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RLS algortimo

RLS algortimo

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Jacob Benesty

INRS-EMT, Université du Québec, 800 de la Gauchetière Ouest, Suite 6900, Montréal, Québec, Canada H5A 1K6

Email: benesty@inrs-emt.uquebec.ca

Tomas Gänsler

Agere Systems Inc., 1110 American Parkway NE, Allentown, PA 18109-3229, USA

Email: gaensler@agere.com

Received 21 July 2003; Revised 9 October 2003; Recommended for Publication by Hideaki Sakai

The recursive least squares (RLS) algorithm is one of the most popular adaptive algorithms that can be found in the literature, due

to the fact that it is easily and exactly derived from the normal equations. In this paper, we give another interpretation of the RLS

algorithm and show the importance of linear interpolation error energies in the RLS structure. We also give a very eﬃcient way

to recursively estimate the condition number of the input signal covariance matrix thanks to fast versions of the RLS algorithm.

Finally, we quantify the misalignment of the RLS algorithm with respect to the condition number.

Keywords and phrases: adaptive algorithms, normal equations, RLS, fast RLS, condition number, linear interpolation.

estimate the condition number. The proposed method is very

Adaptive algorithms play a very important role in many eﬃcient when combined with the FRLS algorithm; it requires

diverse applications such as communications, acoustics, only L more multiplications per iteration, where L is the

speech, radar, sonar, seismology, and biomedical engineer- length of the adaptive filter. Finally, we show exactly how the

ing [1, 2, 3, 4]. Among the most well-known adaptive filters misalignment of the RLS algorithm is aﬀected by the con-

are the recursive least squares (RLS) and fast RLS (FRLS) al- dition number, output signal-to-noise ratio (SNR), and pa-

gorithms. The latter is a computationally fast version of the rameter choice.

former. Even though the RLS is not as widely used in prac-

tice as the least mean square (LMS), it has a very significant

theoretical interest since it belongs to the Kalman filters fam- 2. RLS ALGORITHM

ily [5]. Also, many adaptive algorithms (including the LMS)

In this section, we briefly derive the classical RLS algorithm

can be seen as approximations of the RLS. Therefore, there

in a system identification context. We try to estimate the im-

is always a need to interpret and understand in new ways the

pulse response of an unknown, linear, and time-invariant

diﬀerent variables that are built in the RLS algorithm.

system by using the least squares method.

The convergence rate, the misalignment, and the numer-

We define the a priori error signal e(n) at time n as fol-

ical stability of adaptive algorithms depend on the condition

lows:

number of the input signal covariance matrix. The higher

this condition number is, the slower the convergence rate is

and/or the less stable the algorithm is. For ill-conditioned in- e(n) = y(n) − ŷ(n), (1)

put signals (like speech), the LMS converges very slowly and

the stability and the misalignment of the FRLS are more af- where

fected. Thus, there is a need to compute the condition num-

ber in order to monitor the behavior of adaptive filters. Un- y(n) = hTt x(n) + w(n) (2)

fortunately, there are no simple ways to estimate this condi-

tion number. is the system output,

The objective of this paper is threefold. We first give an-

other interpretation of the RLS algorithm and show the im- T

ht = ht,0 ht,1 · · · ht,L−1 (3)

portance of linear interpolation error energies in the RLS

332 EURASIP Journal on Applied Signal Processing

is the true (subscript t) impulse response of the system, the 3. AN RLS ALGORITHM BASED ON THE

superscript T denotes the transpose of a vector or a matrix, INTERPOLATION ERRORS

T In this section, we show another way to write the RLS algo-

x(n) = x(n) x(n − 1) · · · x(n − L + 1) (4) rithm. This new formulation, based on linear interpolation,

gives a better insight of the adaptive algorithm structure.

is a vector containing the last L samples of the input signal x, We would like to minimize the criterion [6, 7]:

and w is a white Gaussian noise (uncorrelated with x) with 2

n

L

−1

variance σw2 . In (1), n−m

Jint,i (n) = λ − cil (n)x(m − l)

m=0 l =0

ŷ(n) = hT (n − 1)x(n) (5) n

2 (14)

