ) ] = gradient vectors with respect
64
The updating process (3-39) can be considered as afirst order fil-
tering process with an input proportional to the gradient vector.
2 2But the gradients VA.[E (e (n) ) ] and V B [E(e (n) ) ] should be esti-
mated because the output statistics are not available apriori or
an infinite statistical sample would be required to estimate
perfectly the elements of the correlation matrices in Equations
(3-37) and (3-38) . A method of estimating these gradients will
be presented.
Widrow [1] obtained the estimated gradient of the mean square
error function by taking the gradient of a single time sample of
the squared error (instantaneous estimates) when he discussed
the nonrecursive adaptive filter (see previous section)
.
Here, in this thesis work, a new method of estimating
gradients is proposed. This is to approximate the mean squared
2error E(e (n) ] by an average of a finite number of points at every iteration
and take the gradient of this instead of taking the instantaneous
error square, that is, the approximation used is
9 1 L-lE(eMn))~ — I e (n-£) (3-40)
L£ =
2For E(e (n)), the average of the square error for the previous L points
is taken and then gradient is evaluated for the approximate mean
square error. The estimated gradient of mean square error is
V.E[e2(n)] =V,[
Lf1e2(n-£)]A A
£ =
VRE[e2(n)] =V R [
L: 1 e2(n-£)] (3-41)B B
£^0
65
For convenience,the£l) term in (3-4 0) was dropped,
vector notation for error signal
e(n) =f e(n)
e(n-l)
Introducing the
[ e(n-L+l)
then it is seen that the error signal vector is an (Lxl) vector. The
estimated gradient (Equation (3-41) ) can be put into the matrix form
BLl ~ x "' j v
b
Substituting the estimated gradients in Equation (3-39) , the updating
process for the filter coefficients is:
^aVA
VAE[e2(n)] = VA [ e
T(n) J (n)
]
VRE[e2(n)] = V n [ e
T(n) e(n)]
A(n+1) = A(n) + K a V, [£T (n)e(n)]
and
B(n+1) =B~(n) + KbV B [e
T(n)e (n) ] (3-42)
The function e (n) e(n) is a scalar function of the coefficient
vectors A and B, that is,
TT (n)T(n) = f(A, B)
Therefore , by definition, the gradient of f (A, B) with respect
to A and B is
66
21
3A
3f"5"a.
3f3a.
3f
3f3a
N
3B
3fTO
o
3f3b.
3f3bM
It follows that
5A [E(e
2 (n)H = VA [FT(n)i(n)]
*
d (t^ (n)I (n) )—3~a~:
3(FT (n)e(n))3a.
3(£T(n)e(n) ) I
I3aN J
/
27(5) _L*Mal
T 3e (n)
2£ (n)3a.
27(5)9e(n)^a
(3-43)
N /
and
VB (E(e2(n) )) = V
B(£(n)
T£(n) )
3(£(n)T
£ (n)
3b„
3(£(n)T
£(n)3b,
3(£(n)T£(n)
3 bN
r
n)
o2 F(nT 2£^gl
2 F(n)T 9e(n)
3b-
2e(n)T 3e(n)
y v3bN
(3-44)
67
Consider the terms 3e (n) , 3e (n) in equation (3-43) , (3-44) ,
8ar,
9b~p q
where p=l,2,...N and q = 0,1,2,...M.
Since e (n) and e(n) are defined as
e(n) A y(n) - d(n)
— Te (n) = (e(n), e(n-l), . . . e(n-L+l),
3e(n)7a
/3 e(n)~~51
P
3e(n-l)3a
3e(n-L+l)3~£
3y(n)3a
P
3y(n-l)3a_
3y(n-L+l)3a
p=l, 2 , . . .N (3-45)
and
3gpl) _
3ba
r 3e(n)~T5q
3e(n+l)3 bq
3e(n-L+l)3 bq
3y(n)3bq
3y(n-l)3bq
•
3y(n-L+l)
I 3bq
q=0,l,2,...M (3-46)
Note that3e(n) and 3e;(n) are (Lxl) vectors3ap 3bq
and e (n) is an (lxL) vector.
68
Therefore, V^[E(e 2(n))] and V
B[E(e
2(n) )] in Equation (3-41)
are (Nxl) and [(M+l)xl] vectors respectively.
Equations (3-45) and (3-46) may be considered as sensitivity
vectors which tell how much the change in clp and bq affect
the outputs y(n), y(n-l) . . . y(n-L+l)
.
To calculate the elements ofthe estimated gradients of mean square
error in equation (3-41) , we should calculate the sensitivity
vector of equation (3-45) and (3-46) first.
From the recursive equation (3-33)
,
9y(n)a
, M N,
= — I b.X(n-i) - I A.y(n-i)3ap 32lp l i=o
1i=l
X I
= - y(n-p) - I A.ay(n- i}
(3-47)i=i
13 a P
p = 1/ 2, . . . N
lif- - *<»-<> -A^^q = 0, 1, 2, ... M (3-48)
The sensitivity vector components given by Equations (3-47) and
(3-48) can be interpreted as being the response of a linear system
with transfer function.
1 (3-49)H(Z) =
l+a,zX
+ a_ z"2+ . . . aMz-
N
Henceforth fthis will be called. the "sensitivity filter." Equation
(3-49) is ai all pole filter (recursive filter) with input signals
[-y(n-p)] and [x(n-q)] respectively.
69
Now, what are the initial conditions characterizing the re-
cursive relationships of the sensitivity filter?
x(n-q) is the q time units delayed signal of input x(n) to the
adaptive recursive filter of Equation (3-33) and y(n-p) is the p
time units delayed signal of 1he output y(n) of the recursive
filter.
From x(n) = o
y(n) = for n< o ,
x(n-q) and y(n-p) are sequences with the first q elanents and first p
elements zero respectively.And since changes in the ap and bq coeffici-
ents have no effects on the system's response until n=p and n=q
respectively, it follows that the initial conditions are:
gip - o for n=o, 1 ... p-1
a^n) = o for n=o, 1 ... q-1
A summary of this algorithm is
the following
:
1. Calculate the sensitivity vector components through the
sensitivity recursive filter by equations (3-47) and (3-48)
.
