Signal Processing 80 (2000) 2567}2578

Turbo estimation algorithms: general principles,

and applications to modal analysis夽
Letizia Lo Presti, Gabriella Olmo*, Davide Bosetto
Dipartimento di Elettronica, Politecnico di Torino, Corso Duca degli Abruzzi 24 } 10129 Torino, Italy
Received 22 July 1999; received in revised form 6 June 2000

Abstract
In this paper, a new class of parameter estimation algorithms, called turbo estimation algorithms (TEA), is introduced.
The basic idea is that each estimation algorithm (EA) must perform a sort of intrinsic denoising of the input data in order
to achieve reliable estimates. Optimum algorithms implement the best possible noise reduction, compatible with the
problem de"nition and the related lower bounds to the estimation error variance; however, their computational
complexity is often overwhelming, so that in real life one must often resort to suboptimal algorithms; in this case, some
amount of noise could be still eliminated. The TEA methods reduce the residual noise by means of a closed loop
con"guration, in which an external denoising system, fed by the master estimator output, generates an enhanced signal to
be input to the estimator for next iteration. The working principle of such schemes can be described in terms of a more
general turbo principle, well known in an information theory context. In this paper, an example of turbo algorithm for
modal analysis is described, which employs the Tufts and Kumaresan (TK) method as a master EA.  2000 Elsevier
Science B.V. All rights reserved.

Zusammenfassung
In diesem Artikel wird eine neue Klasse von Algorithmen zur ParameterschaK tzung vorgestellt, die sogenannten Turbo
Estimation Algorithms (TEA). Die Grundidee ist, dass jeder SchaK tzalgorithmus (EA) eine Art spezi"sche Rauschunter-
druK ckung der Eingangsdaten leisten muss, um zuverlaK ssige SchaK tzungen zu erzielen. Optimale Algorithmen implemen-
tieren die bestmoK gliche RauschunterdruK ckung, die kompatible zu der Problemde"nition und der entsprechenden unteren
Schranke der SchaK tzfehlervarianz ist. Der damit verbundene Rechenaufwand ist oft sehr gro{, so dass in praktischen
Situationen ein Ausweg in suboptimalen Algorithmen gesucht wird, wobei dadurch das Rauschen immer noch ein wenig
unterdruK ckt werden kann. Die TEA-Methode reduziert das restliche Rauschen durch eine Schleifenkon"guration, in der
ein externes RauschunterdruK ckungssystem, das durch einen HauptschaK tzer gefuK ttert wird, ein verbessertes Signal erzeugt,
welches wiederum das Eingangssignal der naK chsten Iteration ist. Die Arbeitsprinzipien solcher Schemata koK nnen
allgemein als Turboprinzipien bezeichnet werden, welche im Zusammenhang mit der Informationstheorie wohlbekannt
sind. In diesem Artikel wird als Beispiel ein Turboalgorithmus beschrieben, der die Tufts und Kumaresan Methode als
HauptschaK tzalgorithmus verwendet.  2000 Elsevier Science B.V. All rights reserved.

Re2 sume2
Dans cet article, nous introduisons une nouvelle classe d'algorithmes d'estimation de paramètres appeleH s algorithmes
d'estimation turbo (TEA). L'ideH e de base est que chaque algorithme d'estimation (AE) doit reH aliser une sorte de

夽
The authors are with the Signal Analysis and Simulation group at Politecnico di Torino.
* Corresponding author. Tel.: #39-11-5644033; fax: #39-11-5644149.
E-mail addresses: lopresti@polito.it (L.L. Presti), olmo@polito.it (G. Olmo).

0165-1684/00/$ - see front matter  2000 Elsevier Science B.V. All rights reserved.
PII: S 0 1 6 5 - 1 6 8 4 ( 0 0 ) 0 0 1 3 2 - 8
2568 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578

deH bruitage intrinsèque des donneH es d'entreH e pour atteindre des estimations "ables. Des algorithmes optimaux mettent en
wuvre la meilleure reH duction de bruit possible, compatible avec la deH "nition du problème et avec les limites infeH rieures
correspondantes de la variance de l'erreur d'estimation. Cependant, leur complexiteH de calcul est souvent prohibitive, de
sorte que dans la reH aliteH on doit souvent se contenter d'algorithme sous-optimaux. Dans ce cas, une certaine quantiteH de
bruit pourrait encore e( tre eH limineH e. Les meH thodes TEA reH duisent le bruit reH siduel au moyen d'une con"guration en boucle
fermeH e, dans laquelle un systeH me de deH bruitage externe, nourri par la sortie de l'estimateur mam( tre, geH nère un signal
rehausseH qui sera l'entreH e de l'estimateur à l'iteH ration suivante. Le principe de travail de tels scheH mas peut e( tre deH crit en
termes d'un principe plus geH neH ral, le principe turbo, bien connu dans le contexte de la theH orie de l'information. Dans cet
article, un exemple d'algorithme turbo pour une analyse modale est deH crit, qui utilise la meH thode de Tufts et Kumaresan
comme AE mam( tre.  2000 Elsevier Science B.V. All rights reserved.