= λn−m

− cTi (n)x(m)

is the model filter output and m=0

= cTi (n)R(n)ci (n),

T

h(n − 1) = h0 (n − 1) h1 (n − 1) · · · hL−1 (n − 1) (6) with the constraint

We also define the popular RLS error criterion with re-

spect to the modelling filter: where

T

n

2 ci (n) = ci0 (n) ci1 (n) · · · ci(L−1) (n) (16)

n−m

JLS (n) = λ y(m) − hT (n)x(m) , (7)

m=0 is the ith (0 ≤ i ≤ L − 1) interpolator of the signal x(n) and

where λ (0 < λ < 1) is a forgetting factor. The minimization T

ui = 0 · · · 0 1 0 · · · 0 (17)

of (7) leads to the normal equations

is a vector of length L, where its ith component is equal to one

R(n)h(n) = r(n), (8)

and all others are zero. By using the Lagrange multipliers, it

where is easy to see that the solution to this optimization problem

is

n

R(n) = λn−m x(m)xT (m) (9) R(n)ci (n) = −Ei (n)ui , (18)

m=0

where

is an estimate of the input signal covariance matrix and

1

n

Ei (n) = cTi (n)R(n)ci (n) = (19)

uTi R−1 (n)ui

r(n) = λn−m x(m)y(m) (10)

m=0

is the interpolation error energy.

is an estimate of the cross-correlation vector between x and From (18) we find

y. ci (n)

From the normal equations (8), we easily derive the clas- − = R−1 (n)ui , (20)

Ei (n)

sical update for the RLS algorithm [1, 3]:

hence the ith column of R−1 (n) is −ci (n)/Ei (n). We can now

e(n) = y(n) − hT (n − 1)x(n), deduce that R−1 (n) can be factorized as follows:

(11)

h(n) = h(n − 1) + R−1 (n)x(n)e(n).

1 −c10 (n)

· · · −c(L−1)0 (n)

−c01 (n) 1 · · · −c(L−1)1 (n)

A fast version of this algorithm can be deduced by com- R−1 (n) = .. .. .. ..

. . . .

puting recursively the a priori Kalman gain vector k (n) =

−c0(L−1) (n) −c1(L−1) (n) · · · 1

R−1 (n − 1)x(n) [1]. The a posteriori Kalman gain vector

k(n) = R−1 (n)x(n) is related to k (n) by [1]: 1

E (n) 0 · · · 0

0 1

0 · · · 0

k(n) = λ−1 ϕ(n)k (n), (12)

× .. E1 (n)

.. ..

. ..

. . .

where 1

0 0 ···

EL−1 (n)

λ = CT (n)D−1

ϕ(n) = . (13) e (n).

λ + xT (n)R−1 (n − 1)x(n) (21)

New Insights into the RLS Algorithm 333

Furthermore, since R−1 (n) is a symmetric matrix, (21) can Now, if we premultiply both sides of (27) by uTi , we can easily

be written as find that

1 Ei (n) = λEi (n − 1) + ei (n)εi (n). (28)

E (n) 0 ··· 0

0

1

0 ··· 0 This means that the interpolation error energy can be com-

E1 (n)

R−1 (n) =

puted recursively. This relation is well known for the forward

.. .. .. ..

. . . . (i = 0) and backward (i = L) predictors [1]. It is used to

1 obtain fast versions of the RLS algorithm.

0 0 ···

EL−1 (n) Also, the interpolator vectors can be computed recur-

sively:

1 −c01 (n)

· · · −c0(L−1) (n)

−c (n) 1 · · · −c1(L−1) (n)

10 1

×

. . .. ..

ci (n) = ci (n − 1) + k(n)ei (n) . (29)

.

. .

. . . 1 − ki (n)ei (n)

−c(L−1)0 (n) −c(L−1)1 (n) · · · 1

If we premultiply both sides of (29) by −xT (n), we obtain a

−1

= De (n)C(n). relation between the a priori and a posteriori interpolation

(22) error signals:

The first and last columns of R−1 (n) contain, respectively, εi (n) ϕ(n)

= . (30)

the normalized forward and backward predictors and all the ei (n) 1 − ki (n)ei (n)

columns between contain the normalized interpolators.