2. Calculate the estimated gradient by equation (3-43) and (3-44).
3. Calculate the filter coefficient vector by equation (3-42).
4. Calculate the filter output by equation (3-34)
5. Form the e(n) vector i~(n) , then go back to the 1st step.
Note that due to the fact that the gradient of finite point square
error average is used for the estimation of the true gradient of
mean square, this filter cannot give an optimal solution, but the
70
more averaging points are used; the better performance is expected
It shoutl be noted that if L=l, that is,
VA [E(e2(n))] = VA [e
2(n)]
VB [E(e
2(n))] = V
B [e2(n)] ,
this corresponds to using the instantaneous error square for
estimatingthe gradient, and this filter reduce s to the adaptive
recursive filter proposed by White [3] . If the further approxima-
tion is made that the sensitivity components of equation
(3-47) , (3-48) are
3y(n) / x-4-^
—
- = -y(n-p)3ap * tr '
3y(n)3bq
= x(n-q)
then the estimated gradient is
VA [E(e (n) )] = - 2 e(n) y(n-l)
y(n-2)
">
and
VB [E(e
A(n))] = 2 e (n)
y(n-N)
x(n)
x(n-l)
x(n-N)J
= -2e(n)Y
(3-50)
= 2e(n)X
(3-51)
71
Then the filter coefficient updating process is
A(n+1) = A(n)-2k e(n)Ya.
B(n+1) = B(n)+2kben)X (3-52)
where k , k< and filter output
y(n) = B(n)TX-AT (n)Y (3-53)
Equations (3-53) , (3-52) , (3-51) , (3-50) are exactly the same as
the algorithm proposed by Feintuck [2] . This Feintuck algorithm
has an advantage in simplicity when compared with the algorithms
proposed by White [3] and proposed here which require additional
recursive filters to generate the estimates of the gradient.
Thus, it may be useful to extend the Feintuck algorithm to the
two-dimensional recursive adaptive filter for simplicity. In
the next section, the algorithms proposed by White and Feintuck
are extended to the two-dimensional algorithm.
3 . Two-dimensional Recursive Adaptive Filter
In this section, a mathematical model of the adaptive
recursive filter for the processing of two-dimensional signals
is proposed. This can be considered as an extension of Fein-
tuck's algorithm to two-dimensional filters.
Two transversal filters having the same structure as the
linear combinatorial system used in the non-recursive two-
dimensional processor, are used in the recursive processor,
one for the feedforward path and one for the feedback path.
The two-dimensional recursive filter is described by
its transfer function.
72
y (v z2 ' ,L ? b
ijzr iz
2
"j
1=0 3=0 j
X (z 1/ z„) .12n
M N -in -nL _ mn 1 Z
m=o n=0(m,n)^(0,0)
In the spatial domain, the input-output relation ofthedigital fil-
ter is given by
p q M Ny(k f £) I E b- .x(k-i,£-j) - E Z a v (k-m,£-n)
i=0 j = J n=0 n=0mn
(3-54)
The following notation is introduced
B = [b b n , ....b b, _ b, , . . . .b n b_ ft b J00 01 oq 10 11 lq 20 pq
XT = [x(k,£) ,x(k,£-l) , . .x(k,£-q)x(k-l,£)x(k-l,£-l) . . .x(k-l,£-q)
x(k-2, £ x(k-p, £-q) ]
ip
and A = [a^ a QT . .a^ a1(a ir . .a
1Na20 ^
YT=Cy(k,£-l) , y(k,£-2) .. .y(k,£-N) y (k-l,£ ) y (k-l,£-l) .. .y(k-l,£-N)
y(k-2, £) y(k-M,£-N) J
The filter coefficient vectors ~A~ and ~B~ are [ (M+l) (N+l) -1] xl
and (p+1) (q+l)xl, respectively, and the input-output signal vectors
are again (p+l)(q+l)xl and [ (M+l) (N+l) -l]xl, respectively
.
Then equation (22) can be written as
y(k f £) = BTX-ATY (3-55)
Here, to obtain an estimate of gradients of the mean square error
function, a single sample of the square error is taken. That is:
V[ECe (k,£))] =Vte (k,£) ] =2e (k,£) V [e (k,£ ) ] , and again the
adaptation scheme (filter coefficient updating process) is used
in the same fashion as in the nonrecursive case [see Figure 3-*5] .
73
Denoting the error at jth iteration as e(k,£) then
A(j+1) = A(j) + 2kae(j)VA (e(j)]
B (j+1) = B(j) + 2kbe(j)VA (e(j)]
where
e(k,£) = e(j) = y(k,£) - d(k,£)
(3-56)
(3-57)
The components cfthevectorsv, [e (k,£) ] andv„ [e (k,£> ] can beB
calculated as following.
From Equation (3-57)
VA[e(k,£)] =V
A[y(k,£)]
3y(k,£)3a
01
3y(k,£) 3y(k,£) .
.
3y(k,£) 3 y(k,£)3a
OM3a
10
3y(k,£)
3a
3aMN
1M
(3-58)
3a20
and VB[e(k,£)] =Vg[y(k,£)]
3y(k,l) ... 3y(k,£) 3y(k,£)
oo oq 10
3y(k,£) 3y(k,£)3b"
3y(k,£)3bpq
iq
(3-59)
TB20
Note thatV. [e (k, £) ] and V_ [e (k,£) ] havethe same dimensions as A" and
B, that is:
[ (M+l) (N+D-l] x 1 and [ (p+1) (q+1) ] /respectively
.
From the recursive relation of Equation (3-54)
3y(k,£) ri n \ r v ~y(k-m,£-n)Ut—' y(k-r.t-s) - I I a 3^
'J
-
rs m=o n=o rs
and 3y(k,£) = x(k-u,y-v) - M N
Tb~3 y(k-m,£-n)
uv m=o n=o mn(m,n)^C0 f 0)
uv
(3-60)
(3-61)
74
The recursive relationship of Equations (3-60) and (3-61) should
be noted again, which can be implemented by additional recur-
sive filters.
Forming the instantaneous error gradient of Equations
(3-54) , (3-55) using the output of additional recursive filters
of Equations (3-60) (3-61) , the filter coefficient adaptation
process of Equations (3-56) (3-57) can be performed. Note that
this algorithm corresponds to the two-dimensional version of the
algorithm proposed by White [3]
.
If we make the approximation
^ (k * £) = - y(k-r,£-s)9ars
lVk,Z)
= x(k-u, y-v) ,TEuv
then it follows that
V A [e(k,£)] = ry(k,£-l)
y(k,£-2)
y(k-l,£)
y(k-l,£-l)
y(k-2,£)
A -= Y
y(k-M,£-N) )
75
and VB[e(k
f £)] =(x(k,£)
x(k,A-l)
x(k,A-q)
x(k-l,£)
^x(k-p,£-q)
A _= X
Therefore, in this case, the complete algorithm is described by
Ae(k T £) = e(j) = y(k,£)-d(k,£)
A(j+1) = A(j)-2k e(j) Yd
B(j+1) = B(j) +2kbe(j) X
and Y(k*£) = BTX - AY
This is the two-dimensional version of the algorithm proposed
by Feintuck [2]
.
