ABSTRACTS

A.1
EFFICIENT IMPLEMENTATION ON MULTIPROCESSORS :
THE PROBLEM OF SIGNAL PROCESSING APPLICATIONS MODELLING

Laurent Kwiatkowski, Fernand Bori, Jean-Paul Stromboni

Laboratoire d'Informatique Signaux Systmes, UNSA - URA 1376 CNRS
41, Boulevard NAPOLEON III - F06041 NICE cedex - FRANCE
email : kwiatkow@alto.unice.fr

In signal processing area, applications involve a large amount of computation,
suggesting the use of multiprocessors to speed up processing. However, obtaining
good performance is not easy because the machine should take advantage of the
potential parallelisms of the studied application. That is why several parallel
implementation methods using mapping and scheduling algorithms has been
developped. One of development shells aims is application partitioning so that every
part will be processed by a different processor, like SynDEx [1] or Ptolemy[2].
These shells use some graph models to exhibit both potential parallelisms of the
application and the available multiprocessor parallelisms [3], but the task granularity
problem is not considered when the application is modelized.
The purpose of this paper is to emphasize the problem of the task granularity
when the application is modelized by means of a graph and to study the impact on
speedup. As a solution for this problem, this paper presents an original
implementation method based on the variation of the granularity and regular
application size.
Paper

A.2
PROCESSOR ARCHITECTURE FOR EXTENDED LAPPED TRANSFORM

David Akopian and Jaakko Astola

Signal Processing Laboratory,
Tampere University of Technology,
P.O.Box 553, FIN-33101, Tampere, Finland,
e-mails: prog@cs.tut.fi, jta@cs.tut.fi

ABSTRACT
This paper is devoted to implementation of Extended Lapped Transform (ELT),
which is among the most efficient factorization methods for  paraunitary
filterbanks. First we utilize thespecial form of the matrices in the
part of factorization of ELT to process data at input data rate in a
pipelined structure with minimal number of processor elements without
inserting additional delays. Next we suggest an algorithm for DCT-IV
transform, the other part of ELT factorization, with a constant geometry
structure suitable for the use of perfect-shuffle network.
Paper

A.3
Real-Time Obstacle Detection using Stereo Vision

Massimo Bertozzi, Alberto Broggi, Alessandra Fascioli

Dipartimento di Ingegneria dell'Informazione
Universita' di Parma, I-43100 Parma, Italy
Tel. +39-521-905707 Fax. +39-521-905723
e-mail: {bertozzi,broggi,fascal}@CE.UniPR.IT

This work presents a low-cost stereo vision system aimed to the real-time detection
of generic obstacles (without constraints on symmetry or shape) on the path
Thanks to a geometrical transform the perspective effect is removed from both
left and right stereo images. The difference between the results is used for
the detection of free-space in front of the vehicle.
The output of the processing is displayed on both an on-board monitor and a
control-panel to give a visual feedback to the driver. The system was tested
on MOB-LAB experimental land vehicle, which was driven for more than 3000 km
along extra-urban roads and freeways at speeds up to 80 km/h, and demonstrated
its robustness with respect to shadows and changing illumination conditions,
different road textures, and vehicle movement.
Paper

A.4
IMPLEMENTATION OF A FAST MPEG-2 COMPLIANT HUFFMAN DECODER

Mikael Karlsson Rudberg (mikaelr@isy.liu.se) and Lars Wanhammar (larsw@isy.liu.se)
Tel: +46 13 284059; fax: +46 13 139282

ABSTRACT
In this paper a 100 Mbit/s Huffman decoder implementation is presented.
A novel approach where a parallel decoding of data mixed with a serial
input has been used. The critical path has been reduced and a significant
increase in throughput is achieved. The decoder is aimed at the MPEG-2 Video
decoding standard and has therefore been designed to meet the required performance.
Paper

A.5
IMPLEMENTATION OF KOGBETLIANTZ'S SVD ALGORITHM USING
ORTHONORMAL MICRO--ROTATIONS

Jurgen Gotze (+), Peter Rieder (++) and Josef A. Nossek (++)

(+)  ECE ,Rice University, Houston, TX 77251--1892, U.S.A.
jugo@ece.rice.edu
(++) TU Munich, Arcisstr. 21, 80290 Munich, Germany
peri@nws.e-technik.tu-muenchen.de

In this paper the implementation of Kogbetliantz's SVD algorithm
using orthonormal micro--rotations is presented. An orthonormal
micro--rotation is a rotation by an angle of a given set of
micro--rotation angles which are choosen such that the rotation can be
implemented by a small amount of shift--add operations. All
computations (evaluation and application of the rotations) can
entirely be referred to orthonormal micro--rotations. Simulations show
the reduced computational complexity of Kogbetliantz's SVD algorithm
based on orthonormal micro--rotations comparded to the standard
Kogbetliantz SVD algorithm.
Paper

A.6
A VLSI ARCHITECTURE FOR REAL TIME OBJECT DETECTION ON HIGH RESOLUTION IMAGES

M. Wosnitza
M. Thaler
G. Troester

Electronic Laboratory
Swiss Institute of Technology Zuerich (ETHZ)
Gloriastrasse 35
CH-8092 Zuerich

ABSTRACT
This paper describes a VLSI-based SIMD multiprocessor system for the
implementation of a set of basic object detection algorithms.
The system architecture takes advantage of modern fast EDRAM-technology to
support the communication requirements of 800 Mbytes/s between main memory and
processors imposed by high resolution images.
A specialized processing element (PE) architecture for implementation in VLSI
which efficiently implements the basic set of algorithms is presented.
The performance of a single PE is discussed with respect to the different
algorithms.
A system consisting of 4 processing elements realized in 0.6mu
CMOS-technology is able to localize a 128x128 pixel
template in a 1024x1024 pixel image at a rate of 10 frames/second
(sustained performance  2.1 MOps/s).
Paper

A.7
Title:
A SINGLE CHIP MOTION ESTIMATOR DEDICATED TO MPEG2 MP@HL

Authors:
Takao ONOYE, Gen FUJITA, Masamichi TAKATSU, Isao SHIRAKAWA,
and Kenji MATSUMURA*

Affiliations:
Dept. Inf. Sys. Eng., Osaka University
{onoe, fujita, taka2, sirakawa}@ise.eng.osaka-u.ac.jp

*K.C.S. Co., Ltd.
Naka-Kosaka, Higashi-Osaka, Osaka, 577 Japan
matsu@k-c-s.k-c-s.co.jp

Abstract:
A single chip motion estimator dedicated to MPEG2 MP@HL is developed.
Adopting a two-level hierarchical searching algorithm in detecting
motion vectors, the computational labor can be reduced by 1/70.  A
novel mechanism is introduced into the full-search procedure, which
attempts the maximum possible reuse of reference pixels in order to
reduce the bandwidth of the frame memory interface.  The proposed
motion estimator is integrated in a 0.6um triple-metal CMOS chip with
the input clock rate up to 133MHz, which enables the real time motion
estimation.
Paper

A.8
VLSI DESIGN OF A PARALLEL ARCHITECTURE
2-D RANK ORDER FILTER

R. Roncella, R. Saletti, G. Savoia

Dipartimento di Ingegneria dell'Informazione:
Elettronica, Informatica, Telecomunicazioni,
Universit di Pisa,
Via Diotisalvi 2, 56126 Pisa (Italy)
tel: +39-50-568511; fax: +39-50-568522
E-mail: roncella@iet.unipi.it

A VLSI parallel architecture implementing a new algorithm for 2-D rank
order filtering, based on repeated maximum finding operations, is presented
in this paper, and the design of a programmable demonstrator chip realised
in standard-cell 1 um CMOS technology is described. The chip has programmable
window size and selectable rank, it can work with unitary throughput at
25 MHz, in the worst case, and its area is 7 x 5.5 sq.mm.
Paper

A.9
A NOVEL VLSI ARCHITECTURE FOR BLOCK
MATCHING ALGORITHMS*

Chen-Yi Lee

Dept. of Electronics Engineering, National Chiao Tung University 1001,
University Road, Hsinchu 300, Taiwan, ROC
Tel: 886-35-731849;
Email: cylee@cc.llctll.edu.tw

This paper presents a new VLSI architecture for full search
block matching motion estimation (ME) algorithm. The
proposed VLSI architecture has three specific features: (1) it
has a processor element (PE) array which provides sufficient
computational power and achieves 100% hardware
efficiency, where PE's work in a systolic style, (2) it
contains stream memory banks which provide scheduled
data flow needed in PE for computing mean absolute error
(MAE); and (3) it has minimal memory access bandwidth to
save I/O pin-count. As a result, the proposed architecture
allows to reach cost-effective ME hardware solution.
Paper

A.10
SYNTHESIS OF MEMORY-BASED VLSI ARCHITECTURES FOR DISCRETE WAVELET TRANSFORMS

Seonil Choi, Jongwoo Bae and Viktor K. Prasanna

Integrated Media Systems Center
Department of Electrical Engineering-Systems
University of Southern California
Los Angeles, CA 90089-2562
WWW:http://www.usc.edu/dept/ceng/prasanna/home.html
{seonil, jongwoo, prasanna}@halcyon.usc.edu

ABSTRACT
We propose novel VLSI architectures for computing the Discrete Wavelet
Transforms.  The proposed architectures employ a memory-based approach.
ROM look-up tables are used for the implementation of complex
computational modules.  Compared with known architectures that employ
traditional hardware computational modules, the proposed architectures
are faster and are area-efficient.  The memory-based architecture is
used to implement the block-based DWT with parallel I/O.  The resulting
architectures are area-efficient and have high throughput and low
latency.  These architectures are suitable for low-power single-chip
implementations which are useful for DWT-based mobile/visual
communication systems.
Paper

AE.1
RESIDUAL SIGNAL IN SUB-BAND ACOUSTIC ECHO CANCELLERS

O. Tanrikulu, B. Baykal, A. G. Constantinides, J. A. Chambers

Sig. Proc. Sec., Dept. of EE. Eng.,
Imperial College of Sci., Tech. and Med.,
London SW7 2BT, UK
Email: o.tanrikulu@ic.ac.uk

All-pass based Power Symmetric QMF-IIR (PS-QMF-IIR) and
Aliasing Cancellation QMF-FIR (AC-QMF-FIR) sub-band
decomposition approaches are studied in the context of
Acoustic Echo Cancellation. The properties of the residual
echo signal are obtained. For both filter types, if the
filters have very sharp transition-bands, the residual echo
signal contains tonal components. It is shown that these can
be efficiently removed by using notch filters. Experimental
results indicate that PS-QMF-IIR filters are better suited
for this application than FIR filter based sub-band
approaches, when combined with the notch filters presented.
Paper

AE.2

AN IMPROVED ECHO SHAPING ALGORITHM FOR ACOUSTIC ECHO CONTROL

Rainer Martin and Stefan Gustafsson
IND, Aachen University of Technology
52056 Aachen, Germany
Tel: +49 241 806984; fax: +49 241 8888186
e-mail: martin@ind.rwth-aachen.de

This paper describes and analyses an improved algorithm for hands-free
telephony which uses an acoustic echo canceller combined with an
additional FIR-filter (called "echo shaping filter") in the sending
path of the hands-free telephone. The algorithm controlling the filter
is motivated by an approximation of an optimal least squares filter.
Simulation results show that the algorithm allows to reduce the order
of the echo canceller significantly, still providing high echo attenuation
and low distortion of the near end speech signal during double talk.
The modulation of the background noise caused by the echo shaping
filter can be reduced by adding artificially generated noise to
the output signal ("comfort noise").


Paper

AE.3
ACOUSTIC ECHO CANCELLATION AND NOISE REDUCTION IN THE FREQUENCY-DOMAIN: A GLOBAL
OPTIMISATION

F.Capman, J.Boudy, P.Lockwood
MATRA COMMUNICATION, Speech Processing Department
rue J.P.Timbaud, 78392 Bois d'Arcy Cedex, BP 26, FRANCE
phone: (+33-1) 34-60-76-84 fax: (+33-1) 34-60-88-32
e-mail: fcapman@matra-com.fr

Abstract: The design of an efficient and robust hands-free system is now
required by the growth of mobile radio and teleconference communications.
The use of Frequency-Domain Adaptive Filters in the context of acoustic
echo cancellation has been extensively studied in the literature. These
algorithms are well-suited for long impulse response modeling and for
correlated input signals like speech. A global optimisation of a frequency-
domain acoustic echo cancellation algorithm with noise reduction is presented
in this paper. This optimisation leads to both reduced complexity and
improved performances when compared to classical cascaded structures.
Paper

AE.4
REALIZATION OF AN ACOUSTIC ECHO CANCELLER ON A SINGLE DSP

Gerard Egelmeers, Piet Sommen and Jacob de Boer

Eindhoven University of Technology (TUE)
P.O.Box 513, 5600 MB Eindhoven, The Netherlands
Tel: +31 40 2473634; fax: +31 40 2455674
e-mail: p.c.w.sommen@ele.tue.nl

An Acoustic Echo Canceller (AEC) based on the Decoupled Partitioned
Block Frequency Domain Adaptive Filter (DPBFDAF) [3,4] is implemented
on a single Digital Signal Processor (DSP), the TMS320C30. This
flexible setup makes it possible to choose the sample frequency (fs),
the number of coefficients (N) of the adaptive filter and the
processing delay independent of one another (only limited by the
total complexity). Two implementation examples are given: one with
N=2016 and fs=7 kHz with a processing delay of 1.6 msec., the other
one with N=2560 and fs=13kHz with a processing delay of 6.5 msec.
It is shown that the setup works both for a white noise input signal
and a real speech signal.
Paper

AE.5
SUBBAND ACOUSTIC ECHO CONTROL USING NON-CRITICAL FREQUENCY SAMPLING

P. A. Naylor and J. E. Hart

Dept. Electrical and Electronic Engineering, Imperial College, London, UK.
email: p.naylor@ic.ac.uk

Aliasing is often generated in critically decimated subband schemes which
can reduce the performance of subband adaptive algorithms. This paper
investigates non-critical decimation schemes in which the generation of
aliasing in the subbands is avoided by down-sampling the subband signals by
a smaller factor than would normally be expected, thereby allowing for
analysis filters with finite transition bands. The implementations of two such
non-critical schemes are presented, one using FIR and one using IIR filter
banks. Simulation results for acoustic echo control using both USASI noise
and male speech signals show the non-critical schemes performance
in comparison to critically decimated filter bank approaches.
Paper

AP.1
SIMULTANEOUS SCHUR DECOMPOSITION OF SEVERAL MATRICES TO ACHIEVE
AUTOMATIC PAIRING IN MULTIDIMENSIONAL HARMONIC RETRIEVAL PROBLEMS

Martin Haardt (1), Knut Hueper (1), John B. Moore (2), and
Josef A. Nossek (1)

(1) Institute of Network Theory and Circuit Design,
Technical University of Munich, D-80290 Munich, Germany
Phone:  +49 (89) 289-28511
Fax:    +49 (89) 289-68504
E-Mail: maha@nws.e-technik.tu-muenchen.de
(2) Department of Systems Engineering,
Australian National University, Canberra ACT 0200, Australia

This paper presents a new Jacobi-type method to calculate a
simultaneous Schur decomposition (SSD) of several real-valued,
non-symmetric matrices by minimizing an appropriate cost function.
Thereby, the SSD reveals the average eigenstructure'' of these
non-symmetric matrices.  This enables an R-dimensional extension of
Unitary ESPRIT to estimate several undamped R-dimensional modes or
frequencies along with their correct pairing in multidimensional
harmonic retrieval problems.  Unitary ESPRIT is an ESPRIT-type
high-resolution frequency estimation technique that is formulated in
terms of real-valued computations throughout.  For each of the R
dimensions, the corresponding frequency estimates are obtained from
the real eigenvalues of a real-valued matrix.  The SSD jointly
estimates the eigenvalues of all R matrices and, thereby, achieves
automatic pairing of the estimated R-dimensional modes via a
closed-form procedure, that neither requires any search nor any other
heuristic pairing strategy.  Finally, we show how R-dimensional
harmonic retrieval problems (with R > 2) occur in array signal
processing and model-based object recognition applications.
Paper

AP.2
A UNIFIED APPROACH TO ROBUST ADAPTIVE BEAMFORMING
IN MOVING JAMMER ENVIRONMENT

Alex B. Gershman, Ulrich Nickel, Johann F. Bohme

Electrical Engineering Dept., Ruhr University, Bochum, Germany
Electronics Dept., FGAN-FFM, Wachtberg, Germany
e-mail: gsh@sth.ruhr-uni-bochum.de

The performance of adaptive beamforming algorithms is known
to degrade in rapidly moving jammer environments.
This degradation occurs due to the jammer motion that
may bring the jammers out of the sharp nulls of the adapted
directional pattern. Below, we develop a unified approach
allowing to make a wide class of adaptive array algorithms
robust against possible jammer motion. This is achieved by
means of artificial broadening of the null width in all jammer
directions. Data-dependent sidelobe derivative constraints are
used which do not require any a priori information about the
jammers. The robust modifications of several well known
Paper

AP.3
AN ADAPTIVE ESPRIT ALGORITHM BASED ON PERTURBATION
OF UNSYMMETRICAL MATRICES

Qing-Guang Liu and Benoit Champagne

INRS-Telecommunications
16 Place du Commerce
qingliu@inrs-telecom.uquebec.ca

ABSTRACT
Many subspace updating algorithms based on the eigenvalue
decomposition (EVD) of array covariance matrices have been
proposed and used in high-resolution array processing algorithms
in recent years. In some applications (i.e. ESPRIT algorithms),
however, the EVD of an unsymmetrical matrix is also needed.
In this paper, an EVD updating approach for an unsymmetrical matrix
is presented based on its first-order perturbation analysis.
By jointly using this approach and a subspace updating method in
an ESPRIT algorithm, a completely adaptive ESPRIT algorithm is
obtained. The evaluation of the complexity and the performance
of this algorithm is given in the paper.
Paper

AP.4
Title:
AN ALGORITHM FOR MULTI-SOURCE BEAMFORMING AND MULTI-TARGET TRACKING: FURTHER
RESULTS

Authors:
Sofiene AFFES (1),(3), Saeed GAZOR (2) and Yves GRENIER (3)

Affiliations:
(1) INRS-Telecommunications, 16, Place du Commerce, Ile des Soeurs, Verdun,
e-mail: affes@inrs-telecom.uquebec.ca
(2) Isfahan University of Technology, Electrical Engineering Dept, Isfahan,
Iran
(3) ENST, Dept Signal, 46 rue Barrault, 75634 Paris, Cedex 13, France

Abstract:
We herein propose an optimal beamformer for the extraction and the tracking
of partially- or fully-coherent sources in colored noise. We adaptively
implement it in a simple structure and combine it with a source-subspace''
tracking procedure. We finally show its effectiveness and its fast tracking
capacity by simulations.
Paper

AP.5

ARRAY SELF CALIBRATION: IDENTIFIABILITY ISSUES

Pierre Comon (*) and Laurent Deruaz

Thomson-Sintra ASM, BP157, F-06903 Sophia-Antipolis Cedex
comon@asm.thomson.fr
(*) also I3S-CNRS, 250 av Einstein, Sophia-Antipolis, F-06560 Valbonne
http://wwwi3s.unice.fr  comon@alto.unice.fr

Array self calibration consists of identifying array shape distortions
and deviations to gain and phase sensor responses, in an unknown
source field. Conditions of local identifiability of these parameters
are established (small perturbations), and turn out to depend on the
type of array (ie linear, surface, volume) and the type of field (ie
near or far). The minimal number of sources and sensors is calculated
in each case, and the nature of the remaining degrees of freedom is
interpreted (eg translation, rotation). With an additional knowledge,
that can be provided by a manoeuvre or by a perfect sensor, it is
shown that the latter parameters can be in turn identified.
Paper

AP.6
MULTIPLE SIGNAL DETECTION AND PARAMETER ESTIMATION USING SENSOR ARRAYS
WITH PHASE UNCERTAINTIES

D. Maiwald and U. Nickel

FGAN-FFM, Neuenahrer Str. 20, D--53343 Wachtberg
email: maiwald@elserv.ffm.fgan.de

In this paper a procedure is outlined for performing both sensor array
calibration and signal detection/direction of arrival estimation
simultaneously. The source directions are unknown. Sensor array
calibration is done using a least squares technique. Signal detection
and direction of arrival estimation is performed by a multiple test
procedure based on $F$-tests.
The algorithm is studied by simulations and by numerical experiments
with data measured by an experimental radar array with $8$ elements.
Paper

AP.8
A GENERALIZED CORRELATION FUNCTION FOR MAGNIFIED/REDUCED SIGNALS

Axel Busboom, Hans Dieter Schotten, and Harald Elders-Boll

Institut fuer Elektrische Nachrichtentechnik
RWTH Aachen, D-52056 Aachen, Germany
Tel: +49 241 807678; fax: +49 241 8888196
e-mail: busboom@ient.rwth-aachen.de

A generalization of the correlation function is explored which, besides a
relative time shift between the signals to be correlated, also takes into
account different scalings on the time axis (i.e., magnification/reduction).
It is shown how the generalized correlation function for continous signals
can be sampled and computed without loss of information and thus can be
described by discrete-time signals.  Envisaged applications comprise coded
aperture imaging, measurement, radar, and digital communications.  Special
attention is paid to tomographic imaging using coded apertures. It is demonstrated
how individual slices of an object can be reconstructed by correlating the
recorded image with suitably designed decoding filters using the generalized
correlation function.
Paper

AP.9

OPTIMAL TIME INVARIANT AND WIDELY LINEAR SPATIAL FILTERING FOR RADIOCOMMUNICATIONS

Pascal Chevalier

Thomson-CSF-Communications, 66 rue du Foss Blanc, 92231 Gennevilliers, France
Tel: 33 1 46 13 26 98 ; Fax: 33 1 46 13 25 55