We define, respectively, the a priori and a posteriori in- We now give another interpretation of the RLS algorithm:

terpolation error signals as εl (n)e(n)

hl (n) = hl (n − 1) +

ei (n) = −cTi (n − 1)x(n), εi (n) = −cTi (n)x(n). (23) El (n)

el (n)e(n)

Using expression (22), we now have an interesting inter- = hl (n − 1) + ϕ(n) , l = 0, 1, . . . , L − 1.

λEl (n − 1)

pretation of the a priori and a posteriori Kalman gain vec- (31)

tors:

In Sections 4 and 5, we will show how the linear interpo-

k (n) lation error energies appear naturally in the condition num-

= R−1 (n − 1)x(n) ber formulation.

e0 (n) e1 (n) eL−1 (n) T

= ··· ,

E0 (n − 1) E1 (n − 1) EL−1 (n − 1) 4. CONDITION NUMBER OF THE INPUT SIGNAL

(24) COVARIANCE MATRIX

k(n)

= R−1 (n)x(n) Usually, the condition number is computed by using the 2-

norm matrix. In the context of RLS equations, it is more con-

ε0 (n) ε1 (n) εL−1 (n) T

= ··· . venient to use a diﬀerent norm as explained below.

E0 (n) E1 (n) EL−1 (n) The covariance matrix R(n) is symmetric and positive

The ith component of the a priori (resp., a posteriori) definite. It can be diagonalized as follows:

Kalman gain vector is the ith a priori (resp., a posteriori) in-

QT (n)R(n)Q(n) = Λ(n), (32)

terpolation error signal normalized with the ith interpolation

error energy at time n − 1 (resp., n). where

Writing (18) at time n and n − 1, we obtain

QT (n)Q(n) = Q(n)QT (n) = I,

R(n)ci (n) λR(n − 1)ci (n − 1) (33)

− = ui = − . (25) Λ(n) = diag λ0 (n), λ1 (n), . . . , λL−1 (n) ,

Ei (n) λEi (n − 1)

Replacing λR(n − 1) in (25) by and 0 < λ0 (n) ≤ λ1 (n) ≤ · · · ≤ λL−1 (n). By definition, the

square root of R(n) is

λR(n − 1) = R(n) − x(n)xT (n), (26)

R1/2 (n) = Q(n)Λ1/2 (n)QT (n). (34)

we get

The condition number of a matrix R(n) is [8]

Ei (n)

ci (n) = ci (n − 1) + k(n)ei (n) . (27) χ R(n) = R(n)R−1 (n),

λEi (n − 1) (35)

334 EURASIP Journal on Applied Signal Processing

where · can be any matrix norm. Note that χ[R(n)] de- Also, since tr[R(n)] ≤ LλL−1 (n) and tr[R−1 (n)] ≤ L/λ0 (n),

pends on the underlying norm and the subscripts will be we obtain

used to distinguish the diﬀerent condition numbers. Usually,

we take the convention that χ[R(n)] = ∞ for a singular ma- −1

tr R(n) λL−1 (n)

tr R(n) tr R (n) ≤ L ≤ L2 . (43)

trix R(n). λ0 (n) λ0 (n)

Consider the following norm:

Therefore, we deduce that

1/2

1 T

R(n) = tr R (n)R(n) . (36) 1

E

L χ2 R(n) ≤ χE2 R1/2 (n) ≤ χ2 R(n) . (44)

L2

We can easily check that, indeed, ·E is a matrix norm since

for any real matrices A and B and a real scalar γ, the following According to the previous expression, χE2 [R1/2 (n)] is then

three conditions are satisfied: a measure of the condition number of the matrix R(n).

In Section 5, we will show how to recursively compute

(i) AE ≥ 0 and AE = 0 if and only if A = 0L×L , χE2 [R1/2 (n)].

(ii) A + BE ≤ AE + BE ,

(iii) γAE = |γ|AE .