Keywords: Parameter estimation; Modal analysis; Iterative algorithm

1. Introduction EAs are suboptimal, and some amount of theoret-

The performance of any estimation algorithm
(EA) is a!ected by the noise which impairs the
measured data samples. Given an input signal-to-
noise ratio (SNR) and the noise statistical proper-
ties, the performance of each algorithm depends on
its capability of extracting the useful information
from the available data, getting rid of the largest
possible amount of noise. In other words, each EA
performs a sort of intrinsic denoising of the data.
A very intuitive example of such a mechanism, in
the "eld of modal analysis, is o!ered by those
methods which exploit the properties of the princi-
pal eigenvectors of the autocorrelation matrix R
VV
[3]. In fact, under the assumption that the signal is
the superimposition of a known number p of modes
in white additive gaussian noise (AWGN), it can be
shown that the signal subspace is spanned by the
p principal eigenvectors of R , whereas the remain-
VV
ing M!p eigenvectors, where M is the linear di-
mension of R , span the noise subspace. This
VV
simple result makes it possible to achieve a separ-
ation between the signal and the noise subspaces,
i.e. to operate a sort of data denoising. Of course, as
is pointed out in [3], some amount of noise is
actually present also in the signal subspace; how-
ever, this noise cannot be separated from the signal.
The presence of a certain amount of unavoidable
noise leads to lower bounds to the estimation vari-
ance [15], which are reached by maximum likeli-
hood estimators (MLE). However, MLE is often
infeasible due to the associated computational
complexity. Therefore, most commonly employed
ically removable noise still a!ects the estimation.
Following these considerations, it is worth ques-
tioning if it is possible to improve the performance
of a given suboptimal EA, operating an external
reduction of the residual noise which is not sup-
pressed by the EA but does not overlap with the
signal. This task could be done by means of a closed
loop con"guration, in which an external denoising
system (EDS), fed by the master EA output, reduces
the residual noise generating an enhanced signal to
be input to the EA for next iteration. The obtained
class of algorithms are named turbo estimation algo-
rithms (TEA).
This paper is organized as follows. Section 2 in-
troduces the turbo estimation principle and de-
scribes the intrinsic and external denoising
operations. Section 3.1 reports the basics of the
modal analysis problem. In Section 3.2 a modi"ed
TK algorithm is described and interpreted in the
TEA context; speci"cations are given for the design
of the EDS. Sections 4 and 5 report the experi-
mental results and a complexity analysis of the
proposed algorithm. Finally, in Section 6 con-
clusions are drawn and future research directions
are identi"ed.
2. TEA description
The general scheme of a TEA is reported in
Fig. 1, where y is the data vector, containing N sam-
ples of the input process. This latter is the sum of
a random noise process realization and a signal
 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578
Fig. 1. Scheme of a turbo estimation algorithm.
component, which is a known function of an M-
dimensional, deterministic but unknown parameter
vector q"[q , q ,2, q ]. The vector parameter
  +
must be estimated by means of some proper EA,
yielding a vector estimate q( .
In the following, we will make the assumption
that in the noise-free case a procedure does exist
able to exactly obtain the vector parameter q from
a "nite number of available data; in other words,
we do not consider ill-posed problems. Moreover,
we assume that, in the presence of noise, an e$cient
estimator of q does exist, and consequently tight
bounds to the variance of each component of the
vector estimation error e"q( !q can be obtained
by inverting the Fisher information matrix [15].
These variances are zero only in the noise-free case,
when the e$cient estimator must yield q( "q.
As discussed, each EA implements some sort of
noise reduction, which can be active if the EA is
actually able to improve the input SNR (for
example, by means of "ltering operations), or pass-
ive if, due to the algorithm structure, not all the
noise components present in the input signal a!ect
the estimation process (as in the periodogram,
where the estimation is not a!ected by the noise
components at frequencies other than those to be
estimated). In the MLE, all the possible denoising is
performed, and the estimation variance reaches its
lower bound [15]. However, in many cases the
complexity of the optimal algorithm is overwhelm-
ing, and suboptimal solutions must be adopted. In
this case an external reduction of the noise not
suppressed by the EA, but a!ecting its performance
(which will be named residual noise) can be tried.
This basic principle is implemented in the block
diagram of Fig. 1, where, at each iteration k*1 the
EA is fed with s , which is an improved version of
I
the measured input signal y, processed by an EDS
2569
which in turn is controlled by the (k!1)th iter-
ation estimates produced by the EA.
The convergence check (CC) module controls the
iterative procedure termination on the basis of
a predetermined convergence criterion, which takes
into account the variations of the estimated para-
meter values during the iterations.
A TEA scheme can achieve the MLE if the resid-
ual noise is completely suppressed by the EDS after
an arbitrary number of iterations. If the residual
noise can be only reduced by the EDS, a subopti-