The classical optimal array filtering problem assumes stationary signals and
consists to implement a complex linear and Time Invariant (TI) filter, optimizing a second
order criterion at the output under some possible constraints. Optimal for stationary signals
this approach is sub-optimal for non stationary signals for which the optimal complex filters
are Time Variant (TV) and, under some conditions of non circularity, Widely Linear (WL).
The purpose of this paper is to present the interest of WL structures of spatial filtering
with respect to linear ones in non stationary radiocommunications environments.
Paper

AS.1
A NEW ROBUST ADAPTIVE STEP SIZE LMS ALGORITHM

Dimitrios I. Pazaitis and Anthony G. Constantinides

Department of Electrical and Electronic Engineering,
Signal Processing Section, Imperial College, Exhibition Road, London SW7 2BT
e-mail : {d.pazaitis, a.constantinides}@ic.ac.uk

In this contribution a new robust technique for adjusting the step size
of the Least Mean Squares (LMS) adaptive algorithm is introduced. The
proposed method exhibits faster convergence, enhanced tracking ability
and lower steady state excess error compared to the fixed step size LMS
and other previously developed variable step size algorithms, while retaining
much of the LMS computational simplicity.
A theoretical behaviour analysis is conducted and equations regarding
the evolution of the weight error vector correlation matrix together with
convergence bounds are established. Extensive simulation results support
the theoretical analysis and confirm the desirable characteristics of
the proposed algorithm.
Paper

AS.2
A NON STATIONARY LMS ALGORITHM FOR ADAPTIVE TRACKING
OF A MARKOV TIME-VARYING SYSTEM

M. TURKI, M. JAIDANE-SAIDANE
L.S.Telecoms, ENIT, Campus Universitaire, Le Belvedere, Tunis, TUNISIA
Telephone: (216)1514700; E-Mail: Jaidane@enit.rnrt.tn

Abstract

We propose in this paper a new adaptive algorithm which is
designed to track system represented by a filter which has a
P order markovian time evolution. The Non Stationary LMS
(NSLMS) algorithm is able to identify the unknown order
and parameters of the markov model. An analysis of the
performances of the adaptive filter when the input is i.i.d.
shows that the NSLMS presents better performances than
the classical LMS. In particular, this superiority occurs
when the system time evolution is so fast that the tracking
with LMS is harmful.
Paper

AS.3
ANALYSIS OF AN LMS ADAPTIVE FEEDFORWARD CONTROLLER FOR PERIODIC DISTURBANCE
REJECTION: NON-WIENER SOLUTIONS FOR THE LMS ALGORITHM WITH A NOISY REFERENCE-REVISITED

Neil  J.  Bershad (1)         and       Jose Carlos M. Bermudez (2)

(1) Department of Electrical and Computer Engineering, University of California,
(2) Laboratorio de Intrumentacao Eletronica (LINSE), Departamento de Engenharia
Eletrica, Universidade Federal de Santa Catarina, C.P. 476, 88.040-900,
Florianopolis, SC, Brazil, bermudez@linse.ufsc.br

LMS adaptive cancellation has been found to be effective in various applications
of active noise control of periodic disturbances. A deterministic periodic
waveform can be used for the reference when the period of the disturbance
is known a priori.  However,  the algorithm behavior is determined by
so-called Non-Wiener solutions. This paper presents a new vector subspace
model for simplifying the analysis of the Non-Wiener behavior.  The LMS
weights are modelled as a deterministic time-varying mean plus a zero-mean
fluctuating part.  Each weight component is analyzed separately with the
subspace model.
Paper

AS.4
A DESIGN METHOD FOR OVERSAMPLED PARAUNITARY DFT FILTER BANKS USING HOUSEHOLDER
FACTORIZATION

K.Kajita, H.Kobayashi, S.Muramatsu, A.Yamada and H.Kiya

Dept. of Elec. & Info. Eng., Tokyo Metropolitan University(e-mail:kajita@isys.eei.metro.ac.jp)

In this work, we propose a design method for oversampled FIR DFT filter banks
which have the paraunitary property, where the number of channel M is the multiple
of decimation ratio D and the filter length is the multiple of M. Our proposed
method is based on Householder factorization, which can keep the perfect reconstruction
condition and the paraunitary property of filter banks in optimization process.
In addition, we examine the linear phase property for oversampled DFT filter
banks, and the design method of oversampled linear phase DFT filter banks.
In order to show the effectiveness of our method, we give some design examples.
Paper

AS.5
ADAPTIVE+DARWINIAN APPROACH FOR THE ESTIMATION AND TRACKING OF TIME DELAYS

Armando Malanda Trigueros
Anibal R. Figueiras-Vidal
Gerald Cain

Universidad Publica de Navarra (Spain). malanda@upna.es
University of Westminster (U.K.). gerry@cmsa.westminster.ac.uk

Abstract
The problem of time delay estimation is tackled with three different
algorithms: a gradient-like scheme, a Darwinian Algorithm (a global
optimisation procedure inspired on Nature's evolution mechanisms) and
a third approach, mixture of the previous two. While the gradient scheme
easily finds an accurate estimate when easily initialised, it misleads
the track when badly initialised or when jumps occur in the delay. The
Darwinian algorithm appears more robust to delay changes but too slow and
less accurate. Our combined solution outperforms the other two in conver-
gence capabilities, without notably degrading accuracy nor speed.
Paper

AS.6
AGAINST REFERENCE SIGNAL POWER FLUCTUATION AVAILABLE FOR
ACOUSTIC ECHO CANCELLER SYSTEMS

Kensaku FUJII and Juro OHGA
Multimedia Systems Laboratories (L40), Fujitsu
Laboratories Ltd.
4-1-1 Kamikodanaka, Nakahara-ku,--Kawasaki, 211-88, Japan
Tel: +88-44-777-1111, Fax: +88-44-754-2741,
fujiken@flab.fujitsu.co.jp

The ERLE (echo return loss enhancement)
iterates greatly up and down, if the
the reference signal power fluctuation.
This paper presents a method of always
maintaining the specified ERLE, even when
the adjustment is continued in voiceless
noise terms. The method is based on the
'summational' NLMS (normalised least
mean square) algorithm in which the
coefficients are updated after the
reference signal norm, and the product
of the residual echo and the reference
signal have been summed up for continues
iterations (a block). The SNLMS
algorithm can keep the ERLE at the
specified level, if the coefficients are
updated after the summed norm has been
amounted to a value which was evaluated
from a given surrounding noise power.
Paper

AS.7
A Cost Function for Constant Amplitude Signals based on
Statistical Referencett

Josep Sala-Alvarez
Department of Signal Theory and Communications (GPS)
Universitat Politecnica de Catalunya
c/ Gran Capit s/n, Modul D5
08034 Barcelona, Spain
Tel: +34-3-401 64 40; Fax: +34-3-401 64 47
E-mail l: alvarez@gps.tsc.upc.es

ABSTRACT

The equalisation of constant amplitude signals is considered in the
scope of this paper. A criterion based on the probability density
function (pdf) of the signal of interest is proposed. The objective
is to derive a suitable soft-decision scheme, more robust than the
classical CMA algorithm that ensures recoverability of the signal.
Paper

AS.8
ON THE PROBLEM OF BLIND EQUALIZATION CONSIDERING ABRUPT CHANGES IN THE
CHANNEL CHARACTERISTICS

Catharina Carlemalm, Bo Wahlberg

S3-Automatic Control
Royal Institute of Technology (KTH)
S-100 44 Stockholm
SWEDEN
cath@s3.kth.se, bo@s3.kth.se

The problem of blind equalization in a digital communication system is
considered. Unfortunately, the circuit might suffer from abrupt changes.
Thus, it is criticalnot to ignore this phenomenon when the problem of blind
equalization is analyzed. The proposed method, which is based on an Ito
stochastic differential calculus approach, describes the dynamics of
the output signal with an infinite impulse response (IIR) model where the
involved taps are modeled as time-varying cadlag (continu a droite limites a
gauche) processes. Therefore, nonlinear and time-variant changes in
the channel characteristics are included.
Paper

AS.9
SOURCE INDEPENDENT BLIND EQUALIZATION WITH FRACTIONALLY-SPACED SAMPLING

Joao Gomes, Victor Barroso

Instituto Superior Tecnico - Instituto de Sistemas e Robotica
Av. Rovisco Pais, Torre Norte 7
1096 Lisboa Codex, Portugal
Tel: +351-1-8418296 Fax: +351-1-8418291
jpg@isr.ist.utl.pt, vab@isr.ist.utl.pt

A generalization of the super-exponential blind equalization algorithm for
fractionally-spaced sampling is presented. Taking advantage of the
increased degrees of freedom in selecting higher order statistics of
cyclostationary signals, two different cost functions are proposed for
blind equalization. One of them allows the inverse of a bandlimited
continuous channel to be identified without aliasing, and the other leads
to a blind counterpart of a decision-directed fractionally-spaced equalizer
(FSE). Simulation results document the performance of these algorithms.
Paper

AS.10
SOFT DECISION SOLUTION TO ILL CONVERGENCE
OF BLIND DECISION FEEDBACK EQUALIZERS

Sofiane Cherif(l)(2), A/Meriem Jaidane(l), Sylvie Marcos(3)

(1) Laboratoire des Systemes de Telecommunications, ENIT, BP 37, Le Belvedere-Tunis, TUNISIA
Tel : + 216 (1) 514700; fax : + 216 (1) 510729; e-mail : jaidane@enit.rnrt.tn
(2) Ecole Superieure des Postes et des Telecommunications de Tunis, 9083 Cite El Ghazala, TUNISIA
Tel : + 216 (1) 762000; fax : + 216 (1) 762819; e-mail : cherif@espttn.esptt.tn
(3) Laboratoire des Signaux et Systemes, CNRS-ESE, 91192 Gif/Yvette ceded FRANCE
Tel : + 33 (1) 69851729; fax : + 33 (1) 69413060; e-mail : marcos@lss.supelec.fr

ABSTRACT

Decision Feedback Equalisers ( DFE) for blind equalization are
subject to ill-convergence. In this paper we prove that the
algorithms may be blind to the global minimum due to the error
surface structure. The use of a soft decision in the decision device
during a pseudo-training phase solve partially the problem of
ill-convergence of DFE.
Paper

BI.1
WIDEBAND BLIND IDENTIFICATION AND SEPARATION OF INDEPENDENT SOURCES

Wang Jun

DSP Division, Department of Radio Engineering
Southeast University
Nanjing 210096, P.R.China
e-mail: cwwu@seu.edu.cn

Abstract: Two higher-order spectra methods, one bispectra and one trispectra,
for solving the wideband blind  identification and signal separation problem
are presented.  The methods are universal in the sense that they does
not impose any restrictions on the probability ditribution of the input
signals provided that they are asymmetrically distributed for the bispectra
method and non-Gaussian for the trispectra one. Two criteria, which state
sufficient conditions for identification and  sepapration, have been proved.
Algorithms are developed based on the criteria, whose efficiency is verified
by the simulations.
Paper

BI.2
SUBSPACE METHOD FOR BLIND SEPARATION OF SOURCES IN CONVOLUTIVE MIXTURE.

Ali MANSOUR (1,3), Christian JUTTEN (1,3, 4) and  Philippe LOUBATON (2,3)

1  INPG-TIRF, 46 avenue F\'{e}lix Viallet, 38031 Grenoble Cedex (France)
2 Univ. de Marne la Vall\'{e}e, 2 rue de la Butte Verte, 93166 Noisy-Le-Grand
Cedex (France)
3  GdR Traitement du Signal et des Images, CNRS
4 Professor in Institut des Sciences et Techniques de Grenoble (ISTG)
of Universit\'e Joseph Fourier.

mansour@tirf.inpg.fr
chris@tirf.inpg.fr
loubaton@pekin.univ-mlv.fr

For the convolutive mixture, a subspace method to separate the sources
is proposed. It is showed that after using only the second order statistic
but more sensors than sources, the convolutive mixture can be itentified
up to instantaneou mixture. Furthermore, the sources can be separated
by any algorithm for instantaneous mixture (based in generally on the
fourth order statistics).
Paper

BI.3
BLIND SEPARATION OF WIDE-BAND SOURCES :
APPLICATION TO ROTATING MACHINE SIGNALS

V. Capdevielle, Ch. Serviere, J-L. Lacoume

CEPHAG serviere@cephag.observ-gr.fr

We propose an extension of the narrow band source separation algorithms
to the case of wide band sources, which is developed in frequency domain.
We mainly focus on the separation of convolutive mixtures of rotating
machine noises and develop two specific points. In the first point, we
study the feasibility of the separation of periodic signals, with regard
to the hypothesis of random and non gaussian sources. The second point
consists in the reconstruction of the spectra of the estimated sources
from the signals identified at each frequency bin. Indeed, the source
associated to the ith identified signal is not necessarily the same from
one frequency bin to another. In this paper, we theoretically prove the
feasibility of the separation of rotating machine noises and propose a
solution in order to reconstruct the source spectra. The algorithm is
then illustrated with experimental results, including the procedures of
separation and reconstruction.
Paper

BI.4
BLIND SOURCE SEPARATION BY SIMULTANEOUS THIRD-ORDER TENSOR DIAGONALIZATION

Lieven De Lathauwer, Bart De Moor, Joos Vandewalle

K.U.Leuven - E.E. Dept.- ESAT - SISTA
Kard. Mercierlaan 94, B-3001 Leuven (Heverlee), Belgium
tel: 32/16/321805 	fax: 32/16/321986
e-mail: Lieven.DeLathauwer@esat.kuleuven.ac.be

We develop a technique for Blind Source Separation based on simultaneous
diagonalization of (linear combinations of) third-order tensor slices'' of the
fourth-order cumulant. It will be shown that, in a Jacobi-type iteration scheme,
the computation of an elementary rotation can be reformulated in terms of a
simultaneous matrix diagonalization.
Paper

BI.5

SECOND ORDER BLIND IDENTIFICATION OF CONVOLUTIVE MIXTURES
WITH TEMPORALLY CORRELATED SOURCES: A SUBSPACE BASED APPROACH

A. Gorokhov and P. Loubaton

Telecom Paris, Dept. Signal
46 rue Barrault
75634 Paris Cedex 13 FRANCE

UF SPI (EEA) Universite de Marne la Vallee
2 rue de la Butte Verte
93166 Noisy-le-Grand Cedex FRANCE

This contribution addresses the blind identification of
Multiple Input Multiple Output (MIMO) linear FIR systems
having a number of inputs less than the number of outputs.
Recent publications have proposed an efficient second order
identification method in the Single Input Multiple Output
(SIMO) case. Based on a subspace analysis, it allows a perfect
recovery of the system parameters and excitation in a noise
free environment. In this paper we indicate how to extend the
original subspace based approach to the general MIMO case.
Paper

BI.6

DIRECTION FINDING AFTER BLIND IDENTIFICATION OF SOURCES STEERING VECTORS:
THE BLIND-MAXCOR AND BLIND-MUSIC METHODS

P. Chevalier, G. Benoit and A. Ferrol

Thomson-CSF-Communications, 66 rue du Foss Blanc, 92231 Gennevilliers, France
Tel: 33 1 46 13 26 98 ; Fax: 33 1 46 13 25 55

To find the direction of arrival (DOA) of P sources impinging on an array of N
sensors, actual second and fourth order direction finding (DF) methods try to solve a
P-dimensional problem from the statistics of the data. The purpose of this paper is to
present a new approach of DF, based on a first step of blind identification of sources
steering vectors, aiming, for some of these methods, at reducing the problem dimension
before DF. Two new methods, the Blind-MAXCOR and the Blind-MUSIC methods, are proposed
and their performance are compared to that of MUSIC method.
Paper

BI.7
BLIND BEAMFORMING IN A CYCLOSTATIONARY CONTEXT USING
AN OPTIMALLY WEIGHTED QUADRATIC COST FUNCTION

C. VIGNAT AND P. LOUBATON

Universit de Marne la Valle
Unit de Formation S.P.I.
2 rue de la Butte Verte
93166 NOISY LE GRAND CEDEX
e-mail: vignat@univ-mlv.fr

This paper addresses the problem of blind beamforming
in a cyclostationary context. We show the equivalence
between the SCORE algorithm derived by Gardner et al.,
and the minimization of an optimally weighted quadratic
cost function. This approach allows us to justify,
from a statistical point of view, the
relevance of the SCORE algorithm.
Paper

C.1
VOICE CONTROLLED MOBILE PHONE
FOR CAR ENVIRONMENT

Ivan Bourmeyster(1), Jamil Chaoui(2), Silvio Cucchi(3), Nicola Griggio(3), Alessandro Guido(3),
Giuliano Moroni(3), Anlonello Riccio(3), Marco Stanzani(3), Fabio Valente(3)
(1) Alcatel Mobile Phones,(2) formerly at Alcatel Mobile Phones - 32, avenue Kleber,92707 Colombes, France
Tel:+33 146521706;fax:+33 146528025
(3) Alcatel Corporate Research Centre - Via Trento, 30, 20059 Vimercate (Milano), Italy
Tel: +39 39 686 4077; fax: +39 39 686 3587

ABSTRACT

The development of an application of speech processing in a
car environment is addressed. The main objective is to provide the
user of a vehicular phone with a powerful and friendly
bidirectional vocal interface. In particular, the paper focusses on
the speech recogniser component of the interface as it was
specifically designed and tuned to operate in the very hostile
acoustic environment of a moving car.

The recogniser operates in a fully speaker dependent mode so
enabling the user to store his/her personal agenda of frequent
called parties.

For the training, three repetitions of each vocabulary word are
recommended, although the performances remain still satisfactory
with only two repetitions.

Reliable performance assessment was conducted with
particular attention to the aspect of robustness of the recogniser
against spurious noises. Standard procedures (SAM oriented)
were used to guarantee the repeatability of any test. An outlook
on future improvements is also given.
Paper

C.2

A NEW ERROR CONCEALMENT TECHNIQUE FOR AUDIO TRANSMISSION WITH PACKET LOSS

Alexander Stenger, Khaled Ben Younes, Richard Reng, Bernd Girod

Telecommunications Institute, University of Erlangen-Nuremberg
Cauerstrasse 7, 91058 Erlangen, Germany
stenger@nt.e-technik.uni-erlangen.de
younes@nt.e-technik.uni-erlangen.de
reng@vs-ulm.dasa.de
girod@nt.e-technik.uni-erlangen.de

We present a new error concealment technique for audio transmission
over packet networks with high packet loss rate. Unlike other techniques it
them. This is done by a time-domain algorithm, WSOLA, whose parameters are
redefined so that short audio segments like lost packets can be extended.
Particular attention is paid to the additional delay introduced by the new
technique. For subjective hearing tests, single and double packet loss is
simulated at high packet loss rates, and the new technique is compared to
previous proposals by category judgment and component judgment of sound
quality. Mean Opinion Score (MOS) curves show that sound distortions due to
packet repetition can be reduced.
Paper

C.3
TRANSMISSION OF VARIABLE-RATE ENCODED SPEECH SAMPLES ON

Fulvio Babich, Sergio Carrato and Francesca Vatta

D.E.E.I., University of Trieste
via A. Valerio, 10, 34127 Trieste, Italy
Tel. +39 40 6763458 - 6767147; Fax: +39 40 6763460
e-mail: babich, vatta@univ.trieste.it
e-mail: carrato@imagets.univ.trieste.it

ABSTRACT
This paper presents the performance evaluation of different speech coding
techniques in wireless packet switching networks: the goal of our study
is to increase network capacity while maintaining a smooth degradation
of quality at high loads and heavy interference, in order to make it possible
for different kinds of information to coexist in a single network infrastructure.
In the paper we propose a variable-rate multimode and embedded encoding
technique as effective for handling network congestion and channel impairments
that both cause discarding or erasure of frames of information. Therefore
this approach is important not only in TDMA packet switched communications
with statistical multiplexing (leading to greater efficiency and flexibility
than basic TDMA, that assigns a fixed portion of channel resources to
each user), but also in a CDMA-based mobile system that is strictly limited
by interference.
Paper

C.4
CHANNEL EQUALIZATION USING PARTIAL LIKELIHOOD ESTIMATION
AND RECURRENT CANONICAL PIECEWISE LINEAR NETWORK

Information Technology Laboratory
Department of Computer Science and Electrical Engineering
University of Maryland Baltimore County
Baltimore, MD 21228-5938, USA
Tel: (410) 455-3521; fax: (410) 455-3969

A recurrent canonical piecewise linear (RCPL)  network
is proposed based on the canonical piecewise linear (CPL)
structure and is applied to channel equalization. RCPL
network provides savings in computation and implementation
and has a distinct dynamic behavior completely different
than that of finite duration feedforward structure. The
simulations of multilevel signal equalization demonstrate
the superior performance of RCPL equalizer when compared
to the multilayer perceptron equalizer. For the RCPL
network, it is easy to incorporate the a-priori information
into the network structure. A novel blind algorithm is
presented by combining partial likelihood estimation and
RCPL  structure for the binary communications channel.
The simulation results show that RCPL blind equalizer
outperforms the CMA equalizer by orders of magnitude for
blind equalization of nonlinear communication channels.
Paper

C.5
CHANNEL ESTIMATION FOR TRANSFORM MODULATIONS IN MOBILE
COMMUNICATIONS

Meritxell Lamarca, Gregori Vazquez

Department of Signal Theory and Communications
Polytechnic University of Catalonia (UPC)
Barcelona (SPAIN)
e-mail: xell@gps.tsc.upc.es

This paper deals with data-aided channel estimation in systems using OFDM
modulation. We formulate a pilot symbol-based channel estimator and compare it with
the pilot tone one proposed in [1]. Although this paper focuses in flat fading mobile
channels, the results could be easily applied to OFDM systems operating in frequency
selective channels.
Paper