5. RECURSIVE COMPUTATION OF

Also, the E-norm of the identity matrix is equal to one. THE CONDITION NUMBER

We have

The positive number R1/2 (n)2E can be easily calculated re-

1/2 cursively. Indeed, taking the trace of

1/2 1 L

−1

1/2 1

R (n) = tr R(n) = λl (n) ,

E

L L

l=0 R(n) = λR(n − 1) + x(n)xT (n), (45)

1/2

1 L 1

−1

−1/2 1 −1 1/2

R (n)E = tr R (n) = . we get

L L λ (n)

l=0 l

(37) tr R(n) = λ tr R(n − 1) + xT (n)x(n). (46)

Hence, the condition number of R1/2 (n) associated with ·E Therefore,

is

1/2 2 T

R (n) = λR1/2 (n − 1)2 + x (n)x(n) . (47)

χE R 1/2

(n) = R1/2 (n)E R−1/2 (n)E ≥ 1. (38) E E

L

If χ[R(n)] is large, then R(n) is said to be an ill- Note that the inner product xT (n)x(n) can also be computed

conditioned matrix. Note that this is a norm-dependent pro- in a recursive way with two multiplications only at each iter-

perty. However, according to [8], any two condition numbers ation.

χα [R(n)] and χβ [R(n)] are equivalent in that constants c1 and Now we need to determine R−1/2 (n)2E . Thanks to (22),

c2 can be found for which we find that

L

−1

c1 χα R(n) ≤ χβ R(n) ≤ c2 χα R(n) . (39) 1

tr R−1 (n) = . (48)

l=0

El (n)

For example, for the 1- and 2-norm matrices, we can show

[8] that

Using (24), we have

1 1

χ2 R(n) ≤ χ1 R(n) ≤ χ2 R(n) . (40) L

−1

L2 L el (n)εl (n)

kT (n)k (n) = , (49)

We now show the same principle for the E- and 2-norm l=0

El (n)El (n − 1)

matrices. We recall that and replacing in the previous expression:

λ (n)

χ2 R(n) = L−1 . (41) El (n) − λEl (n − 1) = el (n)εl (n), (50)

λ0 (n)

Since tr[R−1 (n)] ≥ 1/λ0 (n) and tr[R(n)] ≥ λL−1 (n), we we obtain

have

L

−1 1 L−1

λL−1 (n) 1

tr R(n) tr R (n) ≥

tr R(n)

−1

≥ . kT (n)k (n) = −λ . (51)

λ0 (n) λ0 (n)

(42)

l =0

El (n − 1) E (n)

l =0 l

New Insights into the RLS Algorithm 335

Thus,

Initialization.

L

−1

1 h(0) = k (0) = a(0) = b(0) = 0,

tr R−1 (n) =

l=0

El (n) α(0) = λ,

(52)

L

−1 Ea (0) = E0 , (positive constant),

1

= λ−1 − kT (n)k (n). L

El (n − 1) 1/2 2

R (0) = E0

−1

l=0 λ−l ,

E

L l=0

Finally, L−1

−1/2 2

R 1 l

−1/2 2 (0)E = λ.

R (n)E LE0 l=0

2 λ−1 ϕ(n)kT (n)k (n) Prediction.

= λ−1 R−1/2 (n − 1)E −

L ea (n) = x(n) − aT (n − 1)x(n − 1),

L

−1

α1 (n) = α(n − 1) + ea2 (n)/Ea (n − 1),

2 λ−1 ϕ(n) el2 (n)

= λ−1 R−1/2 (n − 1)E − .

L l=0

El (n − 1)

2

t(n) 0 1

= + e (n)/Ea (n − 1),

(53) m(n) k (n − 1) −a(n − 1) a

Ea (n) = λ Ea (n − 1) + ea2 (n)/α(n − 1) ,

By using (47) and (53), we see that we easily compute

χE2 [R1/2 (n)] recursively with only an order of L multiplica- a(n) = a(n − 1) + k (n − 1)ea (n)/α(n − 1),

tions per iteration given that k (n) is known. eb (n) = x(n − L) − bT (n − 1)x(n),

Note that we could have used the inverse of R(n),

k (n) = t(n) + b(n − 1)m(n),

R−1 (n) = λ−1 R−1 (n − 1) − λ−2 ϕ(n)k (n)kT (n), (54) α(n) = α1 (n) − eb (n)m(n),

b(n) = b(n − 1) + k (n)eb (n)/α(n).

to estimate R−1/2 (n)2E , but we have chosen here to use

Filtering.

the interpolation formulation to better understand the link

among all variables in the RLS algorithm, and especially to e(n) = y(n) − hT (n − 1)x(n),

emphasize the role of the interpolation error energies since

h(n) = h(n − 1) + k (n)e(n)/α(n).