76
IV. ADAPTIVE NOISE CANCELLER
A. THE CONCEPT OF ADAPTIVE NOISE CANCELLING
Noise cancelling is a variation of optimal filtering that
is highly advantageous in many applications. Specially in
Wiener filtering or Kalman filtering, which are optimal,
apriori knowledge of both signal and noise statistics are
required. Adaptive filters, on the other hand, have the
ability to adjust their own parameters automatically, and their
design requires little or no apriori knowledge of signal and
noise statistics while the Wiener approach utilizes a fixed
parameter filter based on known statistics.
Figure (4-1) shows the basic problem and the adaptive
noise cancelling solution to it. It makes use of a reference
input derived from one (or more) sensors located at the points
in the noise field where the signal is weak or undetectable.
This input is filtered and substracted from a primary input con-
taining both signal and noise. As a result the primary noise
is attenuated or eliminated by cancellation.
At first glance, subtracting noise from a signal seems to
be a dangerous procedure. If done improperly it could result
in an increase in output noise power. If, however, filtering
and subtraction are controlled by an appropriate adaptive
process, noise reduction can be accomplished with little risk
of distorting the signal or increasing the output noise level.
77
Signal
tAdaptive
Filter
SYSTEM
OUTPUT
i
Filter* Output
e (error)
FIGURE 4-1. THE ADAPTIVE NOISE CANCELLING CONCEPT
x(j)
x(k,Jl)
Adaptive
Filter
x(k-61,£-6
2)
*
7reference input
error
FIGURE 4-2 NOISE CANCELLING WITHOUT AN EXTERNAL
REFERENCE SOURCE
78
The following argument for the above is mainly due to
Wldrow, et al [4] . In Figure (4-1) , a signal S is transmitted
over a channel to a sensor that also receives a noise N.o
A second sensor receives a noise
N, uncorrelated with the signal but correlated in some un-
known way with the noise N . In addition to these noises,
additive random noises M and M, uncorrelated with each othero 1
and with S, N and N,are present. Then the reference input
is
d = s+ n + yi (4-1)o o
and the primary input
x = N1+ M± (4 _ 2)
The noise N.+ M is filtered to produce an output y that is as
close a replica as possible of N + M . This output is
subtracted from the reference input S + N + M to produce the
system output
z = S + N + M -yo o *
In other words, the practical objective of the noise cancelling
system is to produce a system output z = S + N + M - y that is
best fit in the least square sense to the signal S. This ob-
jective is accomplished by feeding the system output back to the
adaptive filter and adjusting the filter through the LMS
adaptive algorithm (described in previous chapter) to minimize
total system output power. Note that the system output serves
as the error signal for the adaptive process.
79
Assume for the moment that the noises M and M, do not exist,o 1
then if one knew the characteristics of the channels over
which the noise N is transmitted to the reference input, it
would be possible theoretically to design a fixed filter capable
of changing N. into N . That is, if the correct model of this
transmission channel, H(z), is obtained, the adaptive filter
would be simply 1 , a fixed filter.HTiT
Assume that S, N , N, , M , M. , and y are statistically
stationary and have zero means. Assume that S is uncorrelated
with N and N, ard that ^andM^ are uncorrelated with each other and
with S, N and N, , and suppose that N is correlated with N .
o i 1 o
The output z is
z=S+N +M -y (4-3)o o
squaring, one obtains
z2= S
2+ (N + M - y)
2+ 2S (N + M - y) (4-4)o o J o o
Taking expectations of both sides and realizing that S is un-
correlated with N / M and y, yieldso o * J
E[z2
] = E[S2
] + E[(N + M - y)2
] + 2E[S( N + M - y) ]o o o o
= E[S2
) + E[(NQ
+ Mq- y J
2] (4-5)
2The signal power E [S ] will be unaffected as the filter is ad-
2justed to minimize E[z ]. Accordingly, the minimum output power
is
min E[z2
] = E[S2
] + min E [ (N + MQ
- y )
2] (4-6)
2Since the filter is adjusted so that E(z ) is minimized, therefore
2E [ (NQ + MQ
- y) ] is minimized. The filter output y is then a
80
best least squares estimate of the noise N + M . Moreover,
2 2when E [ (N + M -y) ] is minimized, E[(z-S) ] is also mini-
mized. Since, from (4-3)
z-S = MQ
+ NQ
- y (4-7)
Adjusting or adapting the filter to minimize the total output
power is thus equivalent to causing the output z to be a best
least square estimate of the signal S for a given structure and
adjustability of the adaptive filter and for the given reference
input. The output z will contain the signal S plus noise.
From (4-3) , the output noise is given by (N + M - y) . Since
2 2minimizing the E[z ] minimizes the E [N + M - y) ] , minimizing
the total output power minimizes the output noise power. Since
the signal in the output remains constant, minimizing the total
output power maximizes the output signal to noise ratio. Note
that if E[(N - y)2
] = can be achieved , then E[z2
] = E(S2),
therefore y = N + M and z = S. In this case, minimizing output
power causes the output signal to be perfectly noise free. Also
note that, on- the other hand, when the reference input is com-
pletely uncorrelated with the primary input, the filter will "turn
itself off" and will not increase output noise.
In this case, the filter output y will be uncorrelated with
the primary input. The output power will be
E[z2
] - E[(S+ MQ
+ NQ )
2] - 2E[y(s+N
Q+ M )] + E[y
2]
= E[(S + M + M )
2] + E[y 2
] (4-7)o o
2Therefore, minimizing output power requires that E(y )
be minimized, which is accomplished by making all weights
2zero, bringing E [y ] to zero.
It should be noted that in applying adaptive techniques
to a practical systems problem, the key step lies in providing an
appropriate desired response signal for the adaptation process,
that is, the reference input should be provided through the ap-
propriate scheme, while the exact knowledge of statistical
characteristics are not required. In adaptive modeling applica-
tions, the desired response is generally available as the output
of the unknown system to be modeled. And also in the noise
cancelling scheme above, the reference input is available by
sensing noise which is correlated with the noise at the primary
input in some manner.
In next section, the signal filtering problem is discussed
when no external reference input free of signal is available.