mal solution is obtained, whose advantage lies in
the lower estimation error variance with respect to
the master EA, or, alternatively, on the lower com-
putational complexity for the same estimation ac-
curacy. This is a typical characteristic of the
feedback systems, which, in many practical cases,
represent the only feasible solution to match the
quality requirements with given computational
complexity.
From the above discussion, it is clear that TEAs
be classi"ed as approximations of the MLE, in
which one starts from a known EA, and enhances its
performance through an iterative mechanism in
order to approach the MLE, instead of devising an
approximate MLE algorithm from scratch. Actual-
ly, well known approximate MLE algorithms, such
as the Iterative Filtering Algorithm (IFA) [3] can be
re-interpreted within the framework of iterative de-
noising.
If the input SNR is so low that the EA is forced to
operate below (or close to) threshold at the initia-
lization step, and consequently an outlier occurs in
the parameter initial estimates which feeds the
EDS, the well known steep depart of the estimation
error variances from the asymptotic values occurs
[13]. The overall outlier probability is related } but
not equal } to the outlier probability of the master
EA; in fact, due to the external denoising capability,
it is possible that an outlier in the initial estimate is
corrected through subsequent iterations. This error
correction capability of turbo algorithms has been
observed experimentally during the simulations of
TEA systems, and represents a further advantage of
the scheme.
Several improvements can be devised to the basic
TEA scheme; di!erent EAs, or the same EA with
di!erent parameters, can be employed at di!erent
2570 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578
iterations, in order to achieve a trade o! between
the EA robustness (crucial at the "rst iteration,
when all the noise is present) and the estimation
accuracy [10]. Moreover, the EDS can be designed
taking into account some sort of soft information
yielded by the EA, and representing the reliability
of the estimate.
Iterative estimation schemes share some features
with the well known turbo principle [2], introduced
in an information theory context to generalize the
well known turbo codes [1]. In fact, a suboptimal
EA and an EDS block, put in a closed loop con"g-
uration, with a possible exchange of reliability in-
formation, can provide a feasible way to approach
the theoretical limits of the estimation accuracy.
Thanks to this analogy, iterative EA/EDS systems
have been named turbo estimation algorithms
(TEAs).
3. A turbo algorithm for modal analysis
In this section, the basic TEA concepts are de-
scribed by means of an example in the context of
modal analysis. Further developments can be
found in [8]. It is important to remark that the
scope of TEAs is not limited to modal analysis; for
example, in [9,10] TEA schemes are devised for
carrier frequency estimation; so, the main concepts
described in this paper can be easily generalized to
many other estimation problems and master EAs.
3.1. The frequency estimation problem
The problem of estimating closely spaced fre-
quencies of multiple, superimposed sinusoids from
noisy measurements, is a representative of a class of
nonlinear parameter estimation problems with
a vast range of signal processing applications. The
signal y[n] is assumed to be the sum of p complex
undamped exponentials
N
y[n]" A e
pDI L>(I #w[n] (1)
I
I
A being the constant (complex) amplitudes, f the
I I
numeric frequencies,
the initial phases and w[n]
I y[N!1] y[N!2] 2 2 y[N!¸]
an additive noise term. The problem of interest is to
determine the frequencies f given observations of
I
y[n] over a "nite time interval n"0,2, N!1.
The MLE approach to the estimation of the
frequencies and complex amplitudes in Eq. (1)
would be to set up a nonlinear least squares minim-
ization problem, which is computationally burden-
some, and necessary only at very low SNR values.
In the recent years a great deal of attention has
been devoted to model-based eigendecomposition
methods. The use of linear models for the signal to
be processed allows one to convert a nonlinear
estimation problem into the simpler task of estima-
ting the parameters of the linear model [4]. The
nonlinearity is postponed to the second step, which,
in all model-based methods, is devoted to the ex-
traction of the desired information (frequencies in
our case) from the estimated model parameters.
Often in signal processing, a polynomial realization
of the linear model is assumed and used. The poly-
nomial realization, "rst suggested by Prony [12],
exploits the fact that in the absence of noise, the
complex exponential signal is exactly predictable as
a linear combination of its p past samples
N
y[n]" a y[n!k]. (2)
I
I
So, the Prony linear prediction (LP) formulation
converts the nonlinear problem of frequency es-
timation to the linear parameter estimation prob-
lem for this polynomial model. Once the model
parameters are estimated, the frequencies f are
I
obtained from the roots of the polynomial
N
H(z)"1! a z\I.
I
I
As Eq. (2) holds for n"p#1,2, N, when the
order p is known a priori, a linear system of equa-
tions is obtained which, if N'2p, can be solved for
the linear prediction coe$cients a , a ,2, a . The
  N
problem reduces to the solution of the set of linear
equations
y[¸] y[¸!1] 2 2 y[1]
y[¸#1] y[¸] y[2]
$ $ $
$ $ $
 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578
a 3.2. The TKIF method and its interpretation as a TEA