tr[R−1 (n)] = Ll=−01 1/El (n), even though there are indirect

ways to compute this value. Clearly, everything can be writ- Condition Number.

ten in terms of El (n) and this formulation is more natural 1/2 2 T

for the condition number estimation. For example, in the ex- R (n) = λR1/2 (n − 1)2 + x (n)x(n) ,

E E

L

treme cases of an input signal close to a white noise or to a

−1/2 2 2 kT (n)k (n)

predictable process, the value maxl [El (n)]/ minl [El (n)] gives R (n)E = λ−1 R−1/2 (n − 1)E − ,

a good idea of the condition number of the corresponding Lα(n)

2 2

signal covariance matrix. χE2 R 1/2

(n) = R1/2 (n)E R−1/2 (n)E .

It is easy to combine the estimation of the condition

number with an FRLS algorithm. There exist several meth-

ods to compute the a priori Kalman gain vector k (n) in a Algorithm 1: The FRLS algorithm and estimation of the condition

very eﬃcient way. Once this gain vector is determined, the es- number.

timation of χE2 [R1/2 (n)] at each iteration follows immediately

with roughly L more multiplications. Algorithm 1 shows the

combination of an FRLS algorithm with the condition num-

ber estimation of the input signal covariance matrix. It can easily be shown, under certain conditions, that [9]

2 1

6. MISALIGNMENT AND CONDITION NUMBER E ht − h(n)2 ≈ σw2 tr R−1 (n) . (56)

2

We define the normalized misalignment in dB as follows:

Hence, we can write (56) in terms of the interpolation error

energies:

ht − h(n)2

m0 (n) = 10 log10 E 2

2

, (55)

ht

2 2 1 L−1

1

E ht − h(n)2 ≈ σw2 . (57)

where · 2 denotes the 2-norm vector. Equation (55) mea- 2 l=0 El (n)

sures the mismatch between the true impulse response and

the modelling filter. However, we are more interested here to write (56) in terms

336 EURASIP Journal on Applied Signal Processing

of the condition number. Indeed, we have Usually, we take for the exponential window

1/2 2

R (n) = 1 tr R(n) , λ=1−

1

, (64)

E

L K0 L

L−1 (58)

−1/2 2

R 1 1 where K0 ≥ 3. Also, the second term in (63) represents

(n)E = .

L l=0

El (n) roughly the inverse output SNR in dB. We can then rewrite

(63) as follows:

But

m0 (n) ≈ −10 log10 2K0 − oSNR + 10 log10 χE2 R1/2 (n) .

n

(65)

tr R(n) = tr λ n−m

x(m)x (m)

T

m=0 For example, if we take K0 = 5 and an output SNR (oSNR)

n of 39 dB, we obtain

(59)

= λn−m xT (m)x(m)

m=0 m0 (n) ≈ −49 + 10 log10 χE2 R1/2 (n) . (66)

L 2

≈ σ , If the input signal is a white noise, χE2 [R1/2 (n)] = 1, then

1−λ x

m0 (n) ≈ −49 dB. This will be confirmed in the following sec-

for n large and for a stationary signal x with power σx2 . The tion.

condition number is then

L

−1 7. SIMULATIONS

σx2 1

χE2 1/2

R (n) ≈ , (60) In this section, we present some results on the condition

(1 − λ)L l=0 El (n)

number estimation and how this number aﬀects the mis-

alignment in a system identification context. We try to es-

and expression (57) becomes timate an impulse response ht of length L = 512. The same

length is used for the adaptive filter h(n). We run the FRLS al-

2 (1 − λ)L σw2 2 1/2

E ht − h(n)2 ≈ χ R (n) . (61) gorithm with a forgetting factor λ = 1 − 1/(5L). Performance

2 σx2 E of the estimation is measured by means of the normalized

misalignment (55). The input signal x(n) is a speech signal

If we divide both sides of (61) by ht 22 , we get sampled at 8 kHz. The output signal y(n) is obtained by con-

volving ht with x(n) and adding a white Gaussian noise sig-

nal with an SNR of 39 dB. In order to evaluate the condi-

ht − h(n)2 (1 − λ)L σw2

E 2

2

≈ 2 χE R1/2 (n) .