B. NOISE CANCELLING WITHOUT AN EXTERNAL REFERENCE INPUT
This section is concerned with signal filtering
(estimation) a noise-corrupted signal when no external
reference input is available. Here, it is assumed that only the
noise -corrupted signal is available, that is, referring to the
Figure (4-1) of the previous section, the noise free of .signal N^
82
which is correlated with N that corrupted the signal S is not
available; only S + N is available.
It is proposed to estimate the signal S by cancelling the
noise N in some adaptive way. In the following, it is shown
how a reference input can be obtained for the adaptation process
under certain conditions. Assume that the noise corrupted
signal x = S + N is composed of broad band noise N and a narrow
band signal S, then the autocorrelation function of the signal
is broad and that of the noise is narrow. Also assume that noise
N is uncorrelated with the signal and that the mean values of
both signal and noise are zero.
Consider a signal delayed by 6 units,
x(j-6) = S (j-6) + n (j - 6) (4-9)
where 6 is a sufficient number of time units so that the noise
component is decorrelated, but the signal component still
remains correlated.
Then
E[n(j) n (j-6) ] =
E(S(j) S (j-6) ] ? 0, finite (4-10)
For the two-dimensional signal, a signal delayed by &-., 6?
units in the horizontal and vertical direction respectively,
where 6^ and 62
are sufficient length of spatial units such that
the noise field would be decorrelated but the signal field still
83
remains correlated, then
E[N(k,£) N (k-61 ,A-6 2 )] =
E[S(k,£) S (k-6 lt l-62 )] f 0, finite (4-11)
and again it is assumed the signal field and noise field are not
correlated with each other.
Now if this delayed signal is used as a primary input and
the original input used as a reference input to the adaptive filter,
then referring to Figure (4-1) of previous section,
S(j-6) or S(k-6n,& -^-y) can be considered as N-^in Figure (4-1)
and N(j-6) or n(k-<5-i,fl - 62 ) as M^, and S(j) or S(k,£) can be
considered as N and N(j) or N(k,&) as M in Figure (4-2), re- •
spectively
.
From equation (4-10) and the assumptions that the signal and noise
are uncorrelated, it is seen that the assumptions made in the
last section for the various signals holds here.
Therefore, from the argument in section IV-A, , the
filter output would be a good estimate of the signal S . Figure
4-2 shows the noise cancelling (or signal estimation) scheme
xiscussed above.
84
V. EXPERIMENT AND RESULTS
In this chapter, a computer experiment is performed to
check the feasibility of the algorithms derived in Chapter
HT for certain applications. The signal estimation problem
for a noise corrupted signal is treated here for both one-
dimensional and two-dimensional cases. Nonrecursive adaptive
filtering and recursive filtering have been examined and the perfor-
mances of adaptive filters are compared to that of the optimal
Wiener solution. The adaptive noise cancelling scheme is
used for this application.
First, consider a band limited one-dimensional signal S
corrupted by noise N; it is desired to estimate the signal.
If the statistics of both signal and noise are known apriori,
a fixed optimal filter to estimate the signal can be designed
by the Wiener Hopf solution of equation (3-11)
.
Here it is assumed that these statistics are not known
apriori but only that the signal is narrowband and
the noise is a broad band signal and the signal is entirely
uncorrelated with the noise. Then the signal has a wide cor-
relation function while the noise has a narrow correlation
function. Separation of this broadband noise and narrowband
signal is now required for the estimation of the signal.
It is assumed further that thedesired (or reference) signal
which is needed for the adaptive process is not available, that
is, no other possible reference signal is" available whichmay have some
correlation with the signal we want to estimate .
This problem can be considered as an adaptive noise
cancelling problem without reference input. Assuming that the
noise is white, then from the Figure (4-2) one unit delay is
enough to decorrelate the noise component appearing in the adap-
tive filter input from the noise aonponent in the desired signal. These
components will thus appear in the error but not in the filter
output. The narrowband component, on the other hand, will not
be decorrelated by the delay and will appear in the adaptive
filter output.
The input signal would be
x(j) = S(j) + N(j)
where S(j) bandlimited signal
N(j) white noise
and the reference input would be
d(j) = S(j-l) + N(j-l) .
The form of autocorrelation function of the signal is assumed as
R(m) = p
'
m 'cos w m
For the purpose of computer simulation, the following values
are assigned:
p = 0.95
w = 0.025
and the variance of noise is 0.5.
86
For the optimum filter design using the abovevalues ,a transversal
filter having 10 delays was used. From equation (3-11).
W -1LMS = R x R ,
xx xd
The autocorrelation matrix R was computed as2£X
/' 1.50000
\
0.94970 0.90137 0.85496
.94970 1
.93rJ7—0.
0.
0.
o.
85496
81044
70774
72634
6U7o7
65020
61436
500J0
94970
.90137-
8 5496
,8104-.
76 774
.72o84
.to/67
65 02 J
0.9-.970
1.50 000"
-0.94970-
0.901^7
C.85.V6
0.ei044
.0. 76774
0.72684
0.687o7
0.90137
3. 94970"
. 1.50000-
0.94970
0.90137
0.35496
0.81J44
0.76774
0.72084
C.eiC44 C. 76774 0.72 684 0.68767
0.85496 0.81044 0.76774
(T.90137 0.95496 0.81044
0.94970—0.90137 _ 0.95496
1.50}9J 0.9497) 0.90137
C.S4S70 1.50000 0.94^70
0.9C137 0.94970 1.5OOC0
2.95496— 0.90137 0.94970
C. 91044 0.85496 0.90137
0.76774 0.91044
0.72684
0.76774"
0.91044
0.85496
0.90137
0.94970
1.50000
0.94970
0.65020 0.61436
0.S5496 0.90137
0.697*7
"0.7268 4 ~D
0.76774
_0.8U)44^
0.85496
0.9^137
0.9-.97C
1.5000C C
0.9497C 1
.65020
.68767
.72684
.76774
.81044
.85496
.90137
•9497C
•5033C
and the crosscorrelation matrix R , asxd
Rxd1.00000 0.94970 0.90137 C. 65496 0.81044 0.T6774 0.72684 0.68767 0.65020 0.61436
Then the optimum Wiener Hopf solution gives the filter co-
efficients as
W = Wiener Weight-Vector -/
0.33207
0.21191-
0.1347 5
0.0855b
0.05V27"
_ u»03<.34 -
3. 02169
0.01369—
O.C0370~
_ 0.005<>7
87
For simulation of the bandlimited signal, the state and out-
put equations of example 1) in Section 2 of Chapter II are
used.