a y[¸#1]
 Let us consider the block diagram of Fig. 2. The
a y[¸#2]
 block labeled H (z), where k*0 is the index of the
; $" $ (3)
$ $
$ y[N]
a fK
* I\
or Y ) a"y. When the data is noisy, based on prior
knowledge of the number of harmonics, ¸ is chosen
to be equal to p, and the LP parameters can be
obtained from the least squares solution of the
overdetermined system of equations, [12].
In the last years, there has been a burst of activity
in singular value decomposition (SVD) based
methods for sinusoid retrieval, and they all use the
close-to-p-rank property of Y in the presence of
additive noise at moderate SNR. One of the "rst
methods of this class was introduced by Tufts and
Kumaresan [5], who obtained signi"cant improve-
ments with respect to least square solution in the
presence of noise [6]. Based on the fact that the
noiseless matrix Y has rank p, they suggest that in
the noisy case the matrix Y be replaced by a p-rank
approximant YK before the pseudo inverse opera-
N
tion. To this end, one compute the SVD of matrix
Y, and sets all but the p largest singular values to
zero. With r"min(¸, N!¸)*p
P
Y"URV &" p u *&,
I I I
I
where R is a diagonal matrix containing the ¸ sin-
gular values p of Y, the symbol & denotes trans-
I
pose, U and V are two orthogonal matrices [7]. The
p rank approximation is [7]
N
YK " p u *&.
N I I I
I
This low-rank approximation of Y before the
pseudo-inverse operation considerably improves
the quality of the parameters estimates, because the
(¸!p) singular vectors corresponding to the clus-
ter of small singular values will no be longer con-
tribute to the solution, thus removing a major
source of sensitivity to noise.
2571
I
iterative process, is a digital IIR multiband "lter
performing a noise suppression which, for each
iteration k*1, bene"ts from the knowledge of the
frequency vector estimated at the previous step,
. From now on, the frequency estimates will be
labeled with the iteration index k, as they are pro-
gressively re"ned.
At the initialization step k"0, the "ltering is
omitted, so H (z)"1, s "y, and a rough pre-
 
estimate of f, fK , is obtained using the TK algorithm

with relatively small values of N and ¸ (numerical
values of these parameters are given in Section 4).
Then, at each step of subsequent iteration:
(1) the IIR multiband "lter is designed, exploiting
the knowledge of the previous cycle frequency
estimates fK ;
I\
(2) y is passed through the "lter H (z), giving rise
I
to s ;
I
(3) s is processed by the TK algorithm and a new
I
frequency estimate fK is obtained.
I
The procedure stops at the Kth iteration, with
K determined according to the convergence cri-
terion, and the last iteration yields the "nal esti-
mated frequency vector fK "fK.
)
The described algorithm, being an iterative
method for frequency estimation, and involving
frequency-selective "ltering, can be interpreted as
a TEA. The TK plays the role of the basic EA,
Fig. 2. Scheme of the TKIF algorithm.
2572 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578