2

(62) tion number in diﬀerent situations, a white Gaussian signal is

ht 2 ht σ 2 added to the input x(n) with diﬀerent SNRs. The range of the

2 x 2

input SNR is −10 dB to 50 dB. Therefore, with an input SNR

Finally, we have a formula for the normalized misalign- equal to −10 dB (the white noise dominates the speech), we

ment in dB (which is valid only after convergence of the RLS can expect the condition number of the input signal covari-

algorithm): ance matrix to be close to 1, while with an input SNR of 50 dB

(the speech largely dominates the white noise), the condition

(1 − λ)L σ2 number will be high. Figures 1, 2, 3, 4, 5, 6, and 7 show the

m0 (n) ≈ 10 log10 + 10 log10 w2 evolution in time of the input signal, the normalized mis-

2 ht σ 2

2 x (63) alignment (we approximate the normalized misalignment

+ 10 log10 χE2 R (n) .

1/2 with its instantaneous value), and the condition number of

the input signal covariance matrix with diﬀerent input SNRs

Expression (63) depends on three terms or three factors: the (from −10 dB to 50 dB). We can see that as the input SNR in-

exponential window, the level of noise at the system output, creases, the condition number degrades as expected since the

and the condition number. The closer the exponential win- speech signal is ill-conditioned. As a result, the normalized

dow is to one, the better the misalignment is, but the tracking misalignment is greatly aﬀected by a large value of the con-

abilities of the RLS algorithm will suﬀer a lot. A high level of dition number. As expected, the value of the misalignment

noise as well as an input signal with a large condition num- after convergence in Figure 1 is equal to −49 dB and the con-

ber will obviously degrade the misalignment. With a fixed dition number is almost one. Now compare this to Figure 3.

exponential window and noise, it is interesting to see how In Figure 3, the misalignment is equal to −40 dB and the av-

the misalignment will degrade by increasing the condition erage condition number is 8.2. The higher condition num-

number of the input signal. For example, by increasing the ber in this case degrades the misalignment by 9 dB, which is

condition number from 1 to 10, the misalignment will de- exactly the degradation predicted by formula (63). We can

grade by 10 dB; the simulations confirm this. verify the same trend with the other simulations.

New Insights into the RLS Algorithm 337

×104 ×104

2 1

Input signal

Input signal

1 0.5

0 0

−1 −0.5

−2 −1

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(a) (a)

Misalignment (dB)

Misalignment (dB)

0 0

−10 −10

−20 −20

−30 −30

−40 −40

−50 −50

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(b) (b)

Condition number

Condition number

4 20

3 15

2 10

1 5

0 0

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(c) (c)

Figure 1: Evolution in time of the (a) input signal, (b) normalized Figure 3: The presentation is the same as in Figure 1. The input

misalignment, and (c) condition number of the input signal covari- SNR is 10 dB.

ance matrix. The input SNR is −10 dB.

×104 ×104

1 1

Input signal

Input signal

0.5 0.5

0 0

−0.5 −0.5

−1 −1

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(a) (a)

Misalignment (dB)

Misalignment (dB)

0 0

−10 −10

−20 −20

−30 −30

−40 −40

−50 −50

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(b) (b)

Condition number

Condition number

4 80

3 60

2 40

1 20

0 0

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(c) (c)

Figure 2: The presentation is the same as in Figure 1. The input Figure 4: The presentation is the same as in Figure 1. The input

SNR is 0 dB. SNR is 20 dB.

338 EURASIP Journal on Applied Signal Processing

×104 ×104

1 1

Input signal

Input signal

0.5 0.5

0 0

−0.5 −0.5

−1 −1

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s)

Time (s)

(a)

(a)

Misalignment (dB)

Misalignment (dB)

0 0

−10 −10

−20 −20

−30 −30

−40 −40

−50 −50

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(b) (b)

Condition number

Condition number

200 3000

150 2000

100

50 1000

0 0

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

Time (s) Time (s)

(c) (c)

Figure 5: The presentation is the same as in Figure 1. The input Figure 7: The presentation is the same as in Figure 1. The input

SNR is 30 dB. SNR is 50 dB.