For the nonrecursive adaptive filter application, again
10 delays and = -0.005 as a step size in LMS algorithm were
used, and 2 delays for both feedforward and feedback path and
= -0.001 for ka , kb (equation (3-42) ), were used in the recur-
sive filter application of both Feintuck's algorithm and the
algorithm developed here. Eight points were used for error
square averaging for the gradient estimation (L=8, in Equation
(3-40) ) . The experimental results are plotted in the following
along with the descriptions and optimal solution for the purpose
of comparison. The results indicate the adaptive recursive
filter appears to perform as well as the optimal Wiener filter
once it reaches a steady state condition.
88
*
V f\
H4
FIGURE R-l SIGNAL STATISTICS
Autocorrelation =0.96 m cos (0.025 m)
89
I
I 'I
I kl HIP
!l
I i li
U. H
FIGURE R-2 NOISE CORRUPTED SIGNAL
NOISE: WHITE: Zero Mean
: Variance = 0.5
90
FIGURE R-3 WIENER-HOPF FILTERING
10 Delays are used.
91
FIGURE R-4 WIDROW'S NONRECURSIVE ADAPTIVE
FILTERING
Number of delays used: 10
Stepsize Used: -0.005
92
FIGURE R-5 FILTERING BY FEINTUCK'S ALGORITHM
NUMBER OF DELAYS IN FEED FORWARD PATH: 2
IN FEED BACK PATH : 2
STEPSIZE USED: -0.001
93
FIGURE R-6 FILTERING BY THE ALGORITHM USING A
FINITE POINT MOVING SQUARE ERROR AVERAGE
For the estimation of gradient
Number of Delays in Feed Forward Path: 2
in Feed Back Path : 2
Stepsize Used: -0.001
94
As a second application/ consider an image sensed by an image
sensing device of anNxN sensing elements array. It is assumed
that this image is composed of correlated background and a
three diagonal line target trajectory. This image may
be interfered withby the internal noise of device (assumed white) .
Then the output image includes three types of processes:
x(k,£) = S(k,£) + T(k,£) + w(k,£)
where
S(k,JL) = correlated background
T(k,£) = Target strength (three diagonal line)
W(k,£) = noise.
Again it is assumed that no statistics are known apriori
and the correlated background is a narrowband signal. It is
further assumed that thecorrelated background and noise are
uncorrelated with each other. It is proposed to separate
the three diagonal lines from the background noise. Again, the
same argument holds that this problem is a two-dimensional noise-
cancelling problem in which no reference is available. It is
further assumed that the correlated background is a band pass
process for which the autocorrelation function is
RSB (m,n) = p ri p v
>
cos w^n cos w^n
where Pv, Py rePresent horizontal and vertical direction cor-
relation coefficients respectively.
95
Going back to Figure (4-2) in the previous section, the delay
z, z~ will be sufficient to decorrelate the white noise
and diagonal line target. Then as a result of filtering, the
system output (or the residual field) will be the desired sig
nal. It should be noted that this residual signal is com-
posed of an estimate of three diagonal lines and white
noise as well as seme granularity due to the fixed stepsize[l].
The problem of enhancing the target diagonal line which
is subjected to the noise (white noise and adaptation noise)
is another problem of interest. It will not be considered
in this work.
For the purpose of simulation, the following values
were used:
1) p = p, = 0.96 w,= w = 0.143J v K h h v
2) p = p, = 0.99 wu = w = 0.143J v h h v
The variance of correlated background = 1.0
White noise variance =0.1
Target diagonal line intensity = 1.8
For the optimal filter design, using above values for p =
p, = 0.96, the Wiener-Hopf solution of Equation (3-30)is:
ffLMS " R
xx-1
Rxd
96
where WLHS;Rxx , Rxd
are defined by equations (3-22), (3-28),
(3-29)}
respectively.
Using p=3, q = 3 in equation (3-20),
Rxx=
1.1 JwviC
—1.97969-
3.94J29
j. 97989
0.96cl9
-0r9*t-»3-
>.9; .29
J. 92138
.1.86414
vi. 979B9
0.979C9
«.94v29
0.97989
-e.96trl*-
i. 9^133
J.S4j29
ii.92iid
J.9t;29 „. 97939 J.96 19 ^.92138 v. 94.. 29 ,-.92138
—.-97 56 9 r.9*-29- ry*7«89 .:.96"1^ 0.92138" —0.94J?O-
l.UJuJ 0.97989 0.5*029 0. 97939 0.96r i S t. 92138
s.. 97*69 i.A.CJ 1.97C39 ..94 29 '.97989 ;.96.il9
L.9>o29 C. 97989 I. KCO'J C. 97989 0.94C29 c. 97989
-V.-97-989 V.9V-29 -*, 97989 1 .lv'-vv -.7.97989 r,.9-r»29-
••96*19 .97989 -.94 29 G.97°89 l.lCtwO 0.97989
. .92x38 u. 96019 0.97989 U.94C29 0.97989 l.lOOCQ
C.94~29 -.92138" '".«£ 19 n. 97989 } .94 '29 ~ V.97989
414.88
.92? 3«
94', 29
,9213*
,9601
9
,9-7-9?'-
.94029
97989
.1" "'.
< Rxd>T=
J
l.j,.J .. .97989 . .9-V-.-29 .97989 •;.«6' 19 ^.92133 0.94029 0.92138 0.83414
(WLMS)w.3<-079
... 28 jJ 3
- . 1194
t.24o3i
~. -,9 84'i
-J.J2430
J.lie29
- .)i334
-3.04117
97
Only the recursive simulation is performed and compared with
the nonrecursive Wiener filter. The simulation results are
shown on the following pages and indicate that although the
optimal performance of the Wiener filter is not achieved,
the adaptive recursive filter performs well.