whereas the "lter H (z) can be identi"ed with
I
the EDS module. The algorithm has been named
TK algorithm with iterative xltering (TKIF). The
key feature of TKIF lies in its capability of pro-
gressively rexning the estimates of the linear predic-
tion vector a( , thanks to the noise reduction
performed on the measured data y by the "lter-
ing operation. In fact, let us compare the initial
estimate, [7]
N 1
a( " * u&y,
SIG , wI"(o!d)e
SIG .
p I I
I I
and the estimate at iteration k
N 1
a( " * u&s ,
I p I I I
I I
where the label k has been attached to the matrices
output by the SVD at step k. It can be noticed that
at the kth iteration, the SVD operates in enhanced
SNR conditions; consequently, it is able to yield
enhanced frequency estimates, which in turn can be
used to drive a more e!ective (k#1)th iteration
"lter design.
In order to completely de"ne TKIF, it is neces-
sary do design the EDS "lter. A possible choice is to
employ a multiband "lter H (z), with each band
I
centered around a component f K [m], 1)m)p of
I
the estimated frequency vector fK . In this way, at
I\
each iteration k*1, it suppresses most of the out-
of-band noise power, exploiting the information of
the estimated frequency vector at the previous iter-
ation, fK . An IIR "lter structure has been chosen,
I\
because it can be easily designed from the estimated
vector fK , it involves a low computational complex-
I
ity both in the "ltering and in the design phase, and
can o!er a good frequency selectivity.
pDG , where f are
In the following we propose two actual imple-
mentations of the EDS module. The "rst one
(which leads to the algorithm labelled TKIF-1) is
an all pole "lter whose transfer function in the
z domain at the kth iteration is
z+
HI (z)" , (5)
#"1 “+ (z!pI)
G G out of the N"128 available ones. Obviously, as
where
pI"oe
SIG , uI"2pf [i ]. (6)
G G I
The second choice (leading to the algorithm label-
led TKIF-2) is a modi"cation of the previous one,
motivated by the obtained simulation results as will
be discussed in the following: the "lter transfer
function in the z domain at the kth iteration is
“+ (z!wI)
HI (z)" K K , (7)
#"1 “+ (z!pI)
G G
where
(4) pI"oe
G K
The new parameter d, which assumes values in the
range [0,o], has been introduced for reasons which
will be clari"ed in the following.
3.3. EDS parameter dexnition
This section is devoted to the evaluation of the
EDS "lter parameter e!ects on the algorithm per-
formance. A test signal made of two undamped
modes with frequencies f"[0.20, 0.22], phases
u"[!3.63, 0], amplitudes A"[1, 1] has been
used thorough all the related simulations. A num-
ber N"128 of available data points has been con-
sidered. The original noise contribution is an
AWGN, and all the simulations are carried out
with SNR"0 dB. When the signal consists of
two equal-level complex tones, the CrameH r}Rao
Bounds (CRBs) for the two frequencies are equal as
the mutual interference is reciprocal [13]. Using the
above numerical values and inverting the Fisher
information matrix we obtain that CRB"70 dB
for each frequency.
First of all, the TEA scheme has been tested in an
open loop con"guration: the EDS has been kept
"xed with pole locations in p "oe
G G
the true frequencies. As in this phase we are not
interested in evaluating the e!ects of the "lter initial
conditions on the estimation process, but only to
investigate the parameter selection, after the "lter-
ing operation we skip those data samples which are
a!ected by the "lter transient and employ, for the
actual estimation process, only N "64 samples
G
the simulations are worked out exploiting only the
information carried by the stationary signal seg-
ment, the obtained performance turn out to be
 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578
rather far from the corresponding CRB. On the
other hand, simulation results, worked out with
optimized values of N and reported in Section 4,
G
show that the actual performance of the proposed
algorithm is very close to the CRB.
In summary, the following two-step frequency
estimation procedure is addressed:
(1) pass the input data sequence y through the EDS
"lter to obtain s;
(2) enter the second half of s (N "64 data sam-
G If we impose an insertion gain of about 10 dB per
ples) into the TK algorithm with ¸"48 to
obtain an estimate fK.
One hundred independent realization of the data
vector y with SNR"0 dB have been generated
and the same data sequences have been used for all
the addressed experiments. With this Monte Carlo
method, the following quantities have been esti-
mated and employed as overall performance index.
The average mean-square error (MSE), denoted by
k"+E( f K !f )#E( f K !f ),, (8)
     In order to verify if a better "lter selectivity is
which depends on both the squared bias and the
estimation variance; the average squared-bias
b"+[E( fK )!f ]#[E( fK )!f ], (9)
     the previous evaluation it can be noticed that o can
and the average variance
p"+var[ fK ]#var[ fK ],. (10)
   a value oK0.93 can be worked out, which yields an
In the "rst experiment, we employ the EDS "lter of
Eq. (5). The choice of the pole module o is crucial as
for the TKIF method performance, as it a!ects the
"lter e!ectiveness in rejecting the residual noise.
A natural choice would be to select values of o very
close to unity; however this choice presents two
serious drawbacks. First of all, a high frequency
selectivity involves that the "lter impulse response
time decay is slow; this induces a long transient in
the "ltered signal s , which leads to the need for
I
extra measured data in order to accommodate both
the transient and the steady-state data to be
processed by the EA. Moreover, it has been ex-
perimentally veri"ed that the "lter selectivity is
related to the overall outlier probability; this can be
 It must be remarked that this number of extra data has little
e!ect on the computational complexity, as they are only used in
the "ltering process (see Section 5).
2573
intuitively explained noticing that, when the pre-
estimated frequencies are imprecise (due either to
an outlier occurrence in the initialization EA or to
