IEEE TRANSACTIONS ON AUTOMATIC CONTROL, VOL. 55, NO.

11, NOVEMBER 2010

A Computationally Efficient Adaptive IIR Solution

to Active Noise and Vibration Control Systems
Allahyar Montazeri and Javad Poshtan

AbstractIn spite of special advantages of IIR filters in active noise and

vibration control (ANVC) applications, the multimodal error surface and
instability problem of adaptive IIR filters has prevented its extensive use. To
alleviate these problems, in this paper, a new RLS-based fast array adaptive IIR filters in ANVC applications is proposed. The algorithm is developed with slow adaptation assumption and by transforming the active noise
and vibration control problem to an output-error identification problem.
By derivation of the fast-array equivalent form both computational complexity and numerical stability of the proposed algorithm are improved.
The geometrical illustration of the algorithm, in a simple case, is also given
to unify and complete its mathematical formulation. In spite of low computational complexity of the order ( ), simulation results confirm high
convergence speed of the proposed algorithm and also its ability to reach
the lower minimum mean square error in comparison with commonly used
adaptive IIR algorithms in ANVC systems.
Index TermsAdaptive filters, IIR digital filter stability, noise and vibration control.

I. INTRODUCTION
A large group of adaptive algorithms used in active noise and vibration control (ANVC) applications use FIR filters as the controller;
however, there are many situations in which the use of IIR filters is of
interest. Using IIR filters in ANVC applications was first proposed in
[1], in which the FuLMS algorithm was used for adaptation of filter
weights. There are two main problems with the use of IIR-based adaptive filters in ANVC applications: first, their stability, and second, the
slow convergence rate especially in a multichannel case. Some sufficient conditions for stability and convergence of the FuLMS algorithm
using the ODE method are derived in [2]. It is shown in [3] that the
perfect cancellation assumption of the analysis performed in [2] is not
necessary and it can be relaxed in some sense. To improve the FuLMS
algorithm, some other forms such as FvLMS algorithm [4], and Lattice
IIR filters are recently introduced in [5]. However, all of these algorithms suffer from slow adaptation rate, and cannot ensure convergence
to the global minimum of performance surface. These drawbacks were
the main motivation for using the equation-error method in [6] to derive
an algorithm for which global convergence is assured. Nevertheless, in
the equation-error algorithms, large deviation from global minimum
may occur when measurement noise exists or the order of control filter
is not sufficient. Considering these problems, a SteiglitzMcbride type
adaptive IIR algorithm is proposed in [7], but the stability of the algorithm is assumed beforehand, and there is no guarantee that the poles
of IIR filter in this algorithm do not move outside the unit circle. Other
than the ODE method, another approach less addressed in the literature

Manuscript received February 22, 2008; revised December 11, 2008,

September 01, 2009, and November 24, 2009; accepted March 08, 2010.
Date of publication August 16, 2010; date of current version November 03,
2010. This work was supported by the Signal and System Model Laboratory
(Control Division), Iran University of Science and Technology. Recommended
by Associate Editor J.-F. Zhang.
The authors are with the Electrical Engineering Department, Iran University
of Science and Technology, Tehran 16846, Iran (e-mail: amontazeri@iust.ac.ir;
jposhtan@iust.c.ir).
Color versions of one or more of the figures in this paper are available online
at http://ieeexplore.ieee.org.
Digital Object Identifier 10.1109/TAC.2010.2067670