C.6
DETECTION AND COMPENSATION FOR DISRUPTIVE NON-LINEAR TRAFFIC-FLOW DYNAMICS
IN COMMUNICATION NETWORKS

D.P.A.Greenwood and R.A.Carrasco

School Of Engineering
Staffordshire University
United Kingdom
d.greenw@bss10a.staffs.ac.uk
r.carras@bss10a.staffs.ac.uk

Abstract: A method has been developed for the monitoring of traffic flow
behavioural dynamics in distributed communication networks and the provision
of results from this process to a distributed neural control mechanism which
facilitates localised adaptive traffic routing in order to maintain or regain
flow stability. It has been shown by simulation how the novel method improves
network performance and efficiency beyond that of conventional techniques.
Paper

C.7
SIMULATION OF LAND MOBILE SATCOM LINKS USING DIFFERENT ORBITS
AND MODULATION MODES

Marcel Kohl
Friedrich Jondral

Universitaet Karlsruhe, Nachrichtensysteme
D-76128 Karlsruhe, Germany
Tel: +49 721 6083748; fax: +49 721 6086071
e-mail:  kohl@inss1.etec.uni-karlsruhe.de

The use of SATCOM systems is an essential part of today's
worldwide communications. As the portion of satellite orbits
in low altitudes increases, Doppler shifts often influence
the received signal. Prior to the removal of this eefect,
the exact course and the amount of the Doppler must be known.
Therefore this paper derives the equations to calculate the
orbit and the Doppler shift and shows the behaviour and the
effects caused by LEO and HEO satellites. Finally a method
is proposed to compensate this influence.
Paper

C.8
SCRAMBLING AND ERROR CORRECTION BY MEANS OF LINEAR TIME-VARYING FILTERS

Alban Duverdier and Bernard Lacaze

National Polytechnics Institute of Toulouse
LEN7/GAPSE, 2 rue Camichel, 31071 Toulouse, France
tel: (33) 61 58 83 67
Fax:(33) 61 58 82 37
email: duverdie@len7.enseeiht.fr

In numerous communication applications, it is desirable to scramble the
contents of the information. In addition, we seek to design a scrambling
system which has maximum immunity to additive noise. This paper presents a
method of analogue signal scrambling/unscrambling by means of linear
periodic time-varying filters for any frequency selective noise. It is well
known that linear periodic time-varying filters transform a stationary
process into a cyclostationary signal. This thus spreads the spectral
representation of the input process. The original part of the paper consists
of using this property to reconstruct an initial band-limited process
without error for any frequency selective noise.
Paper

C.9
A FAST LUT+CMAC DATA PREDISTORTER

Francisco J. Gonzalez-Serrano (*) and Anibal R. Figueiras-Vidal (**) and
Antonio Artes-Rodriguez (**)

(*) Grupo de Teoria de Senal
Departamento de Tecnologias de las Comunicaciones
ETSI Telecomunicacion.
36200 VIGO-SPAIN.
Tel : +(34) 86 81 2130
Fax : +(34) 86 81 2116
E-mail : frank@tsc.uvigo.es
and
(**) Grupo de Teoria y Tratamiento de Senal
DSSR - ETSI Telecomunicacion.
Tel : +(34) 1 549 5700
Fax : +(34) 1 336 7350
E-mail : antonio@gtts.ssr.upm.es

The subject of this communication is the compensation of
nonlinearity is caused by the High Power Amplifier (HPA), typically
working close to its saturation point because of energy
constraints. This paper deals with the design of CMAC-based
predistorters for application in digital transmission over nonlinear
channels with memory. A novel hybrid structure composed of a
Look-Up-Table in parallel with a CMAC network is proposed. Finally,
a performance analysis for typical radio channels is presented.
Paper

C.10
DESIGN OF PULSE SHAPING FILTERS AND THEIR APPLICATIONS IN RADIO SYSTEMS

Jong-Jy Shyu, Yo-Chuan Lai

Department of Computer Science and Engineering
Tatung Institute of Technology, Taipei, Taiwan
e-mail: jshyu@cse.ttit.edu.tw

Partial-response signaling is known as correlative
level coding wherein the constraint on waveforms is relaxed so as to allow a
controlled amount of ISI.
In this paper, the Lagrange multiplier approach, which is easy to
incorporate both time- and frequency-domain constraints by minimizing a
quadratic measure of the error in the design bands, is applied to design a
large class of such digital filters for communication in this paper.
Also, the iterative Lagrange multiplier approach combining the Lagrange
multiplier approach and a tree search algorithm is proposed for
designing discrete coefficient pulse shaping FIR digital filters.
System experiments such as an SSB radio system using partial response
signaling are demonstrated to present the usefulness of the proposed algorithm.
Paper

D.1

APPROXIMATE MAXIMUM LIKELIHOOD ESTIMATION IN LASER VELOCIMETRY.

Olivier Besson and Frederic Galtier.

ENSICA, Department of Avionics and Systems.
1, Place Emile Blouin. 31056 Toulouse - France.
besson,galtier@ensica.fr

Abstract: In this paper, we study the estimation of signals of the form
$%s(t)=A.\exp \left\{ -2\alpha ^2f_d^2t^2\right\} .\cos \left( 2\pif_dt\right)$
which are encountered in the measurement of particles velocity in a flow by
means of laser Doppler velocimeters. We derive anApproximate Maximum
Likelihood Estimator of the parameters A  and $f_d$ in the model considered.
The algorithm is based upon replacing the first and second-order derivatives
of the log-likelihood function by approximated and easy to compute expressions.
Numerical examples illustrate the performance of the proposed method and
quantify the influence of the sample size, the frequency $f_d$ and the
parameter $\alpha$. They show that the estimator is statistically efficient
in a wide range of scenarios.
Paper

D.2
INSTRUMENTAL VARIABLE SOLUTION TO AN EXTENDED FRISCH PROBLEM

Petre Stoica, Mats Cedervall, Joakim Sorelius and Torsten Soderstrom

Systems and Control Group, Uppsala University PO Box 27, S-751 03
Uppsala, Sweden; Tel: +46 18 183074; fax: +46 18 503611; e-mail:
petre.stoica@syscon.uu.se

In signal processing and time series analysis applications we often
encounter cases in which a number of (noise-free) variables are linearly
related and we want to make inferences on the number and the form of the
linear relations among those variables from noisy observations of them. The
Frisch problem is concerned with the aforementioned inferences under the
assumption that the components of the observation noise vector are mutually
uncorrelated. In this paper we extend the Frisch problem by allowing the
noise vector components to be correlated in an arbitrary (and unknown) way.
The EXtended FRIsch problem of this paper is called EXFRI for short. To make
EXFRI solvable we basically assume that the observation noise is temporally
white whereas the noise-free signals are temporally correlated. We show
that, under the assumptions made, the EXFRI problem has a computationally
simple and statistically elegant Instrumental Variable (IV) solution,
which is essentially based on a canonical correlation decomposition
procedure.
Paper

D.3
BERNOULLI-GAUSSIAN DECONVOLUTION IN NON-GAUSSIAN NOISE, CONTRIBUTION OF WAVELET DECOMPOSITION

H.Rousseau and P.Duvaut

E.T.I.S. - E.N.S.E.A.,
6, avenue du Ponceau,
95014 CERGY Cedex
e-mail : rousseau@ensea.fr

We introduce a method to restore Bernoulli-Gaussian processes
immerged in a non-gaussian noise. It uses wavelet decomposition to gaussianize'' the noise.
The convergence, after wavelet projection, of some non-gaussian noise to
a gaussian noise quantifies the quality of the
gaussianization'' effect of the wavelet. This property is used to apply a
Bernoulli-Gaussian algorithm at each scale of wavelet decomposition.
After, we use a fusion strategy to
merge all results. We obtain also a new deconvolution algorithm which is
very performant, for all satistical noises, when the noise variance is not
well estimated. When the noise variance is correctly estimated, it improves
the classical Bernoulli-Gaussian algorithm for strongly non-Gaussian noises.
Paper

D.4
MMSE EQUALIZERS FOR MULTITONE SYSTEMS WITHOUT GUARD TIME

L. Vandendorpe

UCL Communications and Remote Sensing Laboratory,
2, place du Levant, B 1348 Louvain-la-Neuve, Belgium.
Phone: +32 10 47 23 12 - Fax: +32 10 47 20 89 -
E-Mail : vandendorpe@tele.ucl.ac.be

Recently the concept of multitone modulation or OFDM has received much
attention. For such a modulation, the dispersiveness of the channel is
classically solved by the technique of guard time. In the present paper
we investigate the performance of OFDM without guard time but with MIMO
equalization. Linear and decision-feedback structures structures are derived
for an MMSE criterion and their performance is assessed by means of their
steady-state behavior. Symbol rate equalizers following channel matched
filters are derived and investigated. It is shown that equalized OFDM
outperforms OFDM with guard time.
Paper

D.5
MSE-BASED REGULARIZATION APPROACH TO
RANK DETERMINATION IN CLS AND TLS ESTIMATION

H. Kagiwada, Y.Aoki, J. Xin andA.Sano

Department of Electrical Engineering, Keio University
3-14-1 Hiyoshi, Kohoku-ku, Yokohama 223, Japan
Tel: +81 45 563 1141; fax: +81 45 563 2773
e-mail: sano@sano.elec.keio.ac jp

The corrected least squares (CLS) approach using an over-
determined model is investigated to decide the number of
sinusoids in additive white noise. Like the total least squares
(TLS) approach, the CLS estimation is different from the
ordinary least squares (LS) method in that the noise variance
is subtracted from the diagonal elements of the correlation
matrix of the noisy observed data. Therefore the inversion of
the resultant matrix becomes ill-conditioned and then adequate
trunc at i on of the eigenv alue decompositi on (EVD) s hould be
done. This paper clarifies how to simultaneously estimate the
noi se variance and truncate the eigenvalues , since they are
mutually dependent. By introducing a multiple number of
regulanzation parameters and determining them to minimize
the MSE of the model parameters, we can give an optimal
scheme for the truncation of eigenvalues. Furthermore, an
iterative algorithm using only observed data is also clarified.
Paper

D.6
ROBUSTNESS ANALYSIS OF MUSIC AND ESPRIT FREQUENCY ESTIMATORS FOR
SINUSOIDAL SIGNALS WITH TIME-VARYING AMPLITUDE

Olivier Besson and Petre Stoica

ENSICA, Department of Avionics and Systems, Place Emile Blouin,
31056 Toulouse, France. besson@ensica.fr
Uppsala University, Systems and Control Group, 75103 Uppsala, Sweden.

Abstract: In this paper, we address the problem of estimating the frequency of
a sinusoidal signal with random, lowpass amplitude. We propose to use MUSIC
and ESPRIT frequency estimators as if the signal had a constant amplitude.
The aim of the paper is to analyze the degradation of performance induced by
the aforementioned mismodelling. Unified expressions for the bias and
variances of the MUSIC and ESPRIT frequency estimators are derived under the
hypothesis of small bandwidth of the signal envelope. Numerical simulations
illustrate the agreement between theoretical and empirical results and study
the influence of the envelope bandwidth onto the frequency estimation
performance.
Paper

D.7
HOS BASED DETECTORS FOR PERIODIC SIGNALS

P.R. White, N. Khalili

ISVR, University of Southampton,
Highfield, Hants, U.K., SO17 1BJ
Tel.: +44 1703 592274, Fax: +44 1703 593033
email: prw@isvr.soton.ac.uk

This paper discusses algorithms for the detection of
periodic pulse-like signals. Such signals exhibit phase
as well as frequency coupling and are thus suitable
for detection using HOS. The algorithm presented herein
can be regarded as an extension to an existing second
order spectral algorithm, to include third order terms.
The results of simulation studies are presented which
demonstrate the performance advantage offered by this
new algorithm.
Paper

D.8
DETECTION OF ABRUPT CHANGES : A TIME-FREQUENCY APPROACH

Helene LAURENT, Christian DONCARLI  and  Philippe POIGNET

Laboratoire d'Automatique de Nantes, U.R.A. C.N.R.S. 823
Ecole Centrale de Nantes/Universite de Nantes
1 rue de la Noe, 44072 NANTES CEDEX, FRANCE
Tel: (33) 40 37 16 00; Fax: (33) 40 37 25 22
e-mail: poignet@lan.ec-nantes.fr

This paper presents a comparison between parametric and non-parametric
approaches of abrupt changes detection in noisy signals. The goal is to
propose an alternative way to be used when the model-based methods do
not work very well because of an unsuitable model structure or a non strictly
stationnary stepwise signal. In this latter case, an analysis of time-frequency
distributions allows the detection of abrupt spectral changes without
any hypothesis and provides some results as good as parametric methods
for the studied type of signals.
Paper

D.9

DETECTION AND ESTIMATION OF CHANGES IN  A   POLYNOMIAL-PHASE
SIGNAL USING THE DPPT

C. Theys, A. Ferrari and G. Alengrin

I3S Universite de Nice-Sophia Antipolis
41, Bd Napoleon III - 06041 NICE cedex - FRANCE
e-mail : theys@unice.fr

This paper is concerned with on-line detection and estimation of changes
in the parameters of a noisy polynomial-phase signal. This problem  arises
in vibration monitoring where the measured signals reflect both the nonstationarities
due to the surrounding excitation, modelled by a polynomial-phase and
the nonstationarities due to changes in the eigen structure, modelled
by a break in the polynomial parameters. Development of a likelihood ratio
test to detect and estimate changes in a polynomial-phase signal requires
accurate estimation of the parameters vector after change, theta1. Use
of the Maximum Likelihood Estimate (MLE) of theta1 is not practically
useful since it involves the optimization of a multi-variable cost function.
We propose to estimate theta1 by using the Discrete Polynomial-Phase Transform
(DPPT) in order to derive a detector having asymptotically the same properties
than the GLR one for a much lower computational cost. Experimental performances,
mean delay to the detection as a function of mean time between false alarms,
will be studied.
Paper

D.10
SYMBOL DECODING BASED ON SIGNAL SUBSPACE DECODING IN MSK

Rafael Ruiz
Margarita Cabrera

Dept. of Signal Theory and Communications, E.T.S.I. Telecomunicacion, UPC.
Apdo. 30002, 08080 Barcelona. SPAIN
e_mail: rafael@gps.tsc.upc.es

ABSTRACT:
The availability of fast processors with architectures tailored to meet
the computational demand of digital signal processing algorithms is widely
applied to demodulation and decodification of CPM signals in some scenes:
Mobiles, AWGN channels,... In this application the number of floating point
operations  executed by each processed symbol is a critical parameter to be
designed, this is to be minimized.  In this paper a method that reduces
significantly the number of operations (until 80%) by symbol for CPM signals
is presented. The decodification stage is performed from the rank reduced
signal subspace obtained by means of an orthogonal decomposition of the signal.
Paper, part 1
Paper, part 2

EI.1
ADAPTIVE NEURAL NETWORKS FOR ROBUST ESTIMATION OF PARAMETERS OF NOISY
HARMONIC SIGNALS

A. Cichocki

FRP Riken - ABS Laboratory, Institute of Physical and Chemical Research, Japan
Tel: +81 48 465 2645; fax: +81 48 462 4633
e-mail: cia@kamo.riken.go.jp

P. Kostyla, T. Lobos, Z. Waclawek

Technical University of  Wroclaw
pl. Grunwaldzki 13, 50-370 Wroclaw, Poland
Tel: +48 71 203448; fax: +48 71 229725
e-mail: lobos@elektryk.ie.pwr.wroc.pl

ABSTRACT
In many applications, very fast methods are required for estimating and
measurement of parameters of harmonic signals distorted by noise. This
follows from the fact that signals have often time varying amplitudes.
Most of the known digital algorithms are not fully parallel, so that the
speed of processing is quite limited. In this paper we propose new parallel
algorithms, which can be implemented by analogue adaptive circuits employing
some neural network principles. The problem of estimation is formulated as an
optimization problem and solved by using the gradient descent method.
Algorithms based on the least-squares (LS), the total least-squares (TLS)
and the robust TLS criteria are developed and compared. The networks process
samples of observed noisy signals and give as a solution the desired parameters
of signal components. Extensive computer simulations confirm the validity and
performance of the proposed  algorithm.
Paper

EI.2
MAXIMUM LIKELIHOOD ESTIMATION OF AR MODULATED SIGNALS

Mounir GHOGHO
National Polytechnics Institute of Toulouse, ENSEEIHT/GAPSE, France
email: ghogho@len7.enseeiht.fr

The desired signal is embedded in both multiplicative and additive noises.
The multiplicative noise is modeled by a Gaussian AR process. Closed forms
expressions are derived for the finite-sample Cramer-Rao bound and for the
maximum likelihood estimator. A cyclic approach is used to initialize the
maximum likelihood algorithm when the signal is a harmonic.
Paper

EI.3
TIME DELAY AND MOTION ESTIMATORS BASED ON
DIGITAL FAST TIME-SCALING OF RANDOM SIGNALS

Gaetano Giunta

INFO-COM Department, University of Rome "La Sapienza",
Via Eudossiana 18, 00184 Rome, Italy
tel.: + 39 6 44585838;  fax: + 39 6 4873300
e-mail: giunta@infocom.ing.uniroma1.it

The estimation of time-delay and time-scaling is required in many signal
processing applications. A parabolic approximation was recently suggested
for fine estimation of time delay from sampled signals. The method directly
extends to scaling estimation by a parallel multi-rate sampling of the
analog received signal. Such rescaling can be implemented by digital techniques
and two efficient algorithms are here devised and analysed.
Paper

EI.4
A SUPER-RESOLUTION METHOD BASED ON THE DISCRETE COSINE TRANSFORMS

Hisashi SAKANE, Kiyoshi NISHIKAWA and Hitoshi KIYA

Dept. of Elec. & Info. Eng., Tokyo Metropolitan University, e-mail: kiya@eei.metro-u.ac.jp

A super-resolution method based on the discrete cosine transform (DCT) is proposed
for a signal with some frequency damage under a type 1 linear-phase (LP) FIR
filter as a damage model. The proposed method can be carried out with real
value operation and is applicable to any DCT in 4 kinds of DCTs. In addtion,
two magnification schemes based on the proposed method to improve the conventional
scheme are described.
Paper

EI.5
ROBUST PARAMETER ESTIMATION FOR PERIODIC POINT PROCESS SIGNALS
USING CIRCULAR STATISTICS

Stephen D. Elton(1) and Benjamin J. Slocumb(2)

(1) Electronics and Surveillance Research Laboratory
Defence Science and Technology Organisation and
Cooperative Research Centre for Robust and Adaptive Systems
P.O. Box 1500, Salisbury, SA 5108, Australia
e-mail: Stephen.Elton@dsto.defence.gov.au
(2) Electronic Systems Laboratory, Georgia Tech Research Institute
Georgia Institute of Technology, Atlanta, GA 30332-0840, U.S.A.
e-mail: Ben.Slocumb@gtri.gatech.edu

We discuss the application of signal parameter estimators for periodic
point process signals with missing data.  The proposed estimation
techniques operate on the observed event arrival time sequence of a
pulse train signal and have application to pulse train signal
classification and signal reconstruction.  The methods we describe are
based on the use of circular statistics and are shown to offer
considerable robustness to a pulse train time series corrupted by
missing pulses.
Paper

EI.6
A METHOD FOR COMPUTING THE INFORMATION MATRIX OF STATIONARY GAUSSIAN
PROCESSES

Jose M. B. Dias   and  Jose M. N. Leitao

Instituto de Telecomunicacoes and
D.E.E.C., Instituto Superior Tecnico
Tel: +351 1 8418464;    fax: +351 1 8418472
Email: edias@beta.ist.utl.pt

This  paper proposes a new method for the efficient computation of the
Fisher information matrix of zero-mean complex stationary Gaussian processes.
Its complexity (measured by the number of floating point operations) is
smaller than the  fastest previously available  procedure.  The key idea
exploited is that the Fisher  information matrix depends only on the sum
of the diagonals of the  inverse covariance matrix derivative (with respect
to the model parameters), rather than on the whole matrix. To obtain the
referred sum, a new efficient technique,  built upon the Trench algorithm
for computing the inverse of a Toeplitz matrix, is  presented.
Paper

EI.7
Title: FULLY BAYESIAN ANALYSIS OF HIDDEN MARKOV MODELS

Authors: Arnaud DOUCET, Patrick DUVAUT

Affiliation: LETI-CEA Technologies Avancees 91191 Gif sur Yvette FRANCE
ENSEA-ETIS Groupe Signal 6, avenue du Ponceau 95014 Cergy Pontoise FRANCE
douceta@ensea.fr - duvaut@ensea.fr

Abstract: In this paper, we present in an unified framework some
applications of stochastic simulation techniques, the Markov chain Monte
Carlo methods, to perform Bayesian inference for a very wide class of hidden
Markov models. Efficient implementation of the Gibbs sampler based on finite
dimensional optimal filters is described. An improved version of this
algorithm is also presented. Two problems of great practical interest in
signal processing are addressed: blind deconvolution of Bernoulli-Gauss
processes and blind equalization of a channel. In simulations, we obtain
very satisfactory results.
Paper

EI.8
Title:	PERFORMANCE ANALYSIS OF A WAVELET BASED WBCAF METHOD FOR TIME DELAY
AND DOPPLER STRETCH ESTIMATION

X. X. Niu    P. C. Ching
Dept. of Electronic Engineering, Chinese University of Hong Kong, Hong
Kong
Tel:	(852) 2609 8275
Fax: 	(852) 2603 5558
Email:	xxniu@ee.cuhk.edu.hk
pcching@ee.cuhk.edu.hk

Y. T. Chan

Abstract:
A wavelet based method for time delay and Doppler stretch estimation has
been proposed. It makes use of the relationship between the wideband cross
ambiguity function (WBCAF) and the cross wavelet transform of the received
signals. This paper derives the Cramer-Rao lower bound (CRLB) and analyses
the performance of the algorithm. It is found that under high SNR, the
method is asymptotically unbiased, and the variances of the estimation
parameters are fairly close to the CRLB. Simulation results are given to
corroborate the theoretical derivation.
Paper

EI.9
A NEW METHOD FOR WAVELETS GENERATION.