simple estimation variance), a very selective "lter-
ing around the pre-estimated frequencies could
completely cut o! some of the true signal compo-
nents, forcing an outlier in the overall estimation
process. On the other hand, if less selective "lters
are employed, an initial outlier can be recovered by
means of the noise suppression mechanism.
sinusoid, the corresponding pole module of Eq. (6)
turns out to be o"0.82; with this value, it can be
veri"ed that the "lter transient does not exceed the
"rst 64 samples of s; in fact, o"0.82 corresponds to
an exponential decay time constant qK5, and
therefore 64 is about 13q. Therefore, the "lter tran-
sient does not a!ect the estimation procedure, as
this latter is based on the second half of the
N"128 data record. The related simulation results
are presented in the "rst row of Table 1.
able to improve the estimation accuracy, we have
carried out a second experiment, modifying the
pole location so as to achieve a larger gain. From
be increased to some extent without exceeding a "l-
ter transient of 64 samples. Imposing 64+5q,
insertion gain of about 14 dB per sinusoid.
The corresponding experimental values of k,
p and b are shown in the second row of Table 1.
As it can be noticed, the performance improvement
obtained using a larger gain EDS "lter is negligible,
as a small variance decrease is paid in terms of
a larger bias. The obtained results reveal that the
performance of the EDS "lter is a!ected by some
factor other than the gain. It can be noticed that the
"lter alters the AWGN condition, which is one of
the assumptions of the TK algorithm in order to
assure proper operation. Therefore, a possible hy-
pothesis can be that the obtained results are nega-
tively a!ected by the presence of colored noise.
In order to test this assumption, the design of the
EDS has been modi"ed so that the "lter e!ectively
enhances the sinusoids but, at the same time, keeps
the output noise spectrum su$ciently smooth (or
`locally whitea). To this end, the EDS "lter of
2574 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578
Table 1
Simulation results for the design of the EDS module. N"128; N "64, SNR"0 dB. The CRB is 70 dB
Method MSE, k
TKIF-1 (o"0.82) 4.66e!7
TKIF-1 (o"0.93) 4.21e!7
TKIF-2 (o"0.93, d"0.18) 3.30e!7
Eq. (7) is employed in the third experiment. The
parameter d determines both the selectivity and the
gain of the pre"lter. Taking o"0.93 and d"0.18,
the results reported in the last row of Table 1 can be
obtained. It can be noticed that this second EDS
"lter achieves improved performance with respect
to the "lter of Eq. (5), especially in terms of esti-
mation bias; the bias reduction represents a
remarkable advantage of HI (z) over HI (z),
#"1 #"1
also con"rmed by the simulation results reported in
Section 4. Moreover, from Table 1, we see that the
second EDS "lter achieves improved performance
also in terms of estimation variance, with an in-
crease of 1 dB switching from HI (z) to HI (z).
#"1 #"1
We remark once again that these results are not
signi"cant as for the actual performance of the
algorithm, but are only focussed on the "lter de-
sign, and therefore must be considered as terms of
comparison of the addressed experiments.
4. The TKIF algorithm: experimental results
From the results of Table 1, it could seem that
the cascade EDS}EA does not allow the algorithm
to reach the CRB with any of the two considered
EDS "lters. Actually, it is indeed possible to ap-
proach the CRB provided that the whole transient,
or a portion of it, is included in the data sequence
passed on to the EA block. So far, the in#uence of
the data "lter initial conditions has been neglected;
the "ltered noise has been considered stationary
and the amplitude of the "ltered sinusoid constant
in time. These assumptions are not realistic, espe-
cially when the "lter transition period becomes
comparable to N. The in#uence of the biasedness of
the frequency estimation algorithm in an iterative
"ltering procedure has already been pointed out in
[3,11], as well as in this paper.
G
MSE in dB
variance, p bias, b !10 log (p)
4.64e!7 1.56e!9 63.3
4.7e!7 3.55e!9 63.8
3.30e!7 2.96e!10 64.8
Simulations have been carried out, assuming that
either N"64 or 128 data samples are available. It
has been experimentally veri"ed that, as the band-
width parameter o increases towards unity, the
MSE of both methods does not decrease monotoni-
cally, but reaches a minimum and then gets in-
creased. A closer examination of the simulation
results reveals that this is due to an increase of the
squared bias b, while the estimation variance
monotonically decreases as o grows towards unity.
Actually, when o is low, never dominates the vari-
ance, and hence k shares the same decreasing
trend of the variance. On the other hand, the
squared bias becomes dominant as o tends to unity,
and a tradeo! e!ect between b and p takes place.
Consequently, using a value of o very close to unity
is no longer a good strategy to minimize the MSE.
Due to the same e!ect, the di!erence between
TKIF-1 and TKIF-2, whose EDS "lter introduces
lower bias, becomes apparent. Note that the es-
timation bias can be reduced also by discarding
¹ initial samples from the sequence after the "rst
M
estimate.
In order to show the performance of TKIF-1 and
TKIF-2 in di!erent SNR conditions, Fig. 3 reports
the estimation variance p as a function of SNR, for
the two situations of N"128 and 64 available
samples. The thin lines represent the CRB for
N"128 and 64. The results are obtained for ex-
perimentally optimized values of the parameters o,
d and ¹ . The "nal estimates are reached in K"3
M
iterations, and the initial estimate is obtained with
TK and ¸"10. The TK algorithm with ¸"10 is
also employed as the EA for the closed loop iter-
ations. It can be noticed that both the curves of
TKIF-1, especially the one with N"64, exceed
the CRB. This is due to the bias present in both
estimates. On the contrary, the variance of TKIF-2
closely follows, but never exceeds, the CRB.
 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578 2575