2671

for the stability analysis of adaptive IIR filters is based on hyperstability theory. This theory is used to analyze the stability of nonlinear
and time-varying closed-loop systems. Although hyperstability theory
roots in control system community [8], its application in signal processing community has also been reported as HARF and SHARF algorithms in [9] and [10]. The use of SHARF and HARF algorithms in
ANVC applications is reported in [11][13]; however, their design and
analysis are not based on a rigorous theory, and a large number of challenging points are to be further investigated. This is effectively due to
the existence of the secondary-path transfer function in ANVC systems
in comparison with the conventional output-error identification problems.
Another approach proposed in the literature to improve the convergence rate of adaptive filters in ANVC applications is the use of
RLS-type algorithms with FIR filter structures [14], [15]. This is due to
this property of RLS-type algorithms that their convergence behavior
is quite independent of the statistics of the incident noise or vibration signal. The main problem in using plain RLS-type algorithms in
ANVC applications is that they suffer from numerical instability due to
finite precision computations. Besides, the computational complexity
of RLS-type algorithms is of order O(n2 ), where n is the length of
the control filter. To overcome these difficulties, fast RLS adaptive
filtering algorithms are proposed in the literature. Examples of such
fast schemes include the fast a posteriori error sequential technique
(FAEST) [16], the fast transversal filter algorithm (FTF) [17], and
array-based RLS filtering methods [18], [19]. In ANVC applications
the FTF algorithm is proposed in [15] to reduce the computational
complexity of RLS-type algorithms to the order of O(n). However,
the comparison study of [20] shows that the performance of the FTF
implementation of RLS algorithms used in ANVC applications is
degraded in comparison with the original RLS algorithm. The use of
array-based methods in ANVC applications is also reported in [21].
Considering this review, in this paper a new fast array RLS-type
adaptive IIR filter which is computationally efficient for real-time implementations, and exhibits faster convergence rate and lower minimum mean square error in comparison with classic adaptive IIR algorithms is introduced in ANVC applications. Design and stability analysis of the original algorithm has been performed in [22] using Popov
hyperstability theory. In fact, it has been shown that under certain SPR
conditions the global convergence of the proposed adaptive IIR filter is
guaranteed, and hence unlike gradient descent type adaptive IIR filters,
it will not be trapped into local minima. It is worthwhile to mention
that the proposed approach, in contrast to statistical ODE method used
in the literature, is established based on quite a deterministic framework, and hence no statistical hypothesis needs to be considered in
the derivation and analysis of the algorithm. After this introduction,
in Section II the RLS-type adaptive IIR algorithm proposed in [22] is
reviewed briefly. Since the computational complexity of the proposed
algorithm is of order O(n2 ), fast array implementation of this adaptive IIR filter along with its geometrical interpretation is developed in
Section III. To show the effectiveness of the proposed solution, the performance of the algorithm is compared with respect to FuLMS and
SHARF IIR adaptive algorithms in Section IV.
II. THE PROPOSED ADAPTIVE IIR FILTER FOR ANVC SYSTEMS
The block diagram of a typical ANVC system is shown in Fig. 1.
It is assumed in this figure that Gdw (q ) 2 RH , Gyu (q ) 2 RH ,
and Grw (q ) 2 RH . The aim is to adapt the coefficients of the IIR
filter C (q; n) such that the sum of squares of a posteriori errors in the
error microphone is minimized. By following the approach presented in
[22], the proposed algorithm for adapting the weights of an IIR filter in

0018-9286/$26.00 2010 IEEE

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IEEE TRANSACTIONS ON AUTOMATIC CONTROL, VOL. 55, NO. 11, NOVEMBER 2010

III. FAST RLS ARRAY IMPLEMENTATION

In order to implement a fast array form of the algorithm mentioned
in Section II, the first thing to be noted is the structure of data in the
^f (n). By splitting the regression vector into ^ 1f (n)
regression vector 
^ 2f (n)
containing samples of the previous outputs of the filter, and 
containing samples of the reference signal, the shift structure of the
regressor for two successive sampling times will be captured by noting
the following relation:

^ T2f (n)
0u^f (n); '^ T1f (n); r^f (n +1); '
Fig. 1. Typical ANVC system.
TABLE I
SUMMARY OF THE PROPOSED ALGORITHM

= '^ T1f (n +1); 0u^f (n 0 nA ); '^ T2f (n +1); r^f (n 0 nB )

(1)

'^ 1f (n) = [0u^f (n 0 1); 1 1 1 1 1 1 ; 0u^f (n 0 nA )]T ;

'^ 2f (n) = [^rf (n); 1 1 1 1 1 1 ; r^f (n 0 nB )]T :