A.Martnez-Gonzalez, L. Ortiz-Balbuena, H. Perez-Meana, E. Sanchez-Sinencio*
and J. C. Sanchez-Garca

Universidad Autonoma Metropolitana Iztapalapa ,Depart.of Electrical Engineering,
CBI Division. Av. Michoacn y Pursima. Col. Vicentina, Iztapalapa. C.P.
09340 Mexico, D.F. Mexico. Tel: (525) 725 46 35; Fax: (525) 725 49 02.
e-mail: leob@xanum.uam.mx
* Texas A & M, Department of Electrical Engineering, College Station,
Texas, U.S.A.

Wavelets operators are very important in most practical applications.
Implementation of these operators in software and in commercial DSP hardware
are popular. We are presenting an alternative hardware implementation
of wavelets operators using mixed-mode signal techniques, that is, a judicious
combination of analog and digital hardware implementations. The approach
is general and can be applied to a number of wavelets types.
Paper

EI.10
Trieste paper 074

1-D SAMPLED DATA RECOGNITION WITH AUGMENTED PROGRAMMED GRAMMAR

P.M. Grant, D.T. Lin, J.M. Hannah and R.D. Pringle

Department of Electrical Engineering,
University of Edinburgh,
Edinburgh,
EH9 3JL,
Scotland

Tel: +44 131 650 5569;
fax: +44 131 650 6554;
email pmg@ee.ed.ac.uk

ABSTRACT

This syntactic parser for pattern recognition, uses a
descriptive grammar to test whether data samples fall within
an expected shape or envelope.  The construction of this
recogniser, which is based on an augmented programmed grammar,
is described and its recognition statistics are simulated on
irregularly sampled pattern waveforms.  It is shown to be
able to correctly recognise 1-D waveforms with a wide range
of sizes or scale factors, within a single grammatical representation.
Paper

EII.1
ORDER DETERMINATION OF STATE SPACE SYSTEMS

Anthony G. Place and Gregory H. Allen

Electrical and Computer Engineering Department
James Cook University of North Queensland
Queensland Australia
Tel: +61 7 814299; fax: +61 7 251348
e-mail: Anthony.Place@jcu.edu.au and Gregory.Allen@jcu.edu.au

Recent techniques proposed for the identification of state space models have
focused on using the singular value decomposition of block Hankel input-output
matrices.  In these procedures the order of the system is determined by
examining the singular values and identifying the separation between the
signal'' and noise'' subspaces.  Order determination of state space
systems requires an understanding of what singular value magnitudes are
expected.  This paper examines how system structure and noise levels
affect the magnitude of singular values.  An order selection criterion formed
from the AIC and MDL is also examined.
Paper

EII.2
THE  BEST  ORDER  OF  LONG  AUTOREGRESSIVE  MODELS
FOR  MOVING  AVERAGE  ESTIMATION

P.M.T. Broersen

Department of Applied Physics, Delft University of Technology
P.O.Box 5046, 2600 GA Delft, The Netherlands
phone + 31 15 278 6419,   fax + 31 15 278 4263,
email broersen@tn.tudelft.nl

ABSTRACT
Durbin's method for Moving Average (MA) estimation uses the estimated
parameters of a long AutoRegressive (AR) model to compute the desired
MA parameters. A theoretical order for that long AR model is infinity,
but very high AR orders lead to inaccurate MA models in the finite
sample practice. A new theoretical argument is presented to derive an
expression for the best finite long AR order for a known MA process
and a given sample size. Intermediate AR models of precisely that
order produce the most accurate MA models. This new order differs from
the best AR order to be used for prediction. An algorithm is presented
that enables use of the theory for the best long AR order in known
processes to data of an unknown process.
Paper

EII.3
Title : A CLASS OF REAL-TIME AR IDENTIFICATION ALGORITHMS IN THE CASE
OF MISSING OBSERVATIONS.

Authors : Sina Mirsaidi and Jacques Oksman

Affiliation : SUPELEC, Service des Mesures, Plateau de Moulon, 91192 Gif-sur-Yvette
Cedex, FRANCE.
Tel : (33) 1 69.85.12.12
Fax : (33) 1 69.85.12.34
E-mails : Mirsaidi@soleil.supelec.fr, Oksman@supelec.fr.

Abstract : This paper deals with the problem of adaptive AR estimation
from incomplete observations. The method is based on the optimization
of a weighted squared error criterion. Various approximates of this criterion
lead to different algorithms. The formal description of these algorithms
are given and their performances in stationary and non-stationary environments
are compared.
Paper

EII.4
UNSUPERVISED RESTORATION OF GENERALIZED MULTISENSOR HIDDEN MARKOV CHAINS

Nathalie Giordana and Wojciech Pieczynski
Departement Signal et Image
Institut National des Telecommunications
9 rue Charles Fourier, 91000 Evry cedex France
Tel: (33 1) 60764425; fax: (33 1) 60764433
e-mail: Nathalie.Giordana@int-evry.fr
Wojciech.Pieczynski@int-evry.fr

This work addresses the problem of generalized multisensor Hidden Markov
Chain estimation with application to unsupervised restoration. A Hidden
Markov Chain is said to be generalized'' when the exact nature of the
noise components is not known; we assume however, that each of them belongs
to a finite known set of families of distributions. The observed process
is a mixture of distributions  and the problem of estimating such a generalized''
mixture thus contains a supplementary difficulty: one has to label, for
each state and each sensor, the exact nature of the corresponding distribution.
In this work we propose a  general procedure with application to estimating
generalized multisensor Hidden Markov Chains.
Paper

EII.5
APPLICATION OF HIDDEN MARKOV MODELS TO BLIND CHANNEL ESTIMATION AND DATA
DETECTION IN A GSM ENVIRONMENT

Carles Antn-Haro, Jos A.R. Fonollosa and Javier R. Fonollosa.

Dpt. of Signal Theory and Communications. Universitat Politcnica de Catalunya.
c/ Gran Capit s/n. 08034 Barcelona (SPAIN)
Tel: +34-3-4016454, Fax: +34-3-4016447, e-mail: carles@gps.tsc.upc.es

In this paper, we present an algorithm based on the Hidden Markov Models
(HMM) theory to solve the problem of blind channel estimation and sequence
detection in mobile digital communications. The environment in which the
algorithm is tested is the Paneuropean Mobile Radio System, also known as
GSM. In this system, a large part in each burst is devoted to allocate a
training sequence used to obtain a channel estimate. The algorithm presented
would not require this sequence, and that would imply an increase of the
system capacity. Performance, evaluated for standard test channels, is
close to that of non-blind algorithms.
Paper

EII.6
ESTIMATING PIECEWISE LINEAR MODELS USING COMBINATORIAL OPTIMIZATION
TECHNIQUES

Marco Mattavelli *, Edoardo Amaldi #

* Signal Processing Laboratory, Swiss Federal Institute of
Technology, CH-1015 Lausanne, Switzerland, Tel: +41 21 693 4807,
E-mail: marco.mattavelli@lts.de.epfl.ch.
# School of Operations Research and Center for Applied
Mathematics, Cornell University, Ithaca, NY 14853, USA, E-mail
amaldi@cs.cornell.edu.

A wide range of image and signal processing problems have been
formulated as ill-posed linear inverse problems.  Due to the
importance of discontinuities and non-stationarity, piecewise linear
models are a natural step towards more realistic results.  Although
there have been some attempts to extend classical approaches to deal
with discontinuities, finding at the same time the piecewise
decomposition and the corresponding model parameters remains a major
challenge. A new approach based on partitioning inconsistent linear
systems into a minimum number of consistent subsystems MIN PCS is
proposed for solving ill-posed problems whose formulation as linear
inverse problems with discrete data fails to take into account
discontinuities. In spite of the NP-hardness of MIN PCS, satisfactory
approximate solutions can be obtained using simple but effective
variants of an algorithm which has been extensively studied in the
artificial neural network literature. Our approach presents various
advantages compared to classical alternatives, including a wider range
of applicability and a lower computational complexity.
Paper

EII.7
STRUCTURED TOTAL LEAST SQUARES METHODS IN SIGNAL PROCESSING

Philippe Lemmerling
Sabine Van Huffel
Bart De Moor

Katholieke Universiteit Leuven
philippe.lemmerling@esat.kuleuven.ac.be

In many signal processing applications, one has to solve an
overdetermined system of linear equations Ax=b. The Total Least
Squares (TLS) method finds a Maximum Likelihood (ML) estimate of the
parameter vector x when the noise on the  entries of [A b] is
i.i.d. Gaussian noise with zero mean and equal variance. In many
applications, these last conditions  do not hold because of the
structure present in [A b]. Under those circumstances, the TLS will
not yield a ML estimate of the parameter vector x since the SVD (which
is the standard way to obtain the TLS solution) is not structure
preserving. Therefore, several  structured Total Least Squares
methods have been developed in recent years: the Constrained Total Least
Squares (CTLS) method , the Structured Total Least Squares (STLS)
method  and the Structured Total Least Norm (STLN) method. As opposed
to the ordinary TLS these methods yield a ML estimate of the parameter
vector x, by imposing the structure of the errors on [A b].
Paper

EII.8
TITLE: BAYESIAN DECONVOLUTION OF CYCLOSTATIONARY PROCESSES BASED ON POINT PROCESSES

AUTHORS: Christophe ANDRIEU - Patrick DUVAUT - Arnaud DOUCET

AFFILIATION: ENSEA - ETIS Groupe Signal / 6 avenue du Ponceau 95014 Cergy Cedex France

E-mail: andrieu@ensea.fr - duvaut@ensea.fr - douceta@ensea.fr

ABSTRACT:

In this paper we address the problem of the fully Bayesian deconvolution of a widely
spread class of processes, filtered point processes, whose underlying point process
is a self excited point process. In order to achieve this deconvolution, we perform
powerful stochastic algorithms, the Markov chain Monte Carlo (MCMC), which despite
their power have not been yet widely used in signal processing. We present in this
paper an application to a particular class of weakly cyclostationary processes.
Paper

EII.9
DIFFERENTIAL CEPSTRUM DEFINED ON INTERPOLATED SEQUENCES

Damjan Zazula

University of Maribor
Faculty of Electrical Engineering and Computer Science
Smetanova 17
2000 Maribor
SLOVENIA
Tel.: +386 62 221 112; fax: +386 62 225 013
E-mail: zazula@uni-mb.si

The paper introduces a novel definition of the differential
cepstrum. It is based on the interpolation sequences in the frequency
domain and exists also for the singular signals with no spectral inverse.
Besides, we showed analytically and statistically that such a differential
cepsrtum exhibits lower cepstral aliasing when calculated with the DFT
comparing to the calculation without interpolation. On average, the
improvement is 39 % in case of the interpolation to the half-intervals
and 46 % in case of the quarter-intervals.
Paper

EII.10
MODULATION CLASSIFICATION -- AN UNIFIED VIEW

Peter A.J. Nagy

National Defence Research Establishment, Sweden
P.O. Box 1165, S-581 11 Linkoping, Sweden
E-mail: petna@lin.foa.se

There are many research papers published in modulation classification,
and most of them have a common framework. In this paper we will give
an overview, and the paper contains four topics: 1) Some fundamental
principles, 2) features used for classification, 3) the algorithm
structure, and finally 4) a literature survey.
Paper

ET.1
RECONSTRUCTION OF STRUCTURE AND TEXTURE OF PLANAR ENVIRONMENTS BY DYNAMIC
VISION TECHNIQUES

M. Cossi, G.M. Cortelazzo,  R. Frezza

Tel. +39 49 8277825; fax: +39 49 8277826
e-mail: frezza@dei.unipd.it

ABSTRACT
This work is concerned with the estimate of   structure and texture  of
buildings     from a  video sequence.  The goal includes the recovery of metric
information.   The results could be   conceivably used for many purposes ranging
from   photogrammetric applications to CAD models that  could be applied, for
example,  for virtual visits of sites of artistic and historical significance.
We present  an original algorithm to estimate  both structure and texture
of environments composed by planes like the interiors of most buildings.
From a video sequence of a decorated wall  the algorithm computes a plane that
approximates the wall (structure estimation) and composes a mosaic of the single images to
reproduce the decoration (texture estimation).  The data are organized
so that it is possible to observe the wall from an arbitrary point of view.
Paper

ET.2
ALGORITHMS AND SYSTEMS FOR MODELING MOVING SCENES

V. Michael Bove, Jr.

Media Laboratory, Massachusetts Institute of Technology
Room E15-324, 20 Ames Street, Cambridge MA 02139 USA
vmb@media.mit.edu, http://www.media.mit.edu/~vmb/

In this paper I describe the application of machine-vision techniques
to video coding in order to create what my research group calls
object-oriented television, where moving scenes are represented in
terms of objects (as recovered by analysis methods).  Beyond data
compactness, such a representation offers the ability to add new
degrees of freedom to content creation and display.  I discuss some of
the scene analysis problems (particularly 2-D and 3-D model-fitting
and object segmentation) and the algorithmic approaches my group has
taken to solve them; suggest computational strategies for compact,
powerful, programmable decoding hardware (particularly stream-based
computing combined with automatic resource management); and
demonstrate some of the applications we have developed.
Paper

ET.3
REGION-BASED IMAGE ANNOTATION USING COLOR AND TEXTURE CUES

Eli Saber and A. Murat Tekalp

Xerox Corporation, 435 W. Commercial St., East Rochester, NY 14445, saber@roch803.mc.xerox.com

We present algorithms for automatic image annotation and retrieval based
on pixel-based color, and block- or region-based texture features.
Region formation has been accomplished by utilizing Gibbs random fields
or morphological based operations. Color, and texture indexing may
be knowledge-based (using appropriate training sets) or by example.
The algorithms are designed to: i) offer the user a wide range of options
and flexibilities in order to enhance the outcome of the search and
retrieval operations, and ii) provide a compromise between accuracy
and computational complexity.
Paper

ET.4
ORIENTATION RADIOGRAMS FOR INDEXING AND IDENTIFICATION IN IMAGE DATABASES

S. Michel (1), B. Karoubi (2), J. Bigun (1) and S. Corsini (3)

(1) Signal Processing Laboratory, Swiss Federal Institute of
Technology,CH-1015 Lausanne, Switzerland.
(2) CREATIS,Research Center Associated to CNRS (#1216) and Affiliated
to INSERM, Lyon, France.
(3) Bibliotheque Cantonale et Universitaire Lausanne, CH-1015 Lausanne/Dorigny,
Switzerland.

mch@es1.siemens.ch
karoubi@creatis.insa-lyon.fr
joseph.bigun@epfl.ch

Archival of images in databases, enabling further study with respect to
their contents, is at our focus of attention. The major difficulties are
i) the processing of a large number of images, ii) that the steadily growing
number of images increase the complexity of the pattern recognition problems
to be solved.  We propose orientation radiograms, to be used as image
signatures for shape based queries. These are the projections of a set
of orientation decomposed images (here 6) to axes whose directions change
synchronously with the orientation bands at hand.  The peaks in the radiograms
represent long edges or lines which are important  for the human when
he recognizes or compares images.  We present the results of experiments
based on approximately 400 images in an application concerning typographic
ornament images. Also is presented a comparative study comprising classical
moment invariants.
Paper

ET.6

DIGITAL WATERMARKS FOR AUDIO SIGNALS

Laurence Boney

Departement Signal
ENST
Paris, France  75634
email: boney@email.enst.fr

Ahmed H. Tewfik and Khaled N. Hamdy

Department of Electrical Engineering
University of Minnesota
Minneapolis, MN  55455
email: tewfik@ee.umn.edu, khamdy@ee.umn.edu

In this paper, we present a novel technique for embedding digital
watermarks'' into digital audio signals.  Watermarking is a
technique used to label digital media by hiding copyright or other
information into the underlying data.  The watermark must be
imperceptible and should be robust to attacks and other types of
distortion.  In addition, the watermark also should be undetectable by
all users except the author of the piece.  In our method, the
watermark is generated by filtering a PN-sequence with a filter that
approximates the frequency masking characteristics of the human
auditory system (HAS).  It is then weighted in the time domain to
account for temporal masking.  We discuss the detection of the
watermark and assess the robustness of our watermarking approach to
attacks and various signal manipulations.
Paper

ET.7

EMBEDDING PARAMETRIC DIGITAL SIGNATURES IN IMAGES

Adrian G. Bors and Ioannis Pitas

Department of Informatics, University of Thessaloniki,
Thessaloniki 540 06, Greece,

A new approach to digital image signatures (watermarks) is proposed in this study.
An image signature algorithm consists of two stages~: signature casting
and signature detection. In the first stage,
small changes are embedded in the image which afterwards are identified in
the second stage. After chosing certain pixel blocks
from the image, a constraint is embedded among their Discrete Cosine
Transform (DCT) coefficients. Two different embedding rules are
proposed. The first one employs a linear type constraint among the
selected DCT coefficients and the second assigns circular detection
regions, similar to the vector quantization techniques. The resistance
of the digital signature to JPEG compression and to filtering are
analyzed.
Paper

ET.8
A NEW SPEECH SCRAMBLING METHOD:
COMPARATIVE ANALYSIS AND A FAST ALGORITHM

V. D. Delic, V. Senk, and V. S. Milosevic

University of Novi Sad, Faculty of Technical Sciences,
Tel: (381 21) 350 244; fax: (381 21) 59 449
e-mail: tlk_delic@uns.ns.ac.yu

ABSTRACT: Conventional speech scrambling concept is based on permutation of time
segments and/or frequency subbands. Although this approach is regarded as an
insecure speech encryption method, almost all published scramblers are of
that type. We found out that a linear combination based on Hadamard matrices
instead of conventional permutation gives better cryptographic performances,
maintaining all the good features of the scrambling concept. The new scrambling
method provides a large keyspace and a simpler key selection. It attains
negligible residual intelligibility and higher degree of cryptanalytic
immunity. The price of these great improvements is a potential complexity
increase. That is why we designed a fast algorithm for the new scrambling
method.

Paper

FI.1
LINEAR FILTERING AND IRREGULAR SAMPLING

R.J.Martin

GEC Hirst Research Centre, Elstree Way, Borehamwood, Herts WD6 1RX, UK
R.Martin@hirst.gmmt.gecm.com

We show how to suppress coloured noise by subtraction (rather than convolution).
The method generalises to nonuniform sampling. It can also be used for identifying
narrow-band signals in noisy backgrounds.
Paper

FI.2
MULTIRESOLUTION ANALYSIS USING ORTHOGONAL POLYNOMIAL APPROXIMATION

Rupendra Kumar and
Pradip Sircar (Corresponding author. email: sircar@iitk.ernet.in)

Department of Electrical Engineering
Indian Institute of Technology Kanpur
KANPUR 208 016, INDIA

Multiresolution decomposition of signals has been conventionally carried
out by the wavelet representation. In this paper, the orthogonal polynomial
approximation has been employed for multiresolution analysis. It is demonstrated
that the proposed technique based on polynomial approximation has certain
distinct advantages over the conventional method employing wavelet representation.
Paper

FI.3
PERFORMANCE EVALUATION OF D-ALPHA FILTERS

M. TABIZA, PH. BOLON
LAMII/CESALP, Universit de Savoie
B.P. 806 - F.74016 Annecy Cedex, France
(CNRS G1047 Information-Signal-Image)
e-mail: bolon@univ-savoie.fr; tabiza@esia.univ-savoie.fr

We study the output variance of a class of nonlinear filters, called da-filters.
In general, it is impossible to obtain an explicit expression of the output
variance because of the implicit Input/Output relationship, except for
a=1 (median filter), a=2 (mean filter) and a= (midrange filter). In this
paper, we develop a new approach to the computation of the filter output
variance. It is based on a linearisation of the filter output about the
order statistics expected values. This approximation is valid for a >
1. It allows optimal a-values to be computed. Experimental results are
presented. They are compared to those of L-filters and with theoretical
lower bounds (Bhattacharyya system of lower bounds).
Paper

FI.4
NONLINEAR DYNAMICS OF BANDPASS SIGMA-DELTA MODULATION

Orla Feely and David Fitzgerald

Department of Electronic and Electrical Engineering
University College Dublin
Dublin 4, Ireland
tel:  +353-1-706 1852
fax:  +353-1-283 0921
e-mail:  Orla.Feely@ucd.ie

ABSTRACT
Much research attention in recent years has been
focussed on the subject of oversampled analogue-to-
digital and digital-to-analogue conversion, based on the
principle of sigma-delta modulation.  Theoretical
analysis of these conversion methods has been
complicated by their nonlinear nature, precluding the
application of standard linear circuit analysis methods.
In recent years a number of researchers have undertaken
a study of sigma-delta modulation based on nonlinear
methods.  This paper summarises the results that have
been obtained by this study in the case of bandpass
sigma-delta modulation, and shows how these results
can be extended to handle certain circuit nonidealities.
Paper

FI.5
ELIMINATION OF LIMIT CYCLES IN A DIRECT FORM DELTA OPERATOR FILTER

Juha Kauraniemi
Timo I. Laakso

Laboratory of Signal Processing and Computer Technology
Helsinki University of Technology
Otakaari 5 A
FIN-02150 Espoo
Finland
Email: Juha.Kauraniemi@hut.fi
School of Electronic and Manufactoring System Engineering
University of Westminster
115 New Cavendish Street
London W1M 8JS
United Kingdom
Email: laaksot@cmsa.westminster.ac.uk