98
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FIGURE R-7 CORRELATED BACKGROUND + THREE STREAKS +
WHITE NOISE
Background: R(m,n) = . 9^m
' 0. 96^' cos(0.143 m) cos ( 0. 143n)
Variance =1.0
White noise: zero mean
variance = 0.1
Streaks intensity: 1.8
99
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FIGURE R- 8 resjdom, SIGNAL AFT£R wie(jer filtering ^^^ ^White Noise + Streaks
100
tuuimuummiiitutuuiumtutumiiutmttmiiutuu tun a i i it an 4 u u u 1 i u :nn tintmif-- - - — -- '.'-,/ n «••,«« n-- -.'/" •
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i tt ft/rn •* .--•••;//»'i ti / * n ./'//*•----///*
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tt -1 I III I It II * */' •/ /f^/ ////* //// * Uttl tl
r i it itm ** • •••" trrrr**-**um*rrr*f**tm*ifiu'-a — "*u • •' • • mrrrrtt t1 1 nit itt • "it • -i 'i f i /*/*/ *m t n t i**mmum i "i • - -*u "tm**i****ut{l-. i ...— -/ ntrt •*n%r *i *mm-ru i in in m- fit*itm*t*tii ft*
. -t i n*i it tu*ui 1 1 1; 1 1 1 taut*n i it i .// i intuitu • f i 1 1 1 1 1 < t <• " i t-it u 't*i**ta* • nun 'i* i • "i > ' 1 1 in /«/..!/ -ittt in *t t ti 1 1 ii n i '« •• t*- * • * t*i*ruu*xi it** i*it i - fin tit i in • • f'l-ift mm****- • t t tr ' u • turn ti t r mil -m r-rrtr t*t-nit t «•«; ••//• m*- tit tut- /*•%•• •-•/ '/• tu-i t ti •***t*'tt i* ii n «// itn i titu mi *rr-">rrr- in - i/;,..... .... . , |. . •/ >•/// n am* t in- mu*u t> t i i a - •/// tn*i • ti> 1 1 mi-u *;//;* n**uiiuti****--i***-*iti* -lint u • - f- • rtm -**t >nt *'t it a < -t* n tm i/i«cs. • * r i*' 1 1 a atan 1 1 t» mi ii i i '-m i it ft i ii m n.'t *m i
//*/../- rf' + -i-i>*-i*iuui-iri*iti i in u* at tin* •• rr-wnutriu •/
.
-it+tt*m*tumii-i»-ii-it*iiL'- t •nftuuii*m iiiiiu t tt **i -ttuu-im!/• /•«/ •mmi'mtutit *uim*mu m*ii * tat -• a f i tt* m*i *nt n tu * in a atmi a i • t a i rmrrirn ••tani i u *ia \uuu i * < * i*i- *• 1 1 * ft •- utt mt*u*f t* tt m*t i it tin mttraititnnutttittta f-r-u ii*ii f -nt in t< 1 1*- 1* r 1 1nm i /* / t-itim-ttt /• // m <i t- titm tt i*lat'tauiat i -// * i*utii t-t i ittt***i tun mil -• ami • im tarn taiiii irtirrt*lati 'it > -<• 'ti 1 1 i 'i aiaamtata '-i i • it i / // a **ui it *ut :tt it t iu 1 1 ni ?* rittmuati timn . *•* \
*•-• *-t •*! itiiitu tt im- i ufrr*-ia*iiut'iiaamii i ttm nmnrtnit ti tit in in i n tt'i" i -it* •tit t tit- j"i i a u im? ta i* t in n *m r - • t tir t
- j t"mmnma '•ai- 1' * *< i i»* *• "f-ni'-i'm- i m-- 1* <mnm imntm aim ••' •'!» / •<??' -mm ••nt'-i t • i *'••» "it *i-*> -•• a* 1 1 ii ina tm '•• it ' n • i
•' 'i tm tut ftn i • :;• -inmmimii > i- "/ -//// t* --i •a* n tut i tiw m^ in 'in '.•• it • a ami it am •• i in- t- imitn
~tn*ftiiiiitr.tt < nam taa%i in* .//*-- -mi nma 'aa 'in t in -r • t "/;//« •/ inn
tin **am*itt t i tt i -'tint — tmi •//-«•.-;»- •/ n-mnmm tt i t it • ' it rt^mmtmn it -,».... >ii* '•amunii-<*iT4*iituami<*'>i'- * aa-L t • \n / t vt i limitin a i 'it-taitt * t a i ni ia*tm*a t n • - '*- /-- // / lima i t i* • -if -nm tint-m nt itiiitttaii'ii at •tiitf • •a*nt**:*m if*n- -«••• /• /-// am t t *t r tt rm ''ti--—mm-n-—-**+~+*.i~"~t—inriuit-t*»iti+-"fmm-- — — *t tt t* in •/// tin * i-utt 1 1m--*m-it-um i//<,H4....».i*i / - mi ni ttr-am * 'tt * •'!••///• •/ / ta *i until a tiiaia --mum*t - . .^-...* .. , ti tun iia*i*atiui> •rr-"*~"- ••"imr**i *mat a -
• -t* <*t t >mummmt */*tit • >/ -n in timaiati t ,m timi • n- mt/tr/^ rttttit* r ta *t t in -n *ri t>rr+r imi trim-"//m^.«*» .-.«' / H nati*»n.u it u*tm—-• ' mnm tttim ta • im*tmm n itmm < •*
tat u /,'. , uumtan n i-- t *•• + •-> >t turn * tutu -t ia in i*u nm it t-t ••
ttiutiu*ti*uuaa*'*ti'U'tiiatiuiti--f **•*-'imam t • it •! t ; i u t n •i>*mm i **irm •
lf<*i*" i /-/ n -utt/ia •'t**i****tmi itt"---** *n tui'-ii* ••tin- nam : -tt ; it *m •*// //m f • -i '. < n>' *i . *m taatm i- um n+*-*r mi 1 1am it i-i- 1 1- -nimt'm-ut+m-»»ii-*.t*~infiiiitamiiti*tmt*utaauitiutm-*i m m 1111*1111 a tmi*t n* mtir*ir---*<*iiiuiU'inuiittf'titiiuutuuitf 1 t" t*i-mmi m 1 1* -in t uttai /tmm *im* «•-»•••aa im in - inm mm u it ami-am- • *m •mnm 11 >r - / •mmm-im -imi «*»ltft**t*******tai'tmn< \ • 1mmm t* 1 :mi tun 1 tm-- ill m 11 tint i*tmm —. /•*ti9*********atn uui.-'*tttiiiitimm**mtitn ttttai*- 1 tu-.m* ittt*- m*i—">*•••m a-, turn *-um tun >t t—t •mt- uti-i tt it tin t--i •///»• //-*• •• •--inaiuttfun-— **> • nit at 1m -11111mi 11m * 1 a n- \ia 1 • t-t - / " 1 1- 1 mt ?/*/**/-*»•»«.