(2)

where

Now assuming that 1 and 2 are constant and independent of time

index n, lets define

(n + 1) =

1+




g(n + 1) =

1
0
1
^ (n + 1)F (n)^'f (n + 1) ;
1
'f (n + 1)
0
1 F (n)^
T
'f (n + 1)
'^ f (n + 1)F (n)^

'Tf

1 + 

g 0 (n + 1) = 2 g(n + 1); 0 (n + 1) = 1 (n + 1):
1

(3)
(4)

Then we will get

2
F (n)^
'f (n +1);
21

001 (n +1)=1+ 2 '^ Tf (n +1)F (n)^
'f (n +1):
1

g0 (n +1) 001 (n +1)=

(5)

By partitioning g 0 (n) and the adaptation gain matrix F (n) as

Fig. 1 at time n +1 is summarized in Table I. Here it is assumed that an

^ yu (q) is available. In
estimation of secondary-path transfer function G
the algorithm the two weighting sequences 1 (n) and 2 (n) determine
how adaptation gain changes in time, and are especially useful for nonstationary environments. Moreover, e(n) and "^(n) are a priori and a
posteriori errors, and v0 (n) is the filtered a priori error. In the fifth row
of Table I, hi is the coefficient of an FIR filter used to filter a priori
and a posteriori errors. The conditions for the stability of the proposed
algorithm are investigated in [22] and are not repeated here due to lack
of space. In fact, it was shown that by slow adaptation assumption it is
possible to transform the ANVC problem in Fig. 1 to an output error
identification problem. Hence it is reasonable to make the hypothesis
that, such a value of parameter vector for the controller C (q; n) exists,
i.e., 9 ; "^(n+1)j^(n)= = 0. If the sufficient conditions for the stability
of the algorithm as stated in [22] holds, then the difference equation in
step 7 of Table I, with the input g (n + 1)v0 (n + 1) defined in other
steps, will be asymptotically stable and converges to the equilibrium
point  .
Remark 1: Since the adaptive feedforward control structure in Fig. 1
is basically a noise cancellation problem, the convergence of the parameters of the controller to the true parameter is not a matter. This
will allow the algorithm to be used without any restriction on the persistent excitation condition of the reference signal.

F (n) =

F11 (n)n 2n
F21 (n)(n +1)2n

F12 (n)n 2(n +1)

F22 (n)(n +1)2(n +1)

(6)

we may write the update equations in (5) in a more suitable array form
using lemma 1.
Lemma 1: The time-update equations in (5) can be rewritten in the
following forms:

g10 (n + 1) 001 (n + 1)

g20 (n + 1) 001 (n + 1)

g10 (n) 001 (n)

g20 (n) 001 (n)

0u^f (n)

+ 01 1 F (n) r^f'^(1nf (+n)1)

'^ 2f (n)
0u^f (n) T
'^ 1f (n)
001 (n + 1) = 001 (n) +
r^f (n + 1)
'^ 2f (n)
0u^f (n)
'^ 1f (n)
2 F (n)
r^f (n + 1)
'^ 2f (n)

(7)

(8)

IEEE TRANSACTIONS ON AUTOMATIC CONTROL, VOL. 55, NO. 11, NOVEMBER 2010

2673

where
2
F (n) =
1

F11 (n)

F12 (n)

F21 (n)

F22 (n)

F11 (n

F21 (n

0 1)
0 1)

F12 (n

F22 (n

0 1)

0 1)

Proof: The proof is not given here due to the lack of space. It is
^ (n) in both sides of the
obtained by partitioning F (n), g 0 (n+1), and 
f
expressions in (5), and subtracting them for two consecutive indices.
Equations (7) and (8) in lemma1 show that the update of the gain
vector g 0 (n) 001 (n) as well as 001 (n) depends only on the value of
F (n), and hence for fast implementation of the algorithm it is required
to compute the time-update of F (n) with an order of O(n) operations.
For this purpose we start by the initialization of the algorithm with the
following parameters:
1