Delta operator realizations have been found to be robust against roundoff errors
when high sampling rate relative to signal bandwidth is used. In this paper zero
input limit cycles in the transposed direct form delta operator structure are
studied. It is shown that the limit cycles of the basic delta structure are much
lower in amplitude than those of the direct form delay structure for narrowband
lowpass filters. Moreover, by certain modifications to the delta operator the zero
input limit cycles can be completely avoided. It is also shown that narrowband
lowpass filters with both low roundoff noise and absence of limit cycles can be
implemented.
Paper

FII.1
ILL-CONDITIONING OF NON-MINIMUM PHASE SYSTEMS

S. Hashemi & J. K. Hammond

Institute of Sound and Vibration Research (ISVR),
University of Southampton

ABSTRACT
The typical inverse problem is the recovery of the input, x, given data,
y and the knowledge of the system A.  Such problems occur frequently in
instrumental science.  For the Linear Time Invariant (LTI) systems the
governing equation can be expressed in matrix form, y=Ax.
In this paper the problem of ill-conditioning of non-minimum phase systems
and the relation of the phase structure of the system to the singular
values of its system matrix is discussed.
Paper

FII.2
FLEXIBLE NONUNIFORM FILTER BANKS USING
ALLPASS TRANSFORMATION OF MULTIPLE ORDER

M. Kappelan, B. Strauss, P. Vary

Institute of Communication Systems and Data Processing (IND)
RWTH Aachen, University of Technology
D-52056 Aachen, Germany
Tel: +49 (0)241 80 6959;     Fax: +49 (0)241 8888 186
e-mail: kiwi@ind.rwth-aachen.de

This paper deals with allpass frequency transformations of uniform filter
banks to achieve nonuniform bandwidths. The known transformation with an
allpass of first order is extended to an allpass transformation of order K.
Thus the flexibility of the filter bank design can be increased significantly.
Paper

FII.4
ELIMINATION OF CLIKS AND BACKGROUND NOISE FROM ARCHIVE
GRAMOPHONE RECORDINGS USING THE "TWO TRACK MONO" APPROACH

Maciej NIEDZWIECKI

Faculty of Electronics
Department of Automatic Control,
Technical University of Gdansk
ul. Narutowicza 11/12,
Gdansk , Poland
Tel: + 48 58 472519;
fax +48 58 415821
e-mail: maciekn@sunrise.pg.gda.pl

Old gramophone recordings are corrupted with a wideband noise (granulation
noise) and impulsive disturbances (cliks, pops, record scratches) - both
caused by aging and/or mishandling of the vinyl material. The paper presents
an improved method of gramophone noise reduction which makes use of two
signals obtained when a mono record is played back using the stereo equipment.
Paper

FII.5
EFFICIENT ALLOCATION OF POWER-OF-TWO TERMS
IN COMPLEX FIR FILTER DESIGN

Tolga Ciloglu* and Yong Hoon Lee**
*Dept. of Electrical and Electronics Eng., Middle East Tech. Univ., Ankara, 06531, Turkey
e-mail: ciltolga@rorqual.cc.metu.edu.tr
**Dept. of Electrical Eng., Korea Advanced Institute of Science and Technology, Taejon, Korea
e-mail: yohlee@eekaist.kaist.ac.kr

Abstract

The design of discrete coefficient FIR filters with
arbitrary magnitude and phase specifictions and
whose coefficients are expressed as the signed
combination of a few power-of-terms (SPT) is
considered. The total number of SPT terms is fixed
and their distribution among the coefficients is not
restricted. The proposed method is an improved
version of those originally proposed for the design
of linear phase filters [9], [10].
Paper

FII.6
CHEBYSHEV DESIGN OF FIR FILTERS WITH ARBITRARY MAGNITUDE AND PHASE RESPONSES

Mathias Lang

INTHFT, Vienna University of Technology
Gusshausstrasse 25/389, A-1040 Vienna, Austria
Tel: +43 1 58801 3527; fax: +43 1 587 05 83
e-mail: mlang@neptun.nt.tuwien.ac.at

This paper presents a method for the design of nonlinear phase FIR digital
filters with complex or real-valued coefficients using the Chebyshev
error criterion. Three different problems are considered: Complex Chebyshev
approximation with additional weighting of the resulting magnitude and phase
errors, simultaneous Chebyshev approximation of a given magnitude and phase
response, and simultaneous Chebyshev approximation of a given magnitude and
group delay response. A linearization approach leads to a problem formulation
that allows the use of stable algorithms with guaranteed convergence. It is
shown that for this linear approach the simultaneous Chebyshev approximation
of a desired magnitude and phase response is a special case of complex
Chebyshev approximation with independent weighting of the magnitude and phase
errors. Two existing design methods are included in this method as special
cases.
Paper

FII.7
DESIGNING  OF  ROBUST  STABLE  DIGITAL  FILTERS

Mariusz Ziolko
Institute of Electronics AGH
ul.Czarnowiejska 78, 30-054 Krakow, Poland
Tel: + 48 12 173048; fax: +48 12 332398
e-mail: ziolko@uci.agh.edu.pl

The Ackerman-Barmish method was used to establish a set of stable family
of an Infinite Impulse Response (IIR) digital filters. Next, the optimization
method was used to choose a filter which meets design specifications given
in the frequency domain. Designing of lowpass third order IIR filter is
presented as an example.
Paper

FII.9
CEPSTRAL SYNTHESIS OF MINIMUM-PHASE FIR AND IIR DIGITAL FILTERS

P. Nagel

Department of Electrical Engineering, University of Kaiserslautern

A new technique for designing causal and minimum-phase FIR and IIR digital
filters is presented. Here, the deviation from a desired quefrency response
is minimised using the Fletcher-Powell algorithm. As a consequence, this
leads to an optimisation of both log-magnitude response and phase response.
Therefore, the method is of special interest for both equalisers and
allpasses. It works with real parameters which represent the poles and zeros
of the system.
Paper

HOS.1
ARMA MODEL IDENTIFICATION USING HIGHER ORDER STATISTICS AND FISHER
INFORMATION CONCEPTS

Eric LE CARPENTIER and Jean-Luc VUATTOUX

Laboratoire d'Automatique de Nantes, URA C.N.R.S. 823,
Ecole Centrale de Nantes/Universite de Nantes,
1 rue de la Noe, 44072 Nantes cedex 03, France.
Tel: (33) 40 37 16 46.
Fax: (33) 40 37 25 22
e-mail: lecarpentier@lan.ec-nantes.fr

The problem of estimating the parameters of a non causal
ARMA system, driven by an unknown input noise with unknown symmetrical
probability density function (PDF) is addressed. A maximum likelihood
approach is proposed in this paper. The main idea of our approach is
that the assumed PDF of the input noise is the PDF minimizing the Fisher
information among PDFs matching the estimated cumulants of $2nd$ and
$4th$ order. This minimization problemis hard to solve, so we use an
over-parameterized PDF model, which is a gaussian mixture. We obtain
two different models for the classes of sub-Gaussian and super-Gaussian
PDFs. For this latter class, we get the most robust estimator in Huber's
sense, among these generated by this class. A new parameter estimation
method is given and its robustness and optimality properties are detailed.
The performances of the resulting identification scheme are compared to
those of another higher order method.
Paper

HOS.2
ARMA Parameter Estimation Through Enhanced Double
MA Modelling

Achilleas G. Stogioglou and Stephen McLaughlin
Signals and Systems Group, Department of Electrical Engineering,
The University of Edinburgh

ABSTRACT

This paper considers the application of MA cumu-
lant enhancement to the identification of the para-
meters of a causal nonminimum phase ARMA(p, q)
system which is excited by an unobservable inde-
pendent identically distributed (IID) non-Gaussian
process. The method proposed in this paper is
based on the double MA method of [1]. The cumu-
lant enhancement is used to improve the cumulants
of the two intermediate MA models which result
from the decomposition of the original ARMA(p, q)
model. Simulation results are presented to demon-
strate the effects of cumulant enhancement on the
estimated ARMA parameters.
Paper

HOS.3
DETECTION AND CLASSIFICATION OF NOISY AR AND ARMA PROCESSES

Jean-Yves TOURNERET, Karine VAREILLE and Martial COULON

ENSEEIHT/GAPSE, National Polytechnics Institute of Toulouse
2 rue Camichel, 31071 Toulouse, France
email: tournere@len7.enseeiht.fr

The paper focuses on the detection and the classification of noisy AR and
ARMA processes. These two kinds of processes cannot be distinguished by
means of their second-order statistics, since they are Spectrally Equivalent
(SE). Higher-order statistics are shown to be an efficient tool for their
detection. A Neyman-Pearson (NP) test, based on these higher-order
statistics, is then studied. The performance of the NP test provides a
reference for comparing suboptimal detector performances.

Paper

HOS.4
HIGHER ORDER DETECTION TEST FOR DETERMINISTIC SIGNALS

Claire Chichereau, Bruno Flament, Roland Blanpain
LETI (CEA-Technologies Avancees) DSYS
CEA - Grenoble - 17, rue des Martyrs
38054 Grenoble Cedex 9 - France
Tel: +33 76 88 95 42; fax +33 76 88 51 59
e-mail: chichereau@dsys.ceng.cea.fr

In the contex of electromagnetic signals, we want to detect a transient
in a non stationnary gaussian noise by a higher order statistic test.
In this paper, we use a new formalism (an extension of Gardner's work)
that enables us to evaluate theoretically the response of higher order
statistic test for detection. We develop the theoretical ground and we
prove that higher order statistic detection test provides a very short
delay detection. We apply our methods  to simulation of a simple and typical
example : the kurtosis.
Paper

HOS.5
DETERMINING THE FALSE-ALARM PERFORMANCE OF HOS-BASED QUADRATIC PHASE
COUPLING DETECTORS

J W A Fackrell and S McLaughlin

Department of Electrical Engineering, University of Edinburgh, UK
jwaf@ee.ed.ac.uk

Quadratic Phase Coupling (QPC) can be detected using Higher Order
Statistics (HOS) measures.  Previously, the bispectrum, biphase and
bicoherence have been used as components in two QPC-detection algorithms.
In this paper it is shown that the expressions which describe these
detectors reduce to the same form for the white Gaussian noise case.
The performance of these detectors is discussed, and particular
attention is given to false alarms, which occur when QPC is detected
in signals which do not exhibit QPC.  A simple expression is derived
which gives the probability of false alarm (PFA) for QPC detectors.
This expression shows how the PFA increases as the Signal to
Noise Ratio decreases, a relationship which is also observed in a
simulation example.
Paper

HOS.6
LINEAR TIME-VARIANT PROCESSING OF
HIGHER- ORDER ALMO ST-PERIODICALLY
CORRELATED TIME-SERIES

Luciano Izzo	Antonio Napolitano

Universita di Napoli Federico II, Dipartimento di Ingegneria Elettronica
via Claudio 21, I-80125 Napoli, Italy; Tel: +39-81-7683156; Fax: +39-81-7683149

The characterization and linear time-variant processing
of the higher-order almost-periodically correlated time-
series in the fraction-of-time probability framework are
considered. At first, the characterization in the tem-
poral domain is presented by exploiting the expression
of the temporal moment function as a sum of complex
sinusoids whose amplitudes and frequencies are contin-
uous functions of the lag vector. Then, the character-
ization in the frequency domain is considered. Finally,
for both random and nonrandom linear systems, the in-
put/output relationships in terms of generalized cyclic
temporal moment functions and generalized cyclic spec-
tral moment functions are stated. As special cases, lin-
ear almost-periodically time-variant systems as well as
systems performing time-scale changing are also treated.
Paper

HOS.7
TITLE : A HIGHER-ORDER CUMULANT BASED DOA ESTIMATION ALGORITHM

PAPER IDENTIFICATION NUMBER : 384

AUTHOR(S) : W.K.Lai and P.C.Ching

AFFILIATION : Department of Electronic Engineering
The Chinese University of Hong Kong, N.T., Hong Kong
Tel: (852) 2609 8266; fax : (852) 2603 5558

E-MAIL : wklai@ee.cuhk.edu.hk

ABSTRACT
Most of the existing direction-of-arrival estimation algorithms depend
on decomposition of the covariance matrix of the system which in turn
require modeling of the contaminating noise. In this paper, a higher-order
cumulant based algorithm for estimating the direction-of-arrival of m
narrowband far field sources impinging on an array with n uniformly spaced
sensors is proposed. Due to the unique property of higher order cumulant,
the proposed method is shown to be at least theoretically independent
of the additive Gaussian noise. The algorithm first evaluates the 2r-th
order cumulant from the output of the system. By making use of these output
cumulants, we obtain a new vector in which its elements are the coefficients
of an equation whose roots are the DOA of the sources. The validity of
the algorithm is demonstrated by extensive computer simulations.
Paper

HOS.8
TITLE : SOME PROPERTIES AND ALGORITHMS FOR FOURTH ORDER SPECTRAL ANALYSIS OF
COMPLEX SIGNALS

AUTHORS : Cecile HUET and Joel LE ROUX

I3S, University of Nice Sophia Antipolis - CNRS
250 rue Albert Einstein Sophia Antipolis 06560 Valbonne FRANCE
Tel: +33 92.94.26.82; fax: +33 92.94.28.96
e-mail: huet@alto.unice.fr - leroux@alto.unice.fr

ABSTRACT
Some algorithms for linear system identification based on fourth order
spectra are given. They extend algorithms developed in the case of
third order statistics. We also give a method for phase unwrapping for
fourth order spectra and we establish a link between algorithms based
on kurtosis maximization and identification method in the frequency domain.
Keywords : higher order statistics (HOS) - higher order spectra - blind identification
- kurtosis maximization - phase unwrapping.
Paper

HOS.9
STATIONARY MOMENTS OF A POLYNOMIAL PHASE SIGNAL,
APPLICATION TO PARAMETER ESTIMATION

A. Ferrari, C. Theys and G. Alengrin

I3S Universit\'e de Nice-Sophia Antipolis
41, Bd Napol\'eon III - 06041 NICE cedex - FRANCE
e-mail : ferrari@unice.fr

This communication addresses the problem of estimating the parameters
of a polynomial phase signal using an original approach: although this
signal is clearly non stationary, some of its high order moments are
shift invariant. The condition verified by the delays of these
stationary'' moments is derived in the noiseless and noisy case.  It
is demonstrate that the only identifiable phase parameter is the
highest order coefficient, the estimation requiring moments of order
at least the double of the phase degree.  An algorithm relying on
these high order moments is derived and its performances are presented
and compared to a recent algorithm.
Paper