Moreover, the TKIF-2 threshold is lower, thanks
to the better selectivity of the EDS "lter.
The numerical performance index for the same
SNR values are presented in Table 2 for N"64
and in Table 3 for N"128. These results show that
both TKIF-1 and TKIF-2 achieve a MSE very
close to the CRB. Moreover, a closer examination
of Tables 2 and 3 reveals that the bias of TKIF-1 is
not negligible, but instead it plays a role compara-
ble to the standard deviation in the mean square
error de"nition.
Finally, it is worth noticing that the performance
of both algorithms degrades dramatically when the
SNR falls below 2.5 dB. The poor initial accuracy of
TK with ¸"10 is largely responsible for this thre-
shold. In fact, simulations have shown that the
threshold can be drastically reduced if the two
frequencies corresponding to the largest absolute
values of the data discrete Fourier transform are
taken as the initial estimates.

5. The TKIF method: analysis of complexity

In this section, we address the topic of analyzing

the computational complexity of TKIF. It is worth
Fig. 3. Plot of !10 log (var) against SNR in dB for TKIF-1 and recalling that the objective of TEA is either to
TKIF-2; the thin lines indicate the CRBs for N"64 and reduce the basic EA complexity, maintaining the
N"128. same performance, or to yield higher estimation

Table 2
TKIF-1 and TKIF-2 estimation indices in the case N"64

TKIF-1 TKIF-2

SNR k b p k b p

4 1.41e!2 1.40e!2 9.85e!5 8.55e!7 2.15e!8 8.33e!7
5 1.01e!6 1.22e!7 8.85e!7 4.62e!7 1.52e!8 4.47e!7
6 2.54e!7 9.10e!8 1.63e!7 3.24e!7 6.55e!9 3.17e!7
10 1.46e!7 8.43e!8 6.12e!8 1.21e!7 2.98e!9 1.18e!7
20 8.23e!8 7.78e!8 4.53e!9 1.17e!8 8.65e!10 1.08e!8
30 7.65e!8 7.61e!8 4.33e!10 1.16e!9 1.54e!10 1.01e!9