Theorem 1: If F (n) can be factored as 21 Ln Sn Ln at time n,

where Ln is a matrix of rank four and of dimension (M + 2) 2 4,
and Sn is a diagonal signature matrix with +1 and 01 in the principal
diagonal, then the following properties holds for F (n + 1) at time
n + 1:
T
1) F (n + 1) can be factored as 21 Ln+1 Sn+1 Ln+1 where Ln+1 is
a matrix of rank four and of dimension (M + 2) 2 4, and relates
to Ln with an array algorithm.
2) The signature matrix Sn+1 will remain unchanged and coincides
with Sn .
Proof: The proof is obtained by construction of the array algorithm. By noting the assumption of theorem 1, it can be seen that relations (7) and (8) are in the norm-preserving and inner-product matrix
form, and hence there is a J-unitary transformation 2 such that
A
D

F (0) = 5

01
511
01
521

01
512
01
522

01
01
512 = 521 = 0
21
..
01
..
511 = 
.
.

A =

; F ( 1) = 1 F (0);

D=

..

0
n +n +2

(9)

1

In this case, F (0) will be initialized by a matrix of M + 2 by M + 2

with M = nA + nB +1, whose rank is four and has two positive eigenvalues and two negative eigenvalues. This matrix can be for example
as follows:

2
F (0) = 1 2

111

..
.

..

..

..
.

..
.

1 Ln ;

..

0
0

(10)

(13)

g20 (n + 1) 00(1=2) (n + 1)
0

and we can write

00

g2 (n)

00

0
(n)

g1 (n)

00

(n)

0
(n)

g2 (n)

00

+ Ln Sn Ln

(n)
=YY

..
.

; E = Ln :

(n)

X = 00(1=2) (n + 1), and Y =

1 1 1 0M1

00

g10 (n + 1) 00(1=2) (n + 1)

g1 (n)

T
+ ZSn Z :

(14)

Multiplying two vectors in the left and right hand sides of (14) and
using (5), it can be shown that

F (0) = 1 L0 S0 L0 ;
1

..
.

2

..
.

..
.

..
.

..
.

1

0
1

S0 =

By squaring both sides of (12), and substituting (13) into (12), it can
be easily seen that

Since F (0) is a matrix with rank four it can be factored as

L0 =

g10 (n) 00 (n)

111
111

0
n
1

..
.

111

..
.

n
1
0

111

..
.

111 0
111

(12)

' 1f (n)
rf (n + 1)
' 2f (n)

(n); B =

g2 (n)

n +1

111

..

00

1

..
.

..
.

0uf (n)

0
0
n +2
1

X
Y

111

01
522 = 

2=

where

g (0) = g (0) = 0; (0) = 1 ; (0) = 1;

0

B
E

1

0
0

01

01

ZSn Z

..
.

..
.

..
.

1

1

F (n + 1):

(15)

Expression (15) tells that F (n + 1), as for F (n), can be factorized

by a matrix of rank four and the signature matrix Sn which is remained

(11)

where L0 is a matrix with the size of (M + 2) 2 4, and S0 is the initial

signature matrix. Considering the initialization above and theorem 2
stated below, the array implementation of the algorithm proposed in
the previous section can be derived.

1
unchanged. As a result, it can be deduced that Z = 0
1 Ln+1 and
Sn+1 = Sn .
Considering the discussions above, the fast-array implementation of
the proposed algorithm is summurized in Table II. The computational
complexity of this algorithm is in order of O(M ) where M is the
number of coefficients of the IIR control filter.
Remark 2: The algorithm described in Section II and its fast-array
form in Section III can be described based on the geometrical representation of the RLS algorithm. This interpretation is based on [23]
and due to lack of space, only some main points are explained here.
This will help to unify the presentation of the paper. For illustration
purpose, the case with one coefficient filter is considered. Besides, for

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IEEE TRANSACTIONS ON AUTOMATIC CONTROL, VOL. 55, NO. 11, NOVEMBER 2010

TABLE II
SUMMARY OF THE PROPOSED RLS-BASED FAST-ARRAY ADAPTIVE IIR FILTER

Fig. 2. Geometrical view of the proposed algorithm in the simplified form.