HOS.10

DETERMINISTIC ESTIMATION OF THE BISPECTRUM AND
ITS APPLICATION TO IMAGE RESTORATION

Moon Gi Kang$and Aggelos K. Katsaggelos$$Dept. of ECE, University of Minnesota, Duluth, MN 55812, USA email: mkang@d.umn.edu$$ Dept. of EECS, Northwestern University, Evanston, IL 60208, USA email: aggk@eecs.nwu.edu While the bispectrum has desirable properties in itself and therefore has a lot of potential to be applied to image restoration, few real-world application results have appeared in the literature. The major problem with this is the difficulty in realizing the expectation operator, due to the lack of realizations. In this paper, the true bispectrum is defined as the expectation of the sample bispectrum, which is the Fourier representation of the triple correlation given one realization. The characteristics of sample bispectrum are analyzed and a way to obtain an estimate of the true bispectrum without stochastic expectation, using the generalized theory of weighted regularization is shown. Paper I.1 A REAL TIME PROCESSING TCELP CODER/DECODER AT 4.8 KBIT/SEC USING LSP SPLIT VECTOR QUANTIZATION. N. NAJA*, A. GOALIC#, A. EL KESRI*, J.M. BOUCHER# and S. SAOUDI# * Laboratoire d'Electronique et Communications, E.M.I., B.P. 765, Rabat-Agdal, MAROC # Dpartement Signal et Communication ENST-Br., B.P. 832 - 29285 Brest - FRANCE e-mail : andre.goalic@enst-bretagne.fr ABSTRACT Before transmission in a narrow band channel, the speech signal has to be compressed. The Code Excited Linear Prediction Coder (CELP) makes it possible to synthesize good quality speech at low bit rates. Different speech linear predictive coding parameters can be used to design the speech spectral envelope. In our coder the Line Spectrum Pairs (LSP), belonging to the frequency domain, enable us to design the vocal track transfer function. In its first version the coder uses a CLSP (LSP cosine function) scalar quantization leading to a rate of 5.45 kbit/sec. In order to reduce the bit rate to a standard (4.8 kbit/sec), vector quantization was introduced in our coder. Paper I.2 SGS-THOMSON DSP D950-CORE AUDIO COMPRESSION A.M. Alvarez, S. George, Yang H. and K. Das DSP R&D Centre - SGS-Thomson Microelectronics 28 Ang Mo Kio Industrial Park 2 - Singapore 569508 Tel: +65 4805726; Fax: +65 4820240; E-mail: antonio.alvarez@st.com ABSTRACT Dolby Laboratories provides a detailed flow of operation of the Composite multichannel audio compression algorithm, from receiving a frame in the input data buffer all the way to outputting PCM audio signals, in the form of a C-like floating-point pseudocode. This pseudocode helps the licensees in understanding the details in implementing a real-time AC-3 decoder. A set of bitstreams or test vectors are designed to exercise different aspects of the AC-3 algorithm and are used for design verification. This paper describes how these tools were used in order to implement the AC-3 decoder algorithm on a DSP-core.  Paper I.3 THE IMPLEMENTATION OF A HIGH-SPEED DATA ACQUISITION SYSTEM FOR A DSP-BASED INSTRUMENT OPERATING IN REAL-TIME Giovanni Bucci, Pierluigi D'Innocenzo and Carmine Landi Dipartimento di Ingegneria Elettrica, Universit di L'Aquila, Localit Monteluco, 67040 L'Aquila - ITALY tel. (39) 862 434434, fax (39) 862 434403, e-mail: bucci@ing.univaq.it, dinnoc@ing.univaq.it ABSTRACT This paper deals with the hardware and software design and implementation of a modular DSP based instrument, which allows real-time measurements to be carried out. It is based on a TMS320C40 DSP, and on a modular high-speed data acquisition system, described in detail. Experimental results showing the system performance are also included in the paper. Paper I.4 A DATAFLOW LANGUAGE FOR SIGNAL PROCESSING MODELING WITH PARALLEL IMPLEMENTATION ISSUES Guilhem de WAILLY, Fernand BOERI Laboratoire d'Informatique, Signaux et Systemes - URA 1376 du CNRS et de l'Universite de Nice-Sophia Antipolis - {gdw|boeri}.unice.fr This paper describes LambdaFlow, a new functional synchronous dataflow language for DSP applications. It is independent of the handled data. It plainly supports the modular design. Its sound semantics allows proofs of programs and time/memory determinisms. The target code is dynamically loaded into the compiler with a target description that is defined with less than twenty lines of definitions. Due to the static feature of the solving model, it is possible to implement programs onto a static parallel architecture. Paper I.5 MOTION COMPENSATED DE-INTERLACER Daniele Bagni PierLuigi lo Muzio Paola Carrai Vittorio Riva Stefano Vedani Philips S.p.A, Philips Research Monza Via Philips, 12, 20252 Monza (MI), Italy e-mail: bagni@monza.research.philips.com Tel. +39.39.203.7804 e-mail: lomuzio@monza.research.philips.com Tel. +39.39.203.7809 Abstract This paper describes a real time hardware prototype for the de-interlacing of Standard Definition video signals. The line rate is doubled in order to remove specific artefacts of interlaced signals, like interline flicker and line crawl. The hardware applies Digital Signal Processing (DSP) in order to achieve a high performance sequential scan conversion of interlaced signals. The programmability of the prototype gives the possibility to use it as a basis for evaluations of real time line-rate conversion algorithms for both 50 Hz or 60 Hz video sequences. Paper I.6 MULTISCALE EDGES DETECTION ALGORITHM IMPLEMENTATION USING FPGA DEVICES M.Paindavoine , Sarifuddin , C.Milan, JC.Grapin LIESIB- Universit de Bourgogne - 6 bd Gabriel 21000 Dijon email: paindav@satie.u-bourgogne.fr Abstract: One of the way to extract edges uses the fast wavelet transform algorithm. This technique allows the detection of multiscale edge and is used to detect all the details which are in a picture by modifying the scale. The real time application for edge detection involves the implementation of the algorithm on an integrated circuit like an FPGA and the development of an appropriated board. This article deals about the implementation of a wavelet transform algorithm onto a FPGA and the development of an electronic board to detect multiscale edges. Paper I.7 DESIGN AND IMPLEMENTATION OF A DIGITAL DOWN CONVERTER CHIP Ulf Sjostrom, Magnus Carlsson* and Magnus Horlin* National Defence Research Establishment, P.O. Box 1165, S-58111 Linkoping, Sweden email: ulfs@lin.foa.se * Department of Electrical Engineering, Linkoping University, S-58183 Linkoping, Sweden email: {magnusc, magnusch}@isy.liu.se The design and implementation of a CMOS ASIC containing a digital down converter and a channel equalizer for a digital array antenna system is presented. The chip performs nearly 342x10^6 MAC/s (multiply-accumulate/second) at an internal bit-rate of 51.6MHz. The circuit is based on a highly flexible architecture with few full-custom bit-serial arithmetic units. Paper I.8 MORPHOLOGICAL STRUCTURING ELEMENT DECOMPOSITION: IMPLEMENTATION AND COMPARISON M. Razaz, D.M.P. Hagyard, P. Atkin. mr@sys.uea.ac.uk, dmph@sys.uea.ac.uk, phil.atkin@synoptics.co.uk School of Information Systems, University of East Anglia, Norwich, NR4 7TJ, UK. Synoptics Ltd., 271 Cambridge Science Park, Cambridge, UK. Structuring element decomposition is used to reduce computation time in performing morphological image processing operations by breaking down a structuring element into simpler components. This paper classifies decomposition algorithms into two broad categories, namely morphological combination and and set theoretic combination classes. Two important structuring element decomposition methods, the tree search and arbitrary shape decomposition algorithms, are discussed and their performances are compared using a series of different structuring element shapes. We found that the tree search decomposition algorithm is restricted to mainly symmetric and convex structuring elements, and its computation time for performing a morphological operation grows exponentially with the size of the element used, whereas the arbitrary shape decomposition algorithm performs the same operation in linear time, and can deal with any structuring element shape. Paper I.9 IMPLEMENTATION OF A EUROPEAN PAGING SYSTEM RECEIVER USING CORDIC ALGORITHM Jarkko Vuori and Jorma Skytta Laboratory of Signal Processing and Computer Technology Helsinki University of Technology Otakaari 5A FIN-02150 ESPOO, Finland Jarkko.Vuori@hut.fi ERMES is a new European paging standard. The data transmission speed is higher than in older systems, e. g. POCSAG, and advanced new features are implemented including intelligent battery sav-ing operation and country roaming. The higher speed is achieved using the more elaborate modulation method 4-PAM/FM which makes the demodulator implementation much harder than in older 2-FSK based paging systems. The objective of this paper is to propose a novel ERMES signal demodulator structure utilising a complex digital phase-locked loop which is implemented using the CORDIC algorithm. Phase-locked loop demodulators have inherently better performance than the normally used discriminator type detectors. Implementation of those phase-locked loop structures using the CORDIC algorithm makes VLSI realizations very feasible. Paper I.10 EVOLUTIONARY DESIGN OF ANALOG FIR FILTERS WITH VARIABLE TIME DELAYS FOR OPTICALLY CONTROLLED MICROWAVE SIGNAL PROCESSORS Andre' Neubauer Department of Communication Engineering Duisburg Gerhard-Mercator-University 47048 Duisburg, Germany Tel: +49 203 379 2842, Fax: +49 203 379 2902 E-mail: neubauer@sent5.uni-duisburg.de This paper presents the application of genetic algorithms to the design of analog FIR filters with variable time delays - specific examples for tunable optically controlled microwave signal processors. Besides the FIR filter coefficients as the standard design parameters, the time delays can additionally be optimized. Because of physical constraints, specific restrictions of the design parameters, however, have to be obeyed. In order to make use of the additional freedom of optimizing the time delays and to observe the design restrictions, non-standard design techniques are needed. To this end, this paper studies the applicability of genetic algorithms to analog FIR filter design. Experimental results and a comparison to a standard design algorithm are given that demonstrate the excellent properties of the proposed evolutionary design technique. Paper IC.1 TIME VARYING WAVELET TRANSFORM FOR IMAGE CODING Yangzhao Xiang , Ruwei Dai Dept. of Intelligent Systems, Institute of Automation, Chinese Academy of Sciences, Beijing 100080, P.R.China E-mail:student@ailab.ia.ac.cn Wavelet with longer support length is used in the smooth area of image to integrate energy effectively, while wavelet with shorter support length is used in the vicinity of edges. These two sets of wavelet transform switch automatically according to the image. The main problem for the design of time varying filter is how to reconstruct exactly the signal during the transition period. An exact reconstruction method between any two sets of the widely used biorthogonal wavelet bases was proposed in this paper. And even more, by feedback of quantized wavelet coefficients, side information is embedded in the coding bit stream. Paper IC.2 LOSSLESS IMAGE COMPRESSION WITH WAVELET TRANSFORM Jianmin Jiang Department of Computer Studies, Loughborough University, United Kingdom Email: j.jiang@lut.ac.uk ABSTRACT: The research work presented in this paper explores new alternatives for lossless image compression where the entropy coding is applied to the wavelet transform coefficients rather than pixels. The advantage of using wavelet transform prior to entropy coding is that the statistical properties of the resulting coefficients can be analysed and exploited before the model is established for arithmetic coding. Experiments show that the proposed algorithm achieves competitive performances to that of JPEG. Paper IC.3 Invariance properties of integral transforms of images Mario Ferraro1, Franco Giulianini2 1 Dipartimento di Fisica Sperimentale, Universita' di Torino, via Giuria 1, Torino, Italy. tel 39-11-6707376, 39-11-6691104 e-mail ferraro@ph.unito.it. 2 Department of Psychology, Northeastern University, Nightingale Hall, 107 Forsythe str. Boston, Mass., 02150 USA. e-mail giulianini@neu.edu ABSTRACT. In this paper previous results on invariance coding are extended in two ways: 1) by proving that there exists a formal relation between the kernel of an integral transform invariant "in the strong sense" and the eigenfunctions of the operator of the transformation 2) by showing that necessary and sufficient conditions for invariance with respect to one-parameter Lie transformation groups can hold for a class of two-parameters transformation groups, and by providing a procedure to compute an integral transform "invariant in the strong sense" with respect to these transformations. Paper IC.4 MATCHED BLOCK TRANSFORM DESIGN TECHNIQUES Hakan Caglar, Sinan Gunturk, Emin Anarim, Bulent Sankur Bogazici University, Department of Electrical and Electronics Engineering, 80815, Bebek, Istanbul-Turkey caglar@busim.ee.boun.edu.tr gunturk@busim.ee.boun.edu.tr anarim@busim.ee.boun.edu.tr sankur@boun.edu.tr In this work, two new design techniques for matched (adaptive) orthogonal block transforms (BT) based partly on Vector Quantization (VQ) are presented. Both techniques start from reference vectors that are adapted to the characteristics of the signal to be coded. Then the corresponding orthogonal block transform is obtained in the first technique via signed permutations of the reference vector, while in the second technique an optimization search in the null space of the reference vector is executed. The resulting transforms represent a signal coding tool that stands between a pure VQ scheme on one extreme and signal independent fixed block transformation like DCT on the other. Paper IC.5 PROGRESSIVE IMAGE CODING FOR VISUAL SURVEILLANCE APPLICATIONS BASED ON STATISTICAL MORPHOLOGICAL SKELETON G.L. Foresti, C.S. Regazzoni and A. Teschioni Department of Biophysical and Electronic Engineering (DIBE), University of Genoa Via all'Opera Pia 11A, 16145 Genova, Italy. Phone +39-10-3532-792, Fax +39-10-3532-134 e-mail: forfe@dibe.unige.it ABSTRACT This paper presents a new shape representation method for progressive image coding at very-low bit rate. A real application in a railway surveillance system for unattended level-crossings is considered. First, semantic information, e.g., classification of possible obstacles provided by a recognition subsystem, is sent to a remote control center; then, binary shape information is transmitted, in order to allow the remote operator to validate the alarm situation. Pictorial information can be required as a further step by the operator of the control center. Paper IC.6 ENHANCED INITIALIZATION METHOD FOR LBG CODEBOOK DESIGN ALGORITHM IN VECTOR QUANTIZATION OF IMAGES Kwok-Tung LO and Shing-Mo CHENG Department of Electronic Engineering The Hong Kong Polytechnic University Hung Hum, Kowloon, Hong Kong Email: enktlo, ensmc@polyu.edu.hk ABSTRACT In this paper, a new initialization method is developed for enhancing the LBG codebook design algorithm in image vector quantization. The proposed method first arranges the training set data according to three different characteristics of the training vector, i.e. mean, variance and shape. A sampling method based on the criterion of maximum error reduction is then developed to select the desired number of representative vectors in the sorted training set as the initial codebook for the LBG algorithm. Computer simulations using real images show that the proposed approach outperforms the random guess and the splitting method. With the new approach, a higher quality of boundary preservation and a better local minimum are obtainable through a fewer number of iteration. Paper IC.7 Fractal Coding of Subbands using an Oriented Partition Kamel Belloulata, Atilla Baskurt, Hugues Benoit-Cattin and Rmy Prost CREATIS, Research Unit - CNRS (#C5515), affiliated to INSERM INSA 502, 69621 Villeurbanne cedex, France. e-mail : belloulata @ creatis.insa-lyon.fr ABSTRACT In this paper, we propose a new image coding scheme based on fractal coding of the coefficients of a wavelet transform, in order to take into account the self-similarity observed in each subband. The original image is first decomposed into subbands containing information in different spatial directions and at different scales, using Finite Impulse Response filters. Subbands are encoded using Local Iterated Function Systems (LIFS), with range and domain blocks presenting horizontal or vertical directionalities. Their sizes are defined according to the correlation lengths in each subband. The proposed method is applied on standard test images, distortion vs rate is compared with the algorithm proposed by Jacquin for fractal coding of the whole image. Simulation results indicate that the proposed method has better performance than the pyramidal vector quantization on high frequency subbands. Paper IC.8 LOW COMPLEXITY SYNTHESIS FILTER BANK FOR SUBBAND CODING OG IMAGES Ingil Sundsbo and Tor A. Ramstad Norwegian University of Technology and Science Faculty of Electrical Engineering and Computer Science N-7034 Trondheim, Norway E-mail: Ingil.Sundsbo@fysel.unit.no ABSTRACT Optimizations are performed to obtain a filter bank for subband coding of images espescially suited for VLSI implementation. Based on a filter bank consisting of two FIR filters combined with an 8 point DCT, we investigate how the quantization of filter coefficients and twiddle factors in different algorithms affects the quality of the filter bank. It is found that a DCT based on the Stasinski algorithm with twiddle factors of only 5 bits together with FIR filter coefficients of 10 bits, gives a filter bank with high coding gain, no blocking artifacts and limited ringing. The VLSI complexity is comparable to that of DCT transforms. Paper IC.9 PERCEPTUAL QUALITY METRIC FOR DIGITALLY CODED COLOR IMAGES Christian J. van den Branden Lambrecht* and Joyce E. Farrell+ *Signal Processing Laboratory, Swiss Federal Institute of Technology, CH-1015 Lausanne, Switzerland, vdb@lts.de.epfl.ch, http://ltswww.epfl.ch/~vdb/ +Imaging Technology Department, Hewlett-Packard Laboratories, 1501 Page Mill Road MS 1U20, Palo Alto, CA 94304, Farrell@hpl.hp.com In this paper, a computational metric that incorporates many aspects of human vision and color perception to predict the quality of color coded images is presented. The proposed distortion measure is built on opponent-colors theory and on a multi-channel model of spatial vision. The metric has been validated by psychophysical data on 400 images and two human observers. Paper IE.1 Title : BIT-BASED WEIGHTED MEAN FILTER Authors and Affiliations : Barun K. Kar, Mitrajit Chatterjee and Dhiraj K. Pradhan Adv. Design Techology (SPS) Dept.of Computer Science Motorola, 2200 E Elliot Texas A & M Univ. Tempe, AZ-85284 College Stn., TX-77843 email: kar@adtaz.sps.mot.com mitrajit,pradhan@cs.tamu.edu Abstract: Linear-Nonlinear hybrid filters that have appeared in literature suffer from some severe disadvantages. They smear edges and are very hardware intensive. These shortcomings can be overcome by having a Bit-based Weighted Mean filtering scheme. This filter starts by calculating the median of a set of sample values. The sample values are then scaled. Those values which lie in the proximity of the median, are granted more weightage. The weighted sample values are then averaged to yield the filter output. Results show that these filters perform much better than their earlier counterparts with respect to edge preservation and minimizing the minimum absolute error criterion when applied to images corrupted by both impulsive and nonimpulsive noise. These filters are also much more hardware efficient than the L, ATM and M filters. Paper IE.2 A RATIONAL FILTER FOR THE REMOVAL OF BLOCKING ARTIFACTS IN IMAGE SEQUENCES CODED AT LOW BITRATE Roberto CASTAGNO (*) and Giovanni RAMPONI (**) (*) Signal Processing Laboratory Swiss Federal Institute of Technology CH-1015 Lausanne SWITZERLAND E-mail castagno@ltssg4.epfl.ch (**) DEEI, University of Trieste via A. Valerio 10 34127 Trieste ITALY E-mail ramponi@imagets.univ.trieste.it In this paper, a simple but effective operator for the reduction of blocking artifacts is presented. The method is based on the Rational Filter approach: the operator is expressed as a ratio between a linear and a polynomial function of the input data. Such filters proved to outperform other conventional methods in other applications, such as noise smoothing, thanks to their capability of adapting gradually to the local image characteristics. The filter is capable of biasing its behaviour in order to achieve good performance both in uniform areas, where linear smoothing is needed, and in textured zones, where nonlinear and directional filtering is required. A detector of activity is embedded in the expression of the operator itself so that the biasing of the behaviour of the filter is smooth and not based on fixed thresholds. The proposed method has been originally designed as a post--processing tool for frames of sequences coded at medium-low bitrate, but gave good results also when applied to JPEG coded images. Paper IE.3 SPECTRAL ESTIMATION FILTERS FOR NOISE REDUCTION IN X-RAY FLUOROSCOPY IMAGING Til Aach and Dietmar Kunz Philips GmbH Research Laboratories Weisshausstr. 2, D-52066 Aachen, Germany e-mail: aach@pfa.research.philips.com In clinical x-ray fluoroscopy, moving images are acquired at very low x-ray dose so that only 10-500 x-ray quanta contribute to each pixel. The resulting Poisson statistic causes the images to be strongly affected by quantum noise, which, in the observed images, is spatially correlated and signal-dependent. In this contribution, we develop a spatial frequency domain method for intra-frame quantum noise reduction, which takes the non-white noise power spectrum into account. Each image is subjected to a block DFT or DCT. The magnitude of each observed spectral coefficient is compared to the expected noise variance for it, which is derived from a suitable quantum noise model. Depending on this comparison, each coefficient is more or less attenuated, leaving the phase unchanged. Finally, the image is back-transformed and re-assembled. Using this method, noise power reductions of 60% are possible. Paper IE.4 NONLINEAR UNSHARP MASKING FOR THE ENHANCEMENT OF DOCUMENT IMAGES Stefano Chiandussi, Giovanni Ramponi D.E.E.I., University of Trieste via A. Valerio, 10, 34127 Trieste, Italy Tel: +39 40 6767147; fax: +39 40 6763460 e-mail: ramponi@imagets.univ.trieste.it A novel operator for the enhancement of the quality of document images is presented in this paper. This operator, which is a quadratic one, is based on the Unsharp Masking (UM) technique, but it is able to limit noise amplification because every pixel of the processed image depends upon a large portion of the input image; in the same time a good response on details is obtained. A formal description of the operator's response to noise is also presented. Paper IE.5 NEURAL NETWORK APPROACH TO BLIND SEPARATION AND ENHANCEMENT OF IMAGES Andrzej CICHOCKI, Wlodzimierz KASPRZAK, Shun-ichi AMARI RIKEN, Frontier Research Program, BIP Group 2--1 Hirosawa, Wako--shi, Saitama 351--01, JAPAN Phone: +81 48 465 2645; Fax: +81 48 462 4633 e-mail: cia@kamo.riken.go.jp In this contribution we propose a new solution for the problem of blind separation of sources (for one dimensional signals and images) in the case that not only the waveform of sources is unknown, but also their number. For this purpose multi-layer neural networks with associated adaptive learning algorithms are developed. The primary source signals can have any non-Gaussian distribution, i.e. they can be sub-Gaussian and/or super-Gaussian. Computer experiments are presented which demonstrate the validity and high performance of the proposed approach. Paper IE.6 Title: NON CAUSAL ADAPTIVE QUADRATIC FILTERS FOR IMAGE FILTERING AND CONTRAST ENHANCEMENT Authors: S. Guillon, P. Baylou Affiliation: Equipe Signal et Image ENSERB and GDR 134 - CNRS BP 99, 33 402 Talence Cedex, FRANCE Tel: +33 56 84 61 40; fax: +33 56 84 84 06 e-mail: seb@goelette.tsi.u-bordeaux.fr Abstract: In image contrast enhancement, quadratic and more generally polynomial filters are a very popular class of nonlinear filters. These filters exhibit good performances in terms of visual quality, but present some drawbacks such as the elimination of usefull information when using a fixed filter. In this paper we propose a new family of adaptive quadratic filters, where a weighted filter mask is adaptively determined according to the minimization of a prediction error. This filter is then used to enhance locally the image contrast. The results we proposed point out the improvement provided by these new filters in comparison with recent approaches. Paper IE.7 LOCALLY ADAPTIVE TECHNIQUES FOR STACK FILTERING Doina Petrescu, Ioan Tabus, Moncef Gabbouj Tampere University of Technology, Tampere, Finland, e-mail: doina@cs.tut.fi, tabus@cs.tut.fi, moncef@cs.tut.fi This paper introduces a new structure for stack filtering, where the filter adapts to the local characteristics encountered in data. Both supervised and unsupervised techniques for optimal design are investigated. We split the image into small regions and select the stack filter to process each region according to the spatial domain or threshold level domain characteristics of the input signal. This method provides a significant improvement potential over the global stack filtering approach. Some local statistics are computed, to build a