itutt -in tit m-tti t itm 1 'a 1 1 1- 1'm < 1 **u 11 in titm t* tint t 1 it 1tm tt *'••*'••tat tt -tt -t f.*i 1 t*it*tf tt 1 t*u 1 : 1 : 1 in- ' n -'i *t t?iuimtt tt* na t-t m m-fT'i-'*niui tatf-a 'i—a auttr t- •/ / '/ t'ti '<i***tu it/ 711 tm tt n tm tr ttut mm* -*«/• /•»/,-,.,„.,,- , R , ,,. /-.nt 1 Utiu i tu tm *t •. / / / 1mnt nt it *-t t 1 111 -it 1 1 mi rat 1 1 it 11 ///• -•>*Itt . -;../-/ a r • a '-.tut niaa mannam tan ita > t-t- nut tt* 1 t-m tin mm 1 t-'fiti****'»*****\-<-*~>ititi*i.iut**tf>nntmtfti na • at-* • 1 1 •/ • un i- m mtin*- " 1tn*.*-ti-i-— <i • — at mat u»*tt*'iu- m 111* */ •iti'tit iittt-r-- in- ••* - •• tm u- t*i1 1 *••*•*-* *—*t *~i * 1m ii'* tint -—tun n" "ii-*- ••>!- if 1 •!-•!• ; ;m-umitft-*-'—nn*-**-ttni t'
ti****,**-—• ami u -i ui-uttmmit . t ''!• an ••••-' ./// mum tuna* 1 • iintf—i— miit*-* + t*>iii-t-ft*Ufit**tuitunt n • •!/•/•' ,. > //..// ..,-.; / !'"••-*mm//...,-h - a, i*u* '* i***t:*itit tun -» tivt-f tr tin tt tr -i mn i*-"tn i»im* a 1*//•*».-// 1 1 1*** 1* tut* i t-m-u 1 at i •'•*' / 1 ***•!!• •tutin • 1 1a tana •!• 1 •*•*•• it mi* tu tit* --it- >uu** i* tat* 1um t •nn.tmn tutu -nt tr r- • 1 u t <i*m- it •am a - ;• an* turn nii***-r.'*m***uumutmt*uuu 1 : 111 11 at 1
//•« • -m a 1 1- nit 1 tm 1 tt' it 1mmm 1
1
ti* ***i*-ni i-mim 1 1 u nuuutm 1 11 \-t 1 -11111 tit *t t-ut r *m it 1 •-•-* 1 1-- rtm 1tt*-»t. tun** •mint -mum* mn 11 u*-*u *u> t ~ tun i •*- ** u u ni t-~ u> mi*nm* n- -tii-*-! a -m*'t .'in ttu •uuiuutiiuu" 1
-- -u wtf it 111 -ii* tt* t- it u mimti' u-iuunu**":i' Muuiuiuui u*uut. mt 1 •"///«•/// •//- n tt tu \fi*ii' i- trim " 1 •«•*/* / u*rm 1it--*** »... /- // a 1 it t*i t **u 'i • 1 ti - -t '.
' tun 1 it *'/' 1 1 *"• 1 11 •u'"ii- - - /»» tm tr tr*tt**— > . . t uir 1 a 1 itu*mm 1 u -i*r- 11 11 1 1 1 1 ,• -1um •*u*- it 1 it tmt 1 m*t**i 1 wn u*ti"*******mu t--t laatttmt n tt t-m 1 it- fat' n um* *m ttmmur m-iimitft- 1 luuiu 1 -uii i nan at 1 11 1
— 11 •' -i • nn 1 1 1 1 u . 1 uut* t 11 1 u iim-im-im-
ft , ,. i*"- 1 itu> t , . -11111* mt 1 1 it ti it 11 nt it n -u < ft tu •t.i 1 1' -in 1 um 1 1- tit-**' 1 - 1*a : u**4 ,-.-,/..- / ., //.-«- ./*./««.. 4./ "•.•///////-!>• 1w 'it i-' 1 1*1 imii . imu ••// 1 im u nit *!*>**tt'itu n t,mum in it it. u *'ii -1 'tm ui-*um-u >i t >i mnttam *m*tm 1 1> m*u 1*tu iit*it'*i t'ti tti 1 /- «•• * '•••! tmtui 1 **i 1 am a 1 11 1am 1 1 > • 1 • immm*ir --1**1
•If •llllllll- ""•!!-* 1 un *, mi 1 4 tut* 'Iff lit •"'*•*••?•• **"f*u 'tut*- tt r* *"-***! i' ant i nuuuttr u*4 -tr— mut»-r*
FIGURE R-9 RESIDUAL SIGNAL AFTER RECURSIVE FILTERING
White Noise + Streaks + Adaptation Noise
101
.2. «.-.< t*« iii**'TP<* 7**.'f*Q i:i*«»r(t« :3*'iM! .f*,ur>*< '.'i*'im« •?•- 1 1 1 tt tuta itntrttrf**—'*<"**-—•-*—•'• »tt it mttt mi tttttf\-*
i /t a 1 t tt an I tt/ 1 1 t-/////*** u 1 1 rm u tttt **mtrf tw""-- *
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: m; I III it till It tt 1 1 1 1 1 1 •»•*•"—-1 '•«M4MitN4^4J'l , : 4 .-. -tUH 111 I iU tlttl .'/ /- - «yii.4. v i»» i*, | ./!/ U/tl lit till till ' '- «« 4MWK- .*».-*.
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w*"»i luiiiiuttiitimutt .r*- • ^- - tttuinu uatimtatti tit it tat tu u unit • i'< - *•• imu tut itnm itiut tt*- .--.••*
( - um mi tt timtimi'i in um it i • \- < .».-.."- .- 'in tum intun iiim t * -.<..-.,- <•••• •mmmu <.U U U l ...-...,/,,,,', ^ /^.^..^,....»,»-«i J--. I Iff It l~..~~**lf t HI till .«4WM.I.H<1.M^-.4. '-11111111 Ifitttu**'**-**>uttutt-- i
* • 't in u*-' •-.••-/////////*•" • i- i tm itmi 1
tttt ttu •»••/// mitt t f i. -tu. —•/////...»"• *timuut"~— \—
—
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mutt i u --I *•• -• • 1 1 1 1 !• * iimn-" — • <.•.-- .. in mt it *u22i2 tutmu i *•• •////. t». .»•«..»•», ////// .•.-..-- . .. .. — n tut tti imm tt ***im tt tu ui-- .......... 1 1.*. ////.».»•*.. .*.. ..////// - .- "-.4 mm it ;it ttu tu »•/ lutimttt* - i ».... ' •i--«////......... ..././/////' *,.«-. .. fc it ttu a jmUluttllltUfUtUItt*'"* —i«- fc ..^i»- i i*» ./////»••«..«.».//////////.. .-v.^..-..- ^..j.//^/,,/ «_-.tauntmumut it -• •».-..*.'•
—
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..—. n tut t tttt f tut tutu nutu t • »• -—-/ntn itttf "-• •-- - * ttu i u t uu it 1 1 um t tut nm • • *** i' 'tint ;luiiuuu /" '•*- 'ttti iiium fit tituumttitf ' •-.--.-•.---••// Jmi 1 1 atum ••"-**'— •- - "••— -I ii imm tutmi u <• -• -•«.• .// 4
rj 1
mu tu 1 11 1tm 1m if"-- :"*»• 1ttuumuinumii--- -— j
t III it 1 11 1 1 itU 1 1 1 1 1 1 n " ~
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FIGURE R-10 Backgrounds + White noise + Streaks
Background: R(m,n) = 0.99m 0.99 n cos (0 . 143m) cos (0.143n)
Variance = 1.0
White noise: zero mean
Variance = 0.1
Streaks Intensity: 1.8
102
»»s?M 13»««?m i2?«5»m* *!)*v?m t."*«rrsi ':'«vf<* ir^*-^T8*
*«*.*/•///• 1*1111
t
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Figure R-10
White Noise + Streaks
103
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104
VI. CONCLUSIONS
The study described herein has developed a new algorithm
for the one-dimensional signal filtering problem and extends
this to two-dimensional processing. It is an adaptive recur-
sive filtering algorithm based on the steepest descent gradient
method which employs the finite point square error for the*
gradient estimation rather than instantaneous square error.