and noting the fact that, the LS estimation of d0 (3) is obtained by projecting it on the basis vector ^
rf (3), a geometrical view for updating
the weights of the IIR filter in the simplified form (i.e., na = nb = 0,
H (q) = 1) similar to the approach of [23] is shown in Fig. 2. Although
this figure does not reflect all the properties of the proposed algorithm,
it helps to have an idea how fast-array form of the algorithm may be
related to its original form. We can write from Fig. 2

b0 (3)^rf (3) = b0 (2)^rf (3) +  (3)

(17)

where  (3) is obtained by projecting  (3) on the ^

rf (3) and using the
similarity of two triangles highlighted by dashed lines. In this case one
can obtain

 (3) = g(3)e(3j2) = g(3)e(3):

(18)

For the proposed algorithm, the update of gain vector g (3) can be obtained in step 3 of Table II. This relation for the simplified case studied
here can be written as

00

(3) [^rf (4)^rf (3)] L3

0
g(3)

1 L3
0

23 =

(4)
 g(4)
00
0

[0 0]
1 1 L4
0

(19)

where  is some known scaling factor. A geometrical interpretation of

this matrix transformation is that by having the values of the first row
of the matrix in the left hand side of (19), the rotation matrix 23 should
be determined in a way to project this row vector on the x axis. Then
by applying the same hyperbolic rotation to the second and third rows
and doing some scaling, the value of g (4) will be determined. It can be
seen that only the information used to update 00(1=2) (3) is required
to determine the angle of rotation.
IV. SIMULATION RESULTS
simplicity the time index is started at n = 1 and ends at
defining the prediction errors in the vector form

e(3) = d0 (3) + b0 (3)^rf (3);

e(3) =

^rf (3) =

e(1j3)
d (1)
e(2j3) ; d (3) = d (2) ;
d (3)
e(3j3)
r^f (1)
r^f (2)
r^f (3)
0

= 3. By

0
0
0

(16)

The performance of the proposed algorithm is evaluated using numerical simulations of the algorithm in a fast-array form. To illustrate
that the convergence rate of the proposed algorithm (unlike the filtered-LMS type algorithms) is not affected by statistical properties of
the reference signal, three kinds of input signal is assumed as the incident unwanted noise. The incident noises are generated by passing
a zero-mean unit-variance white noise through the following filter for
values of  = 0, 0.5, 0.75:

L(q) =

2
1 0 q 1 :
0

(20)

IEEE TRANSACTIONS ON AUTOMATIC CONTROL, VOL. 55, NO. 11, NOVEMBER 2010

Fig. 3. Comparing the performance of three algorithms.

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Fig. 4. Convergence behavior of two coefficients of IIR controller.

The signal w(n) becomes more colored for values of  close to

1. The performance of the proposed algorithm for these three signals
is compared with the FuLMS algorithm, and also the FuLMS with
SHARF smoothing filter. Here, the example studied in [7] and [11]
(without delay in the primary and secondary path) is considered for
simulations. The transfer functions in the ANVC system shown in Fig.
1 are

q09 0 0:4q010 ; G (q) = 1

Gdw (q) = 1 001:05
:1314q01 + 0:25q02 rw
0
5
Gyu (q) = q + 0:7q06 + 0:6q07 :

(21)

In all simulations the results are plotted after ensemble averaging of 100
different realizations of the incident unwanted signal. In all simulations
the estimated secondary path is equal to the real one to avoid any ambiguity. The smoothing filter H (q ) is an FIR filter with two coefficients
h0 = 1 and h1 = 00:85, so that the positive real condition necessary for the stability of the algorithm is satisfied. The controller is selected an IIR filter with a 9th-order numerator and an 8th-order denominator. The performance of the algorithm is compared with FuLMS, and
FuLMS with SHARF smoothing filter for white a noise input signal.
The initial value of the gain matrix is chosen F (1) = 5I , and the
weighting sequences 1 and 2 are fixed equal to one. The mean square
error for three algorithms as well as the convergence behavior of two
coefficients of the control filter is plotted in Figs. 3 and 4. As can be seen
in these figures, although FuLMS and FuLMS with SHARF smoothing
filter exhibit convergence to local optimum, the proposed algorithm
converges to the global optimum with a very faster convergence rate.
The performance of the proposed algorithm in case of three types of
unwanted disturbances is shown in Fig. 5. In this example, the convergence rate has decreased slightly by increasing the autocorrelation of
the input signal, but this is compensated by changing the values of initial gain matrix. The values of initial gain matrix for  = 0, 0.5, 0.75
are selected F (1) = 5I; 7I; 9I respectively.
V. CONCLUSION
This paper deals with the development of a novel fast array adaptive
IIR filter in ANVC applications. Since the proposed algorithm is derived in a quite deterministic framework, its convergence behavior does
not depend on the spectral contents of the incident noise or vibration.