reduced input space which efficiently describes the most important local characteristics of data. Vector quantization is used for clustering the reduced input space into a small number of regions, and then finding a mapping between reduced input space clusters and thefilter space, will result in rules for selecting the best suited stack filter for a given region. The supervised clustering procedures are shown to surpass significantly the global filtering approach. Paper IE.8 LMS REGISTRATION OF RIGID TRANSFORMATIONS C Smith , D R Campbell Department of Electrical and Electronic Engineering University of Paisley High Street Paisley PA1 2BE Scotland, UK Email: cameron.smith@paisley.ac.uk Tel: (+44) 141 848 3428 Fax: (+44) 141 848 3404 ABSTRACT Methods are investigated to improve the registration of images corrupted by rigid displacements using the Least Mean Square (LMS) algorithm. Results show that LMS adaptive registration (LMSAR) is effective for small translational displacements, but fails for large translational displacements where the correlation between the rotation data sets is too weak. In an attempt to improve the robustness of LMSAR, various methods are investigated and a modified LMSAR technique is introduced. The modified LMSAR is compared with the Fourier Shift Theorem (FST) for clean and noisy images where the LMSAR accuracy is similar to the FST for clean images. As expected, the LMSAR appears more susceptible to noise, but the LMSAR offers reduced computation over the FST for circumstances involving searches over a large angular range. Paper IE.9 A NEW STABILIZED ZERO - CROSSING REPRESENTATION IN THE WAVELET TRANSFORM DOMAIN AND ITS APPLICATION TO IMAGE PROCESSING Shinji Watanabe, Takashi Komatsu and Takahiro Saito Department of Electrical Engineering, Kanagawa University 3-27-1 Rokkakubashi, Kanagawa - ku, Yokohama, 221, Japan Tel: +81 45 481 5661 Ext. 3119; fax: +81 45 491 7915 E-mail: kurikuri@cc.kanagawa-u.ac.jp, or ,watanabe@saito-lab.eng1.kanagawa-u.ac.jp ABSTRACT We present a new stabilized zero-crossing representation with a salient feature that the signal reconstruction problem reduces to a typical minimum-norm optimization problem, the solution of which is formulated as a linear simultaneous equation, and develop an iterative algorithm for signal reconstruction. Moreover, we extend them to the two-dimensional case. Furthermore, we introduce a threshold operation based on edge intensity to reduce the amount of information in the stabilized zero-crossing representation, and experimentally demonstrate that the threshold operation works well. Paper IR.1 Title: A SET OF MULTIRESOLUTION TEXTURE FEATURES SUITABLE FOR UNSUPERVISED IMAGE SEGMENTATION Authors: Ioannis Matalas, Stephen Roberts and Harry Hatzakis Affiliation: Department of Electrical and Electronic Engineering Imperial College of Science, Technology and Medicine London SW7 2BT, U.K. email: imatal@ic.ac.uk Abstract: We propose a set of multiresolution features for texture description. Image smoothing at multiple scales using the fast smoothing B-spline transform is performed and a number of features, such as the local area, the normal vector dispersion and the gradient orientation, are computed from each scale. A simple disparity function is applied to assess the discriminative power of these features with comparison to other texture methods. Being effective even for small observation windows, the proposed features are suitable for high-resolution texture segmentation. Paper IR.2 STRUCTURAL FAULT DETECTION IN RANDOM MACRO TEXTURES M Mirmehdi, R Marik, M Petrou, J Kittler University of Surrey, Guildford, Surrey GU2 5XH, UK Tel: 01483 259842 Fax: 01483 34139 email: M.Mirmehdi@ee.surrey.ac.uk In this paper, we present a scheme based on iterative morphology for highlighting defects in random textures. The idea is to identify abnormally sized structures in the texture by determining their persistence when iterative morphological erosion is applied. We present some results from a large testbed database of images of granite and ceramic tiles. Paper IR.3 TEXTURE ANALYSIS: COMPARISON OF AUTOCORRELATION-BASED WITH CUMULANT-BASED APPROACHES Vittorio Murino (1), Cinthya Ottonello (2), Sergio Pagnan (3), Andrea Trucco (2) (1) DIMI -University of Udine Via delle Scienze 206, 33100 Udine, Italy (2) DIBE - University of Genoa Via all'Opera Pia 11A, 16145 Genova, Italy (3) IAN- National Research Council of Italy Torre di Francia, Via De Marini 6, 16149 Genova, Italy swan@dimi.uniud.it (Murino) cinthya@dibe.unige.it (Ottonello) fragola@dibe.unige.it (Trucco) ABSTRACT In this paper the use of 3rd-order cumulants, i.e. triple correlations, is proposed for texture analysis. Properties of such features are derived, with particular attention to insensitivity to symmetrically distributed noises and statistical estimate stabilility. Experimental evaluation of 3rd-order cumulants as descriptive features for textures is carried out in comparison with autocorrelation-based approaches. Paper IR.4 OPTIMAL NEURAL NETWORKS COMBINATION FOR HANDWRITTEN CHARACTER RECOGNITION Bernard Gosselin Signal Processing & Circuit Theory Lab, Faculte Polytechnique de Mons Bd Dolez, 31, B-7000 MONS, Belgium Tel: +32 65 37 41 33 - Fax: +32 65 37 41 29 E-mail: gosselin@tcts.fpms.ac.be ABSTRACT: Several methods of combination of Multilayer Perceptrons (MLPs) for handwritten character recognition are presented and discussed. Recognition tests have shown that cooperation of neural networks using different features vectors can reduce significantly the overall misclassification error rate. The final recognition system consists of a cascade association of small MLPs, which allows minimization of the overall recognition time while retaining a high recognition rate. This system appears to be 50% faster than the best of the individual MLPs, while offering a recognition rate of 99.8 % on unconstrained digits extracted from the NIST 3 database. Paper IR.5 FACIAL FEATURE EXTRACTION USING GENETIC ALGORITHMS Bilgin Esme, Bulent Sankur, Emin Anarim Bogazici University, Electrical Engineering Dept., Bebek, 80815, Istanbul { esme, sankur } @boun.edu.tr Face models are used in such applications as videotelephone, graphic animation and automatic answering devices. Extraction and localization of facial features is the first step in constructing and adapting face models. Typical facial features are the eyes, the lips, the chin contour, and the nostrils. In this work, novel deformable templates in combination with genetic algorithms are used to capture eyes and lips contours Paper IR.6 ELIMINATING TARGET SHADOWS FOR IMPROVED TRACKING AND SHAPE ESTIMATION IN OUTDOOR MONOCULAR DIURNAL SEQUENCES Paolo Gamba(*), Massimiliano Lilla, Alessandro Mecocci() (*) Dipartimento di Elettronica, Universita' di Pavia Via Ferrata, 1, 27100 Pavia, ITALY () Facolta' di Ingegneria, Universita' di Siena Via Roma, 77, 55300 Siena, ITALY We present an efficient method able to extract a shadow model from a scene, exploiting the HLS color components. The algorithm allows to recover target shapes in diurnal scene for improved identification. It is based on the realization of a General Bitmap Model and a more particular Strip Bitmap Model to identify shadow regions. Each pixel in the image is classified as shadow or not by a minimum distance approach to these models. Paper IR.7 SENSOR INTEGRATION IN ASSOCIATIVE VISUAL STRUCTURES Fabio Ancona, Giancarlo Parodi, and Rodolfo Zunino DIBE. - Dept. of Biophysical and Electronic Engineering, University of Genova Via all'Opera Pia 11a, 16145 Genova, Italy Tel: +39 10 3532269; Fax: +39 10 3532175 e-mail: {ancona,gian,zunino}@dibe.unige.it ABSTRACT - The paper describes the use of associative models for integrating different sensors. Integrated associative structures are outlined and related to previous approaches; the enhanced robustness resulting from the integration of Associative Memories (AMs) and Neural Networks (NNs) is shown. Discussion then focuses on how different information sources can cooperate on associative visual recognition. Experimental results on real-image testbeds are reported, which confirm theoretical expectations. Paper M.1 MULTICHANNEL TIME-SERIES MODELLING AND PREDICTION BY WAVELET NETWORKS Ales Prochazka (1) and Jonathan Smith (2) (1) Prague University of Chemical Technology, Department of Computing and Control Engineering, Technicka 1905, 166 28 Prague 6, Czech Republic Tel: +42 2 2435 4198; fax: +42 2 2431 1082 E-mail: prochaz@vscht.cz (2) South Bank University, School of Engineering Systems and Design, 103 Borough Road, London SE1 OAA, England, Tel: +44 171 8157666; fax: +44 171 8157699 E-mail: smithjh@sbu.ac.uk Multichannel time-series result from observations of a given engineering, biomedical, econometric or environmental variable taken at different locations. Processing this type of signal presents problems associated with its extrapolation in given space ranges and its possible prediction. This paper presents a comparison of seasonal AR modelling of such signals and the application of wavelet networks to the system identification and prediction of a particular signal. The choice of wavelet functions and the optimization of their coefficients is discussed as well. Each method suggested in the paper is verified for simulated signals at first and then used for the analysis of real signals, including the observation of air pollution. All algorithms are written in the MATLAB environment. Paper M.2 L-INFINITY BLIND DECONVOLUTION FOR THE GENERALIZED AR MODEL Wenyuan Xu , Mostafa Kaveh Department of Electrical Engineering, University of Minnesota,U.S.A. e-mail: kaveh@ee.umn.edu ABSTRACT: Applying the convex cost function L-infinity to the blind deconvolution of general non-minimum phase AR(u) models is studied. A simple and realizable constraint is proposed for the L-infinity deconvolution. With this constraint, except for a gain, the model parameter is the unique solution of the L-infinity deconvolution. The strong consistency of the estimator of the model parameter defined by the sample version of L-infinity norm is presented. An algorithm is suggested for the iterative computation of the estimator. Simulation examples show the proposed approach works well for apprepriate blind equalization problems. Paper M.3 EXTENSION OF AUTOCOVARIANCE COEFFICIENTS SEQUENCE FOR PERIODICALLY CORRELATED RANDOM PROCESSES BY USING THE PARTIAL AUTOCORRELATION FUNCTION Sophie Lambert Laboratoire LMC-IMAG sophie.lambert@imag.fr The extension of stationary process autocorrelation coefficients sequence is a classical problem in the field of spectral estimation. The periodically correlated processes have pratical importance and an interest according to their connection with stationary multivariate processes. That's why we propose a new approach to resolve the previous problem in this context. We use the partial autocorrelation function of this processes class. The extension is so easy to describe. Next, we extend the maximum entropy method to the degenerate case and show that the solution is given by a Periodic Autoregressive process. Furthermore, the connection with the problem of multivariate stationary processes autocorrelation sequence is presented. Paper M.4 SEVERAL SYNTHESIS TECHNIQUES OF FRACTIONAL BROWNIAN MOTION (FBM). O. Magre - M. Guglielmi Laboratoire d'Automatique de Nantes, U.R.A. C.N.R.S. 823 E.C.N./Universite de Nantes 1 rue de la Noe, 44072 NANTES CEDEX, FRANCE Tel:(33) 40 37 16 44/Fax:(33) 40 37 25 22 E-mail magre@lan.ec-nantes.fr This paper deals with the analysis of fractal signals. Our main subject is to compare the fractal characteristics of a new differential model for this kind of signals and the practical synthesis linked to it with the classical synthesis methods linked to the fractional brownien motion (fbm). We use three classical analysis methods (spectrum approximation, wavelets analysis and Higuchi test) using the fractal characteristics, i.e the 1/f spectrum, self-similarity and the length of a fractal curve, long range dependancies. Paper M.5 ALGEBRAIC LATTICE REALIZATION OF PASSIVE TRANSMISSION LINE SYSTEMS Yoshimi Monden, Masayasu Nagamatsu, and Satoru Okamoto Department of Mathematics and Computer Science Interdisciplinary Faculty of Science and Engineering Shimane University Nishikawatsu, Matsue, 690 Japan e-mail: monden@cis.Shimane-u.ac.jp In this paper, firstly, the Schur-Cohn test known as an algebraic stability test of discrete-time linear systems is presented as a lossless bounded realness test by lossless bounded real lattice realization'' of a given real rational transfer function on the unit disk. Then, by characterizing a discrete model of piecewise constant passive transmission line in terms of a set of physical system parmeters, it is extended to an algebraic algorithm for bounded realness test by bounded real realization'' of a certain class of rational transfer functions, which are general enough to cover almost actual passive transmission lines. Paper M.6 MODEL REDUCTION BY KAUTZ FILTERS Author : A.C. den Brinker Affiliation: Eindhoven University of Technology P.O. Box 513 5600 MB Eindhoven The Netherlands Tel: +31 40 2473628 Fax: +31 40 2448375 Email : A.C.d.Brinker@ele.tue.nl ABSTRACT: A method is presented for model reduction. It is based on the representation of the original model in an (exact) Kautz series. The Kautz series is an orthonormal model and is non-unique: it depends on the ordering of the poles. The ordering of the poles can be chosen such that the last sections contribute least or the first sections contribute most to the overall impulse response of the original system (in a quadratic sense). Having a specific ordering, the reduced model order, say n, can be chosen by considering the energy contained in a truncated representation. The resulting reduced order model is obtained simply by truncation of the Kautz series at the n-th term.  Paper M.7 CHARACTERISATION OF THE WIGNER-VILLE DISTRIBUTION OF K-NOISE Miguel A. Rodriguez and Luis Vergara Dpto Comunicaciones, Universidad Politecnica Valencia Camino de Vera s/n, 46071 Valencia, Spain Tel: +34 6 3877300; fax +34 6 3877309 e-mail:mar@dcom.upv.es ABSTRACT In this paper we present a statistical characterisation of the Wigner-Ville transform of k-noise. The results show that the positive and the negative values of the WignerVille transform may be separately considered k-distributed random variables with small distribution parameters. The characterisation has been done analytically, but simulations have shown a great agreement with our theoretical model. Paper M.8 MINIMAL CONTINUOUS STATE-SPACE PARAMETRIZATIONS Alle-Jan van der Veen Delft University of Technology, Dept. Electrical Eng./DIMES, Delft, The Netherlands allejan@cas.et.tudelft.nl Mats Viberg Chalmers University of Technology, Dept. Applied Electr., S-412 96 Goteborg, Sweden viberg@ae.chalmers.se The authors present a minimal continuous parametrization of all multivariate rational contractive transfer functions. In contrast to traditional minimal parametrizations, this parametrization does not contain any structural indices, which makes it very suitable for identification algorithms that use nonlinear optimization to estimate the parameters. Paper M.9 AM-FM EXPANSIONS FOR IMAGES Marios S. Pattichis and Alan C. Bovik Laboratory for Vision Systems, University of Texas, Austin, TX 78712-1084, USA Tel: (512) 471-2887; fax: (512) 471-1225 e-mail: marios@olive.ece.utexas.edu In this paper we present a novel method for computing AM-FM expansions for images. Given an image, we show how to compute an appropriate AM-FM representation. We also describe a general class of functions for which this approach gives the best results. Then, we compute the AM-FM representation on a real-life texture, and show that it has a compact AM-FM spectrum. Paper MDSP.1 Application oriented insights into the Gabor Transform for Acoustic Signals Processing Ewa ukasik Institute of Computing Science, Pozna University of Technology 60-965 Pozna, Piotrowo 3a, Poland, Tel: +48 61 782373; fax: +48 61 771525 e-mail: LUKASIK@POZN1V.TUP.EDU.PL ABSTRACT The paper presents results of analysis of certain quasi stationary and non stationary signals using Gabor transform and Gabor spectrogram. Initial results are based on the original programs realising Gabor transform, whereas the main part of the work - the comparative analysis of signals by Gabor spectrograms of higher orders and other time-frequency distributions was performed using the commercially available software package: Joint Time Frequency Analysis (JTFA) Toolkit from National Instruments. In most cases the experiments showed superiority of Gabor spectrogram over other methods mainly due to better time and frequency resolution and elimination of some of the cross terms inherent e.g. for Wigner-Ville or Choi-Williams Distributions. Paper MDSP.2 USE OF TIME--FREQUENCY REPRESENTATION FOR TIME DELAY ESTIMATION OF NON STATIONARY MULTICOMPONENT SIGNALS M. Matacchione, L. Lo Presti, G. Olmo Dipartimento di Elettronica, Politecnico Corso Duca degli Abruzzi 24 - 10129 Torino - Italy Ph.: +39-11-5644033 - FAX: +39-11-5644099 - E-mail: "olmo(lopresti)@polito.it" In this paper, we propose three methods for the TOA and TDOA estimation, based on the Choi William Distribution (CWD), and suitable for single as well as multicomponent non stationary signals. The CWD exhibits some very interesting properties, which are exploited for the TOA estimation and which are discussed in the paper; moreover, it turns out to be almost insensitive to even large amounts of noise. The proposed methods are validated by means of numerical examples, which point out their effectiveness in terms of mean value and standard deviation of the estimated TDOA's Paper MDSP.3 BINARY-VALUED WAVELET DECOMPOSITIONS OF BINARY IMAGES Mitchell D. Swanson and Ahmed H. Tewfik Department of Electrical Engineering University of Minnesota Minneapolis, MN 55455 USA mswanson@ee.umn.edu, tewfik@ee.umn.edu We introduce a binary-valued wavelet decomposition of binary images. Based on simple modulo-2 operations, the transform is computationally simple and immune to quantization effects. The new transform behaves like its real-valued counterpart. In particular, it yields an output similar to the thresholded output of a real wavelet transform operating on the underlying binary image. Using a new binary field transform to characterize binary filters, binary wavelets are constructed in terms of 2-band perfect reconstruction filter banks. We include lossless image coding results to illustrate the compactness of the representation. Paper MDSP.4 SPATIO-TEMPORAL WAVELET TRANSFORMS FOR IMAGE SEQUENCE ANALYSIS. J.-P. Leduc and C. Labit IRISA / Centre INRIA - Rennes Campus de Beaulieu avenue du General Leclerc, F-35042 Rennes, France Tel : + (33)-99-847425 Fax: + (33)-99-847171 Email: leduc@irisa.fr This paper intends to present an integrated approach of constructing new spatio-temporal wavelets for discrete signal analysis. The main illustrative field of applications considered here stands as the analysis of digital image sequences. Nevertheless, this can be readily extended to any kind of spatio-temporal signals. Continuous wavelet transforms, continuous series, discreriz;~- series and discrete transforms are considered here in an unified way. The analysis to be developed relies only on dynamic parameters like uniform translation and rotation, on kinematic parameters like velocity and speed and on structural parameters as scale and orientation. This digital processing intends to cover the detection and the focalization on motion-based regions of interest in order to perform tracking, classification, segmentation, multiscale trajectory construction and eventually a selective reconstruction of the useful content. Paper MDSP.5 SELECTION OF SAMPLING GRID AND PREFILTER FOR IMAGE DECIMATION BASED ON SPECTRAL EXTENSION ANALYSIS Federico Pedersini, Augusto Sarti, Stefano Tubaro Dipartimento di Elettronica e Informazione - Politecnico di Milano Piazza L. Da Vinci, 32, 20133 Milano, Italy Tel: +39-2-2399.3647, Fax: +39-2-2399.3413 E-mail: pedersin/sarti/tubaro@elet.polimi.it} ABSTRACT Signal decimation aimed at optimal spectral packing has a variety of applications in areas ranging from array processing to image processing. In this article we propose and discuss a new method for determining decimation grid and prefilter that best fit the spectral extension of any 2D signal defined on an arbitrary sampling lattice. The method has been implemented and tested on digital images in order to evaluate quality degradation due to optimal spectral truncation. Paper MDSP.6 DISCRETE MODELS FOR MULTIDIMENSIONAL SYSTEM SIMULATION Rudolf Rabenstein Lehrstuhl fuer Nachrichtentechnik, Universitaet Erlangen-Nuernberg Cauerstrasse 7, D-91058 Erlangen, Germany Tel: +49 9131 858717; fax: +49 9131 303840 e-mail: rabe@nt.e-technik.uni-erlangen.de Abstract: Multidimensional continuous systems arising from physical applications with distributed parameters are conventionally modelled by partial differential equations. This paper presents an alternate description by transfer functions based on suitably chosen functional transformations. Signal processing techniques lead to discrete simulation models which are suitable for computer implementation. Numerical results show considerable savings in computer time over existing numerical methods. Paper MDSP.7 DIRECTIONAL COMPOSITE MORPHOLOGICAL FILTER IN IMAGE PROCESSING Wei LI and Joseph RONSIN INSA, Laboratoire ARTIST, 20 Avenue des Buttes de Cosmes 35043 RENNES Cedex, FRANCE Telephone: (33) 99 28 65 05, Fax: (33) 99 28 64 95, E-mail: Wei.Li@insa-rennes.fr Abstract In this paper, two pairs of dual morphological filters, the composite morphological filters (CMFs) are introduced. They have distinctive properties comparing with other similar morphological operations. CMFs are used to construct a directional morphological filter in impulsive noise removal procedures. It is proven that it has better noise-removal and detail-preserving abilities than classical morphological filters and other non-linear filters such as median or centre weighted median based filters. A threshold scheme is added to improve the final filtering performances. Paper MDSP.8 A NEW TWO-DIMENSIONAL BLOCK LEAST MEAN SQUARES ADAPTIVE ALGORITHM S. Attallah and M. Najim Equipe Signal/Image and GdR-134-CNRS ENSERB. Av. du Dr. Albert Schweitzer BP 99. 