Table 3
TKIF-1 and TKIF-2 estimation indices in the case N"128

TKIF-1 TKIF-2

SNR k b p k b p

3 4.43e!6 1.14e!7 4.32e!6 1.04e!7 1.80e!9 1.02e!7
5 4.38e!8 7.68e!9 3.61e!8 3.85e!8 1.54e!9 3.70e!8
10 2.57e!8 1.90e!8 6.68e!9 9.96e!9 1.67e!10 9.79e!9
20 2.12e!8 2.06e!8 5.92e!10 1.30e!9 3.63e!10 9.38e!10
2576 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578
accuracy with the same complexity of a single EA
processing. As a metric for computational complex-
ity, we evaluate the number F of #oating point
operations ( yops) involved by TK and by his TEA
versions.
The key property of TKIF is determined by the
TK procedure and by the number of iterations, but
is almost independent of the "ltering operation. Let
us de"ne the complexity of the TK method as
F2)(N, ¸)KF14"(N, ¸)#F(¸), (11)
where F14"(N, ¸) is the SVD complexity, and
F* is the complexity of the procedure of roots
extraction of an ¸th order polynomial. It is known
that [14]
20
F(¸)K ¸, (12)
3 length N"64 and an optimal choice ¸"20, we
whereas the SVD complexity F14" can be approxi-
mated, according to the e$cient SVD computation
described in [14], as
F14"(N, ¸)K14RC#8¸, (13)
where R and C are, respectively, the numbers of
rows and columns of the LP data matrix Y; in our
case, R"N!¸ and C"¸, as it can be inferred
from Eq. (3). Eqs. (12) and (13) put into evidence
that the Prony "lter order ¸ is very critical as for
the algorithm complexity, and must be chosen so as
to minimize the computational cost to the max-
imum possible extent.
The complexity of the TKIF algorithm can be
then de"ned as
F2)'$(N, ¹ , ¸, K)
M decrease very quickly after the "rst iteration,
dent of k) and N is the number of signal samples.
The "ltering complexity is then determined by:
E the number N of input samples to be processed;
E the number M (proportional to N and N ) of
N U
modes which compose the signal;
E the number of transmission zeros.
If the complexity of the TK and TKIF algorithms
are directly compared, it could seem that this latter
is far more complex than the former. Actually, this
consideration holds true when the two algorithms
are compared with the same values of N and ¸,
which, in practice, is not a reasonable assumption.
In fact:
E The complexity of the IIR "ltering operation is
very low, if compared with that of the SVD or
root "nding procedure. Indeed, for a sample
have FK5.3;10 and F14"K3.1;10,
while for the "ltering operation, we need only 64
#oating point operation per pole/zero.
E The TKIF method requires smaller values of
N and ¸ to achieve the same performance of TK.
This fact, which is a consequence of the TEA
concepts, has been experimentally veri"ed, as
will be described in the following. The complex-
ity of the TK is so strongly dependent on N and
¸ so that it can be more economical to process
some iterations with lower N and ¸ than to
perform only one TK estimation with higher
values of N and ¸. In fact, the square and cubic
terms ¸ and ¸ in Eq. (13) of F14" can be
predominant against the linear term KF2) in Eq.
(14).
E The individual estimation error variance
KF2)(N, ¸)#K[F2)(N!¹ , ¸)#F$ ], (14) so that it is often possible to achieve a high
M estimation accuracy with a limited number of
where K is the number of iterations after the pre- iterations.
estimation, ¹ is the number of transient samples
M All these considerations are con"rmed by experi-
to be skipped after the "rst estimation and F$  is mental results. A comparison between the TKIF
the complexity of the IIR "ltering performed by the and the TK algorithms can be made in terms of the
EDS module. This latter is de"ned as complexity ratio
F$ "N(N #N ), c"F2)/F2)'$ (15)
U N
where N and N are, respectively, the numbers of In Section 4, the parameters o, ¹ and d of TKIF-2
U N M
zeros w and poles p of H (z) (supposed indepen- have been optimized for SNR"10 dB and the
G K I
 L.L. Presti et al. / Signal Processing 80 (2000) 2567}2578
two sample lengths N"64 and 128. In these
conditions, the TEA method with TK and ¸"10
as the basic EA attains the CRB with negligible
bias. In order to achieve the same performance with
the TK algorithm alone, ¸"40 when N"64 and
¸"30 when N"128 are necessary. Replacing the
numerical values into Eqs. (11) and (14), we obtain
for the case N"64 (¹ "20):
M
F2)(64, 40)"1.5;10,
F2)'$\(64, 20, 10, 3)"2.8;10,
and consequently c"5.39. Likewise, for N"128
we obtain (note that ¹ "40)
M
F2)(128, 30)"1.6;10,
F2)'$\(128, 40, 10, 3)"5.5;10,
and c"2.91.
The results reveal that the TKIF algorithm is
very e$cient with respect to the TK alone. More
precisely, in the considered numerical examples, the
total number of #ops gets reduced of a factor be-
tween 2.91 and 5.39, depending on the sample data
length N.
6. Conclusions
In this paper we have introduced the class of
turbo estimation algorithms (TEA), which represent
a feasible approximation of optimum parameter
estimation, based on a closed loop con"guration
able to achieve a very e$cient noise suppression.
The denoising mechanism of TEA has been de-
scribed, and an interpretation in terms of a more
general turbo principle has been given. As an
example, the TKIF algorithm has been de"ned and
analyzed in the context of modal analysis; it em-
ploys TK as a master EA and use an iterative
"ltering procedure to perform the external denois-
ing. The algorithm has been tested through simula-
tions and the obtained results show that it is able
either to reduce the complexity of TK, achieving
the same performance, or to yield higher accuracy
estimates with the same complexity of a single TK
processing.
2577
Further research could be spent to re"ne the
theoretical foundations of TEAs, for example by
evaluating the asymptotic performance of a generic
TEA versus its master EA. The TEA capability of
correcting an outlier occurred in the initial estima-
tion step, which has only been experimentally veri-
"ed in the present work, could be veri"ed and
formalized in detail. The conditions to be satis"ed
by an EA in order to be admissible as the master
EA within a turbo scheme should be investigated.
Finally, other turbo algorithms could be devised, in
order to solve various estimation problems other
than the modal analysis one, possibly with exten-
sions to the case of random parameter estimation.
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