Fig. 5. Convergence behavior of error signal for three different unwanted disturbances.

The design and analysis of the proposed algorithm is based on conversion of the original problem to an output-error identification problem,
and then its global convergence can be proved using the hyperstabilty
theory. The fast array form of the algorithm reduces its computational
complexity to the order of O(n). Besides, because of its matrix nature,
it has good numerical stability against round-off and finite precision
errors which is a necessity in real-time implementation of ANVC algorithms. The convergence speed as well as the achievable minimum
mean square error of the proposed algorithm is compared by numerical
simulations with the well known IIR adaptive filters in ANVC applications. The simulations approve the high speed of convergence and
small MMSE of the proposed algorithm in comparison with both the
FuLMS and FuLMS with SHARF smoothing filter algorithms.

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Correction to Output Feedback Stabilization of a

One-Dimensional Schrdinger Equation by
Boundary Observation with Time Delay
Bao-Zhu Guo and Kun-Yi Yang
In the paper [1], the proof of Theorem 2.1 is incorrectly stated for
the proof below the Equation (6) used wrongly the mean value theorem
for the complex function defined by the left side of Equation (6). The
correct statement and proof of Theorem 2.1 should be as follows.
Theorem 2.1: For any given k > 0, there are infinitely many posi0 such that the output feedback control u(t) = iky (t) is
tive 
unstable for the closed-loop system, which is an exponential stabilizing
controller with  = 0.
Proof: Let  be an eigenvalue of the closed-loop system under the
proportional output feedback control u(t) = iky (t), k > 0. Then 
satisfies the characteristic equation

pi sinh(pi) + ik 1 cosh(pi) 1 e0 = 0:

(1)

Set

i! =

pi or  = i! :
2

(2)

The characteristic equation becomes

F (!;  ) = i! sinh(i! ) + ik cosh(i! ) e0i! 

= 0 ! sin(! ) + ik cos(! ) 1 e0i!  = 0:

(3)

Suppose that ! and  satisfy ! 2  = =2. Then (3) becomes

! tan(! ) = k:
Let f (! ) = ! tan(! ). For any integer n > 0, f is continuous in
[n; (=2) + n  ], where  > 0 is chosen small enough so that
f ((=2)  ) > k . Since f (n ) = 0; f (=2 + n  ) > f (=2
 ) > k . By the mean value theorem, there exists an !n (n; (=2)+
n  ) such that f (!n ) = k for any n > 0.
Therefore, we can find !n (n; =2 + n  ), n > 0 and n
with !n2 n = =2 such that F (!n ; n ) = 0. Obviously, n
0 as
. The proof is complete.
n

!1

0
2

REFERENCES
[1] B. Z. Guo and K. Y. Yang, Output feedback stabilization of a onedimensional Schrodinger equation by boundary observation with time
delay, IEEE Trans. Autom. Control, vol. 55, no. 5, pp. 12261232,
May 2010.

Manuscript received May 01, 2010. First published July 26, 2010; current
version published November 03, 2010.
B.-Z. Guo is with the Academy of Mathematics and Systems Science,
Academia Sinica, Beijing 100190, China, the School of Mathematical
Sciences, Shanxi University, Taiyuan 030006, China, and the School of Computational and Applied Mathematics, University of the Witwatersrand, South
Africa. (e-mail: bzguo@iss.ac.cn).
K.-Y. Yang is with the College of Science, North China University of Technology, Beijing 100041, China (e-mail: kyy@amss.ac.cn).
Digital Object Identifier 10.1109/TAC.2010.2051070

0018-9286/$26.00 2010 IEEE