A simplified two-dimensional version of this algorithm
is developed. It is designed to estimate the signal in real-
time operation in cases where the statistics of both signal
and corrupting noise are not available apriori. The algorithm
learns the statistics and adapts even though it is not optimal,
which means that it seeks the minimum of the error criterion.
It should be noted that Widrow's nonrecursive adaptive filtering
algorithm gives the global minimum of performance criterion due
to the fact that for the stationary process, the mean square
error is the quadratic form of weight vectors, but for the re-
cursive adaptive filter, local minima may be found instead of the
global minimum. The computer simulation shows that for the examples
considered here the algorithms presented learn the
statistics of signal and adapt. Several points can be observed
through the experimental results [see Figure R-l through R-12]
.
1) All the algorithms presented here give a satisfactory
result after the transients die out even though they are
not optimal.
105
2) The algorithm which employs the finite point average
square error for the gradient estimates gives more
rapid convergence than Feintuck's algorithm. The
possible reasons may be due to the fact that the output
information is fed back and used for the filter coef-
ficients updating process and the sensitivity information
propagates through the recursive equation as the iteration
proceeds, while Feintuck's algorithm discards the sensi-
tivity information.
3) The algorithm developed here gives the best results among
the various algorithms presented at the expense of
complex hardware. Note that the lEquired number of addi-
tional sensitivity filters (equations (3-47), (3-48))
would be the number of filter coefficients, and due to the
L point averaging process, additional storage elements are
also needed. The possible reason for the good results
may be due to the fact that the averaging process [equation
(3-4 0)] for the gradient estimate gives a smaller
error between true gradient and estimated
gradients than the gradient estimate based on instantaneous
square error does}while both give unbiased gradient
estimates
.
Due to the emerging interest in adaptive recursive
filters, further research on this subject may be worthwhile.
The following are left open for further research;
106
1) Comparison of steady state performance of the recur-
sive adaptive filter with the Kalman filtering tech-
nique. It would lead to a better understanding of the
performance of the recursive filter to express the
filter coefficients9
in terms of steady-state Kalman
filter gains.
2) Mathematical derivation of the bound in step size of
the filter coefficient updating process for convergence
and stability. It is believed that this bound may be at
tainedby setting up the constraints first such that the
value of performance criterion decreases mono-
tomically to a minimum as the iteration progresses.
3
)
Modification of the algorithms for the case that . partial
statistics of signal or noise are available apriori.
4
)
Derivation of the algorithm based on a different per-
formance criterion such as maximum likelihood ratio,
maximum signal to noise ratio, etc.
5) Derivation of the algorithm based on the different mini-
mization techniques such as Newton's method or Fletcher-
Powell methods, etc., for a given performance criterion.
107
V
LIST OF REFERENCES
Widrow, B., "Adaptive Filters," Aspects of Network andSystem Theory, R. E. Kalman and N. DeClaris, Eds., NewYork, Holt, Rinehart and Winston, 1970.
Feintuck, P. L., "An Adaptive Recursive LMS Filter,"Proceedings of IEEE . November 1976.
White, S. A., "An Adaptive Recursive Filter," Proceed-ings 9th Asilomar Conference on Circuits, Systems andComputers , November 1975.
Widrow, B., et al, "Adaptive Noise Cancelling: Principlesand Applications," Proceedings of IEEE , December 1975.
Blake, Ian F., and Thomas, John B., "The Linear RandomProcess," Proceedings of IEEE, October 1968.
Makhoul , John, "Linear Prediction: A Tutorial Review,"Proceedings of IEEE , April 1975.
7. Papoulis, A., "Probability, Random Variables and Sto-chastic Process," New York, McGraw-Hill, 1965.
8. Weiner, N. , "The Extrapolation, Interpolation andSmoothing of Stationary Time Series with EngineeringApplications," John Wiley and Sons, Inc. New York,1949.
9. Kalman, R. E., "A New Approach to Linear Filtering onPrediction Problems," ASME Trans., J. Basic Eng. ,
Vol. 83, March 1960, pp. 34-35.
108
INITIAL DISTRIBUTION LIST
No. Copies
1. Defense Documentation Center 2
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2. Library, Code 0142 2
Naval Postgraduate SchoolMonterey, California 93940
3. Department Chairman, Code 62 1
Department of Electrical EngineeringNaval Postgraduate SchoolMonterey, California 93940
4. Professor Sydney R. Parker, Code 62Px(Thesis Advisor) 1
Department of Electrical EngineeringNaval Postgraduate SchoolMonterey, California 93940
5. Lieutenant Soon-Ju Ko , ROKN 2
732 -1, 1-DO 2-DONGCHEJU , CHEJUDO 590 KOREA
6. Professor Eli I. Jury 1Department of Electrical Engineering andComputer ScienceUniversity of CaliforniaBerkeley, CA 94720
7. Professor Sanjit K. Mitra 1Department of Electrical Engineeringand Computer ScienceUniversity of CaliforniaSanta Barbara, CA 93106
8. Division of Personal Education and Training 2
Republic of Korea Naval HeadquartersDAEBANG-DONG YOUNGDEUNGPO-KUSEOUL 715-18, KOREA
9. Baruch, Even-or 1SMC 2190Naval Postgraduate SchoolMonterey, CA 93940
10. Pereira, Jose 1SMC 2610Naval Postgraduate SchoolMonterey, CA 93940
109
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