×104 8. CONCLUSIONS

1

Input signal

0.5

The RLS algorithm plays a major role in adaptive signal pro-

0

cessing. A very good understanding of its diﬀerent variables

−0.5

may lead to new concepts and new algorithms. In this paper,

−1

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 we have shown that the update equation of the RLS can be

Time (s) written in terms of the a priori or a posteriori interpolation

error signals normalized with their respective interpolation

(a)

error energies. Hence, the interpolation error energy formu-

lation can be further exploited. This formulation has moti-

Misalignment (dB)

0

−10 vated us to propose a simple and an eﬃcient way to estimate

−20 the condition number of the input signal covariance matrix.

−30 We have shown that this condition number can be easily inte-

−40

−50 grated in the FRLS structure at a very low cost from an arith-

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 metic complexity point of view. Finally, we have shown how

Time (s) the misalignment of the RLS depends on the condition num-

(b) ber. A formula was derived, predicting how the misalignment

degrades when the condition number increases. The accu-

Condition number

400

200

REFERENCES

0

0.5 1 1.5 2 2.5 3 3.5 4 4.5 5

[1] M. G. Bellanger, Adaptive Digital Filters and Signal Analysis,

Time (s) Marcel Dekker, New York, NY, USA, 1987.

[2] B. Widrow and S. D. Stearns, Adaptive Signal Processing,

(c)

Prentice-Hall, Englewood Cliﬀs, NJ , USA, 1985.

Figure 6: The presentation is the same as in Figure 1. The input [3] S. Haykin, Adaptive Filter Theory, Prentice-Hall, Upper Saddle

SNR is 40 dB. River, NJ, USA, 4th edition, 2002.

New Insights into the RLS Algorithm 339

cations to Real-World Problems, Springer-Verlag, Berlin, 2003.

[5] A. H. Sayed and T. Kailath, “A state-space approach to adaptive

RLS filtering,” IEEE Signal Processing Magazine, vol. 11, no. 3,

pp. 18–60, 1994.

[6] S. Kay, “Some results in linear interpolation theory,” IEEE

Trans. Acoustics, Speech, and Signal Processing, vol. 31, no. 3,

pp. 746–749, 1983.

[7] B. Picinbono and J.-M. Kerilis, “Some properties of prediction

and interpolation errors,” IEEE Trans. Acoustics, Speech, and

Signal Processing, vol. 36, no. 4, pp. 525–531, 1988.

[8] G. H. Golub and C. F. Van Loan, Matrix Computations, The

Johns Hopkins University Press, Baltimore, MD, USA, 1996.

[9] J. Benesty, T. Gänsler, M. M. Sondhi, and S. L. Gay, Advances

in Network and Acoustic Echo Cancellation, Springer-Verlag,

Berlin, 2001.

ceived M.S. degree in microwaves from

Pierre & Marie Curie University, France,

in 1987, and his Ph.D. degree in control

and signal processing from Orsay Univer-

sity, France, in 1991. During his Ph.D. (from

November 1989 to April 1991), he worked

on adaptive filters and fast algorithms at the

Centre National d’Etudes des Telecomuni-

cations (CNET), Paris, France. From Jan-

uary 1994 to July 1995, he worked at Telecom Paris University.

From October 1995 to May 2003, he was with Bell Laboratories,

Murray Hill, NJ, USA. In May 2003, he joined INRS-EMT, Uni-

versity of Quebec, Montreal, Quebec, Canada, as an Associate Pro-

fessor. His research interests are in acoustic signal processing and

multimedia communications. He is the recipient of the IEEE Signal

Processing Society 2001 Best Paper Award. He coauthored the book

Advances in Network and Acoustic Echo Cancellation (Springer-

Verlag, Berlin, 2001) and coedited/coauthored three more books.

He received his M.S. degree in electrical

engineering and his Ph.D. degree in sig-

nal processing from Lund University, Lund,

Sweden, in 1990 and 1996. From 1997 to

September 1999, he held a position as an

Assistant Professor at Lund University. Dur-

ing 1998, he was employed by Bell Labs,

Lucent Technologies, as a Consultant and

from October 1999, he joined the techni-

cal staﬀ as a member. From 2001, he is with Agere Systems Inc.,

a spin-oﬀ from Lucent Technologies’ Microelectronics group. His

research interests include robust estimation, adaptive filtering,

mono/multichannel echo cancellation, and subband signal pro-

cessing. He coauthored the books Advances in Network and Acoustic

Echo Cancellation and Acoustic Signal Processing for Telecommuni-

cation.

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