33402 Talence Cedex FRANCE e-mail: attallah@goelette.tsi.u-bordeaux.fr ABSTRACT In this paper, a new 2-D block LMS algorithm is presented. This algorithm, which is an exact mathematical formulation of classical 2-D LMS algorithms, presents the advantage of preserving a good convergence as the block size increases. The reduction in the computational complexity is achieved by expoiting the redundancy between successive computations, rather than using disjoint or partially overlapping windows. The latter are known to degrade the convergence when the block size is large. Paper MDSP.9 DESIGN OF 3-D OPTIMAL FIR FILTERS WHICH EXTRACT OBJECTS MOVING ALONG LINEAR TRAJECTORY Katsuya KONDO and Nozomu HAMADA Dept. of Electrical Engineering, Keio University 3-14-1 Hiyoshi, Kohoku-ku, Yokohama 223, Japan Tel: +81 45 5631141(Ext.3360); fax: +81 45 5632773 e-mail: kondo@tkhm.elec.keio.ac.jp We propose a design method of optimal FIR filter which selectively extracts the particular moving object from other moving objects and noise. Stochastic approach is applied to the problem using the information of signals and the probability distribution of velocity vectors. In the method, the frequency response of the proposed Linear Trajectory Filter (LTF) specified by a priori information of the moving object's shape and its velocity vector. In addition, we derive a general formulation of the problem for optimal filter design and its solution for any signal and noise. Through some examples, it is shown that the target object is effectively enhanced in the noisy environment. Paper MDSP.10 A MULTIVARIABLE STEIGLITZ-MCBRIDE METHOD Mehdi Ashari (1), Mamadou Mboup (2) and Phillip A. Regalia (3) 1-Laboratoire des Signaux et Systemes, CNRS-ESE-UPS, 91192 Gif-sur-Yvette, France. e-mail: ashari@lss.supelec.fr 2-Univ. Rene Descartes-Paris V, UFR Math-Info, 45 rue des Saints Peres, 75270 Paris cedex 06, France. e-mail: mboup@math-info.univ-paris5.fr 3-Inst. National des Telecomm., Depart. Signal & Image, 9 rue Charles Fourier, 91011 Evry cedex, France. e-mail: regalia@cosmos.int-evry.fr In this paper, we present an off-line multi-input/multi-output version of the Steiglitz-McBride method, as well as an analytic description of the set of its stationary points. As in the scalar case, the description is given in terms of first- and second-order interpolation constraints, respectively, on the model impulse response and covariance sequences. The constraints are related to the theory of q-Markov covariance equivalent realizations and generalize the work of Inouye and King et al. Paper ME.1 AN EFFICIENT MATCHING APPROACH TO MOTION ANALYSIS OF IMAGES Kostas Girtis*, Theodore Lilas** and Stefanos Kollias** *Department of Informatics, University of Piraeus, Karaoli & Dimitriou 80, 18534 Piraeus, Greece. e-mail: girtis@unipi.gr ** Computer Science Division, National Technical University of Athens, Zografou 15773, Greece. e-mail:stefanos@cs.ntua.gr ABSTRACT Local information in not always enough for efficient motion analysis. Additive processing is required to get accurate results. This has been formulated as the aperture problem. Block matching algorithms are applied between successive images for motion estimation, assuming conservation of local intensity distribution. Matching approaches provide good results when the aperture problem does not exist. However, in regions when the aperture problem exists, additional constraints are required in order to recover the displacement of pixels between consecutive images. In this paper we present a way to improve the performance of optical flow computation at the first early level. Morphological filters are introduced in the matching approach with which we overcome inherent problems of correlation based techniques. Paper ME.2 PARAMETER ESTIMATION OF NON - TRANSLATIONAL MOTION FIELDS Constantinos Dimou Ioannis Pitas Dept. of Informatics, University of Thessaloniki P.O. BOX 451 Thessaloniki, GREECE Tel.: +30-31-996304, FAX: +30-31-996304 e-mail: dinos@zeus.csd.auth.gr pitas@zeus.csd.auth.gr Motion estimation is a very important topic in computer vision and image sequence compression. However, most commonly used motion estimation algorithms do not take into consideration any motion invariances that a certain local motion might possess. In this paper, a technique for estimating the invariant motion parameters of non-translational motion fields is proposed, which leads to more efficient estimation, smoothing or coding of the motion field. It is shown that the algorithm performs well, even in high noise levels, i.e., in the case of noisy output of the motion estimator. Paper ME.3 JOINT MOTION ESTIMATION/SEGMENTATION FOR OBJECT-BASED VIDEO CODING Soo-Chul Han, Lilla Boroczky, and John W. Woods Center for Image Processing Research & ECSE Department Rensselaer Polytechnic Institute Troy, NY 12180-3590 USA sooch@ipl.rpi.edu, lboroczky@vnet.ibm.com, woods@ecse.rpi.edu A video coding scheme is presented in which the coding is performed on individual moving objects. A Markov Random Field model is employed in finding the motion and boundaries of the objects. By guiding the object segmentation process with the spatial color information, meaningful objects representative of the real video scene are extracted. Furthermore, this enables a systematic treatment in handling the covered/uncovered regions, as well as the appearance/disappearance of moving objects. The rate for transmitting object motion and boundary is greatly reduced by use of temporal updating. The interior coding is performed by object-based subband decomposition. Simulations indicate promising results for low bitrate applications. Paper ME.4 A SPIRAL SEARCH ALGORITHM FOR FAST ESTIMATION OF BLOCK MOTION VECTORS Th. Zahariadis and D. Kalivas National Technical University of Athens zahariad@telecom.ntua.gr Develop. Programmes Dept., Intracom S.A. dkal@intranet.gr The most important fast block matching algorithms are analysed and evaluated. Then a new fast search method, the "Spiral Search Algorithm" (SSA), is introduced. It is a three step algorithm which follows a spiral path searching outwards for candidate locations that satisfy the matching criterion. The efficiency of the SSA arises from: (1) the reduction of the candidate locations without leaving out zones of pixels where the mean absolute difference is not evaluated, and (2) the reduction of computations since many candidate locations are being bailed out. A comparison of fast search methods and the Full Search (FS) approach is presented for a number of video sequences. The SSA is proven to be an excellent compromise between quality and speed. Paper ME.5 MOTION CONNECTED OPERATORS FOR IMAGE SEQUENCES Philippe Salembier, Albert Oliveras and Luis Garrido Universitat Politecnica de Catalunya Campus Nord - Modulo D5 C/ Gran Capita, 08034 Barcelona, Spain Tel: (343) 401 74 04 Fax: (343) 401 64 47 E-mail: philippe@gps.tsc.upc.es This paper deals with motion-oriented connected operators. These operators eliminate from an original sequence the components that do not undergo a specific motion (defined as a filtering parameter). As any connected operator, they achieve a simplification of the original image while preserving the contour information of the components that have not be removed. Motion-oriented filtering may have a large number of applications including sequence analysis with motion multi-resolution decomposition or motion estimation. Paper ME.6 Title: EFFECTIVE MOTION FIELD DESCRIPTION BASED ON AFFINE MODELS AND GLOBAL MOTION INFORMATION Authors: Marco Barbieri, Rosa Lancini Affiliation: Signal Processing Laboratory, Swiss Federal Institute of Technology CH-1015 Lausanne, Switzerland Tel: (+39 2) 661 612 40; Fax: (+39 2) 661 004 48 e-mail: barbieri@mailer.cefriel.it CEFRIEL, Politecnico di Milano Via Emanueli 15, I-20216 Milano Tel: (+39 2) 661 612 09; Fax: (+39 2) 661 004 48 e-mail: rosa@mailer.cefriel.it Abstract: In this paper we study the possibility to estimate reliable motion field considering both a global motion (due to camera parameters changes) and local motion (due to the displacement of the image objects). Considering two images I(n) and I(n-k) a first motion field is estimated using a block matching algorithm. From this information, the global motion parameters (horizontal/vertical pan and zoom factor) are estimated and then one of the two images is compensated by the estimated global motion. Then a combination of a block matching and differential algorithm is used to obtain a dense local motion field. Simulation results indicate that the detection and compensation of the global motion are essential for good motion filed estimation and motion compensated prediction. Moreover the local motion field is used as input for a segmentation algorithm based on affine model, in order to detect the moving object present in the scene. Paper ME.7 PAPER TITLE: Multivector Motion Description for Region-based Very Low Bit-rate Video Coding AUTHORS: Luis Salgado, Jose I. Ronda, Jose M. Menendez, Enrique Rendon and Alberto Sanz. AFFILIATION: Grupo de Tratamiento de Imagenes E.T.S. Ingenieros de Telecomunicacion Universidad Politecnica de Madrid E-28040 Madrid, Spain. Contact e-mail: lsa@gti.ssr.upm.es ABSTRACT: In the present paper, a new approach to region motion description and estimation is introduced, which results particularly suitable for segmentation-based coding strategies for very low bit-rate video coding. Region motion is described through a variable number of motion vectors (MV's) applied to specific control points. No information about this control points is required to be transmitted as their determination is based on information available at the decoder. Results show an important net bit-rate saving for QCIF images using this new approach versus the standard translational model. Transmission at rates below 64 kbit/s with very high image quality are achieved. Paper ME.8 MOTION VECTOR OPTIMIZATION OF CONTROL GRID INTERPOLATION AND OVERLAPPED BLOCK MOTION COMPENSATION USING ITERATIVE DYNAMIC PROGRAMMING Michael C. Chen and Alan N. Willson, Jr. Department of Electrical Engineering University of California, Los Angeles Los Angeles, CA 90095-1600, USA e-mail: willson@icsl.ucla.edu Interdependence between motion vectors (MVs), introduced by control grid interpolation (CGI) and overlapped block motion compensation (OBMC) algorithms, is the key to improving temporal prediction performance of conventional block-matching motion compensation schemes. Unfortunately, this dependency makes the problem of finding optimal MVs intractable. While standard schemes that successively optimize each MV are susceptible to severe local minimum problems, we propose a dynamic programming (DP) paradigm, where each horizontal or vertical slice of MVs is jointly determined during an iterative optimization process. To retain reasonably low complexity, our algorithm effectively identifies an initial search region and then chooses a proper search scheme for each MV. In addition, a computationally-efficient multiscale search strategy is employed. The performance of the proposed method is compared with that of the standard optimization techniques, and our experimental results show that the proposed scheme always gives a better rate-distortion performance. Especially for CGI, the PSNR improvements and the percentage of bit-rate savings provided by our algorithm, in some cases, are in excess of 1.0 dB and 20%, respectively. Paper ME.9 MOTION COMPENSATION IN BLOCK DCT CODING BASED ON PERSPECTIVE WARPING L. Capodiferro*, S. Puledda*, G. Jacovitti** * Fondazione Ugo Bordoni c/o ISPT, Viale Europa 190, 00149 Rome, Italy Tel: +39-6-54802132; Fax: +39-6-54804401; email: licia@fub.it ** INFOCOM Dpt., University of Rome "La Sapienza", via Eudossiana 18, 00184 Rome, Italy Tel: +39-6-44585838; Fax: +39-6-4873300; email: gjacov@infocom.ing.uniromal.it In this paper we present a technique for bit-rate reduction in the H263 coder by the introduction of perspective transformation in the advanced prediction option of motion compensation. It is based on the use of the available displacement vectors for estimating the image warping. Since block matching gives not reliable estimates of the warping an adaptive technique discarding inconsistent transformations has been adopted. Paper ME.10 MEAN FIELD APPROXIMATION TO MULTIMODAL MOTION ESTIMATION PROBLEM Thanh Dang Nguyen *, Kalman Fazekas ** * Department of Electrical Engineering 200 Broun Hall Auburn University, AL 36849 USA e-mail: nguyet1@eng.auburn.edu ** Department of Microwave Telecommunications Signal Processing Laboratory Technical University of Budapest Goldman ter 3, Budapest, H-1111 Hungary Tel: +36 1 4631559; Fax: +36 1 4633289 e-mail: t-fazekas@nov.mht.bme.hu ABSTRACT: The 2D Markov Random Field (MRF) model, combined with the Bayesian estimation framework, has proved to be an efficient and reliable computing tool to the optical flow estimation problem. Specifically, we are investigating the multimodal approach, where complementary constraints are imposed on the optical flow model. However, this approach suffers from expensive computational requirements, which is the direct consequence of the large dimensions of the optimization problem. Recently, a deterministic optimization technique, namely the mean field approximation has been proposed, which not only provides satisfactory estimation result, but also reduces the computational cost drastically. Here we apply this new technique to the above mentioned multimodal motion estimation problem. Paper MFI.1 EFFICIENT DESIGN OF LOW DELAY IIR QMF BANKS FOR SPEECH SUBBAND CODING Thomas Kleinmann and Arild Lacroix Institut fuer Angewandte Physik, Johann Wolfgang Goethe-Universitaet, D-60325 Frankfurt am Main, Germany e-mail: kleinmann@iap.uni-frankfurt.de Speech subband coding offers resources for wideband speech processing due to the utilization of masking effects of the human ear. The effectiveness of this method depends on a proper splitting of the signal frequency band. In this paper we propose an efficient design of low delay QMF banks using IIR prototype filters. These filterbanks allow sharp bandsplitting operations with a sufficient amount of subband channels and distinctively low perceptive phase distortions. Paper MFI.2 STATE SPACE BEHAVIOR IN TIME-VARYING BIORTHOGONAL FILTER BANKS Aweke N. Lemma and Ed F. Deprettere Department of Electrical Engineering Delft Universty of Technology Delft, The Netherlands aweke@cas.et.tudelft.nl ed@cas.et.tudelft.nl Using state space representations of biorthogonal filter banks, it is possible to come up with a compact theory for the transition between two time-invariant filter banks. The transition interval depends on the sizes of the common subspaces spanned by the controllability operators of the decomposition filters and by the observability operators of the reconstruction filters. When the respective operators span the same space, the transition can be made arbitrarily short. If it is zero, then the special case of instantaneous transition is reached. Paper MFI.3 SYMMETRIC DELAY FACTORIZATION: A GENERALIZED THEORY FOR PARAUNITARY FILTER BANKS Patrick Rault and Christine Guillemot CCETT dpt RCS/ATI 4 rue du Clos Courtel 35512 Cesson-Sevigne FRANCE Phone: + 33 99 12 44 35 fax: + 33 99 12 40 98 email: prault@ccett.fr or guillemo@ccett.fr The Symmetric Delay Factorization (SDF) is introduced for synthesizing linear phase paraunitary filter banks and is applied successfully for designing Time Varying Filter Banks (TVFB). This paper describes a minimal and complete generalized symmetric delay factorization theory valid for a larger class of paraunitary filter banks. The approach presented here provides a unifying framework for linear phase paraunitary filter banks including linear phase Lapped Orthogonal Transforms (LOT) and for cosine-modulated filter banks, this for an arbitrary number of channels (odd or even). This generalized theory opens new perspectives in the design of time varying filter banks used for image and video compression, especially in the framework of region or object based coding. The generalized symmetric delay factorization relying on lattice structure representations leads also naturally to fast implementation algorithms. Paper MFI.4 A NEW DESIGN METHOD OF LINEAR-PHASE PARAUNITARY FILTER BANKS WITH AN ODD NUMBER OF CHANNELS Shogo MURAMATSU and Hitoshi KIYA Dept. of Elec. & Info. Eng., Tokyo Metropolitan University, e-mail: kiya@eei.metro-u.ac.jp In this work, a new design method of M-channel linear-phase paraunitary filter banks (LPPUFB) is proposed for odd M with a cascade structure. The conventional cascade structure has a problem that one of the filters is restricted to be of length M. In the proposed method, all filters are permitted to be of the same length as each other and longer than M. The significance of our proposed method is verified by showing some design examples. Paper MFI.5 TWO-DIMENSIONAL DIAMOND-SHAPED FILTER BANKS FROM ONE-DIMENSIONAL FILTERS C. W. Kok ECE Dept. University of Wisconsin Madison, 1415 Engineering Drive, Madison, WI 53706 Tel (608)-265-4885 Fax (608)-262-4623 Email ckok@cae.wisc.edu T. Q. Nguyen ECE Dept. University of Wisconsin Madison, 1415 Engineering Drive, Madison, WI 53706 Tel (608)-265-4885 Fax (608)-262-4623 Email nguyen@ece.wisc.edu Nonrectangular transformation is proposed for the design of multidimensional filter banks. The advantage of nonrectangular transformation is the abundance of transformation kernels and their efficient implementations by ladder structures. The design of two-dimensional two-channel filter banks from one-dimensional filters is discussed and design examples are presented. Paper MFII.1 Large The Optimum Approximation in Generalized Time-Frequency Domains and Application to Numerical Simulation of Partial Differential Equations Takuro KIDA Department of Information Processing, Interdisciplinary Graduate School of Science and Engineering, Tokyo Institute of Technology 4259 Nagatsuta, Midori-ku, Yokohama-shi, 227 JAPAN. Tel. 045-924-5481, Fax. 045-921-1156 e-mail kida@ip.titech.ac.jp Extended optimum interpolatory approximation is presented for a certain set of signals having n variables. As the generalized spectrum of a signal, we consider a nu-dimensional vector. These variables can be contained in one of the time domain, the frequency domain or the time-frequency domain. Sometimes, these can be contained in the space-variable domain or in the space-frequency variable domain. To construct the theory across these domains, we assume that the number of variables for a signal and its generalized spectrum are different, in general. Under natural assumption that those generalized spectrums have weighted norms smaller than a given positive number, we prove that the presented approximation has the minimum measure of approximation error among all the linear and the nonlinear approximations using the same generalized sample values. Application to numerical simulation of partial differential equations is considered. In this application, a property for discrete orthogonality associated with the presented approximation plays an essential part. Paper MFII.2 COMPUTATIONALLY EFFICIENT REALIZATION OF MDFT~FILTER~BANKS T. Karp, N. J. Fliege Hamburg University of Technology, Telecommunications Institute, Eissendorfer Str.~40, 21071 Hamburg, Germany, E-mail: karp@tu-harburg.d400.de, fliege@tu-harburg.d400.de A realization of the Modified DFT (MDFT) filter bank introduced in [Fli94,Fli94a,KF95] was proposed in [Fli93b]. The analysis and synthesis filter bank consist each of two DFT polyphase filter banks, one without delay and one delayed by M/2 samples where M represents the number of channels of the MDFT filter bank. In this paper, we will show that the two DFTs can be reduced to a single one for prototypes of the lengths N=rM+1 and N=rM+M/2+1, respectively, by doing some simple combinations with the input signals. Hereby the modulation cost is nearly halved. Paper MFII.3 MMSE FILTER BANKS WITH REDUCED COMPLEXITY T. Karp (1), K. Gosse (2), A. Mertins (3), P. Duhamel (2) (1) Hamburg University of Technology, Telecommunications Institute, D-21071 Hamburg, Germany, karp@tu-harburg.d400.de (2) ENST Paris, Dept. Signal, 46, rue Barrault, F-75634 Paris Cedex 13, France, duhamel@sig.enst.fr (3) Technical Faculty of the Christian-Albrechts-University, Kaiserstr. 2, D-24143 Kiel, Germany, am@techfak.uni-kiel.de This paper focuses on subband coding schemes based on critically decimated paraunitary filter banks. An additional network with matrix A is introduced on the decoder side. We show here how to optimize its coefficients jointly with the quantization steps in order to reduce the reconstruction mean square error (MSE) on the output signal due to quantization and filtering. This optimization is performed under bit-rate constraint. Of course, the resulting overall MMSE filter bank (including A ~=I) does not allow perfect reconstruction (PR), but the signal-to-noise ratio (SNR) is remarkably better than for the PR solutions. The main advantage of such an approach is to preserve the original structure of the filter bank while improving the SNR. Here, we apply the method to modulated filters which can be implemented at very low cost. Paper MFII.4 SIMPLIFIED DESIGN OF LINEAR-PHASE PROTOTYPE FILTERS FOR MODULATED FILTER BANKS Joerg Kliewer University Kiel, Institute for Network and System Theory Kaiserstr.2, D-24143 Kiel, Germany E-Mail: jkl@techfak.uni-kiel.de In this paper a simplified design of linear-phase prototype filters for almost perfect reconstruction modulated filter banks will be presented. It is based on an improved frequency-sampling design where the frequency response of an easily designable Nyquist filter is shaped such that the prototype constraints will be approximately satisfied. This method does not involve any coefficient optimization and results in a computationally more efficient, faster and more stable design process, which is especially well suited for longer filters. Paper MFII.5 A NEW ALGORITHM FOR DESIGNING PROTOTYPE FILTERS FOR M-BAND PSEUDO QMF BANKS Michel Rossi, Jin-Yun Zhang, Willem Steenaart University of Ottawa, Canada Nortel, Ottawa, Canada mrossi@elg.uottawa.ca jinyun@nortel.ca This paper presents a simple and efficient method to design M-band Quadrature Mirror Filter (QMF) banks. This method does not rely on a conventional nonlinear optimization method but rather on an Iterative Least Squares algorithm. The algorithm is rapidly converging, simple to implement and flexible. Its convergence does not depend on the starting point. Moreover iteratively calculated weighting functions can be used to shape the stopband of the prototype filter and the filter bank transfer function, and perform the minimax or the gain constrained least squares approximation. Design examples and a MATLAB program implementing the proposed algorithm are included. Paper MFII.6 SHIFT ERROR IN ITERATED RATIONAL FILTER BANKS Thierry BLU France Telecom --- CNET PAB/STC/SGV 38--40 rue du General Leclerc 92131 Issy-les-Moulineaux, FRANCE tel: (33 1) 45 29 64 42; fax: (33 1) 45 29 52 94 e-mail: blu@issy.cnet.fr For FIR filters, limit functions generated in iterated rational schemes are not invariant under shift operations, unlike what happens in the dyadic case: this feature prevents an analysis iterated rational filter bank (AIRFB) to behave exactly as a discrete wavelet transform, even though an adequate choice of the generating filter makes it possible to minimize its consequences. This paper indicates how to compute the error between an "average" shifted function and these limit functions, an open problem until now. Also connections are pointed out between this shift error and the selectivity of the AIRFB. Paper MFII.7 WEIGHTED LAGRANGIAN INTERPOLATING FIR FILTER Ewa Hermanowicz Faculty of Electronics, Telecommunications and Computer Science Technical University of Gdansk ul. Narutowicza 11/12, 80-952 Gdansk, Poland Tel: +48 58 472578, Fax: +48 58 472870 E-mail: hewa@elka.pg.gda.pl ABSTRACT The aim of this paper is to present an algorithm for the coefficients of a weighted Lagrangian interpolating FIR fiIter. The proposed fiIter is effective in the reduction of amplitude response sidelobes responsible for abasing The novelty of the proposed L-th band interpolating f Iter lies in that it allows for a simultaneous L-fold interpolation and fractional sample delaying of an input signal. The f lter can be recommended for on-line resampling in variable delay situations especially when implemented in the so-called modified Farrow structure. Paper MFII.8 TWO-STAGE POLYPHASE INTERPOLATORS AND DECIMATORS FOR SAMPLE RATE CONVERSIONS WITH PRIME NUMBERS Hkan Johansson and Lars Wanhammar Department of Electrical Engineering, Linkping University S-581 83 Linkping Sweden Tel: +46 13 284421 e-mail: hakanj@isy.liu.se Abstract In this paper we demonstrate that it can be advantageous from a computational point of view to use a two-stage realization instead of a single-stage realization for sample rate conversions with prime numbers. One of the stages performs a conversion by a factor of two using linear-phase, or approximately linear-phase, half-band filters. The other stage changes the sample rate by the rational factor N/2 using a linear-phase FIR filter. The actual filtering can be performed at the lowest of the two sample frequencies involved. It is also possible to exploit the coefficient symmetry of the linear-phase FIR filter in the stage that changes the rate by a rational factor. The overall workload of the two-stage realization can therefore be reduced compared with the corresponding single-stage realization. Paper MFII.9 EQUALIZERS FOR TRANSMULTIPLEXERS IN ORTHOGONAL MULTIPLE CARRIER DATA TRANSMISSION Thomas Wiegand (1) and Norbert J. Fliege (2) (1) Telecommunications Institute University of Erlangen-Nuremberg Cauerstr. 7/NT, 91058 Erlangen, Germany wiegand@nt.e-technik.uni-erlangen.de (2) Telecommunications Institute Hamburg University of Technology Eissendorfer Str. 40, 21071 Hamburg, Germany fliege@tu-harburg.d400.de Orthogonal multiple carrier data transmission systems are efficiently realized using modified DFT transmultiplexer filter banks. In data transmission applications, a non-ideal transmission channel causes distortions such as intersymbol interference and crosstalk between the subrate bands of the transmultiplexer. Hence, in order to equalize these distortions, subband equalizers, which affect the intersymbol interference and crosstalk behavior, are considered for implementation. The special structure of modified DFT transmultiplexers requires a discussion concerning the various possibilities of placing the subband equalizers at the receiver. Wiener solutions and LMS adaptive algorithms for various new subband equalizer structures are derived and compared by means of simulation results. Paper MFII.10 CROSSTALK CANCELLATION AND MEMORY TRUNCATION IN TRANSMULTIPLEXER FILTER BANKS - TRANSMISSION OVER NON-IDEAL CHANNELS Alfred Mertins Technical Faculty of the Christian-Albrechts-University Telecommunications Institute Kaiserstr. 2 D-24143 Kiel, Germany e-mail: am@techfak.uni-kiel.de When transmultiplexers with overlapping frequency bands are used for the transmission of data over non-ideal channels, intersymbol interference and crosstalk between different data channels will arise. This paper addresses the design of optimal linear networks that reduce the effects mentioned above. A receiver structure based on a combination of crosstalk reduction, memory truncation and Viterbi detection will be proposed. The filter design method presented here is based on the maximization of a signal-to-noise ratio (SNR) at the detector input. The SNR will be defined for channel memories being truncated to arbitrary lengths. Thus, low-complexity Viterbi detectors working independently for all data channels can be used. The design of minimum mean squares error (MMSE) equalizer networks is included in the framework. Paper MV.1 REAL-TIME COMPUTATION OF 2-D MOMENTS ON BLOCK REPRESENTED BINARY IMAGES ON THE SCAN LINE ARRAY PROCESSOR Iraklis M. Spiliotis and Basil G. Mertzios Department of Electrical and Computer Engineering Democritus University of Thrace 67100 Xanthi HELLAS Tel: +30 541 79559 Fax: +30 541 26473 e-mail: spiliot@demokritos.cc.duth.gr mertzios@demokritos.cc.duth.gr ABSTRACT This paper presents an algorithm for the real-time computation of 2-D statistical moments on binary images on the Scan Line Array Processor (SLAP). The binary images are represented as sets of nonoverlapping rectangular areas. This representation scheme is called Image Block Representation. The real-time computation of moments in block represented images is achieved by exploiting the rectangular structure of the blocks. The algorithms for image block representation and for the real-time computation of moments are implemented on the Scan Line Array Processor (SLAP). Paper MV.2 SUBPIXEL EDGE LOCALIZATION WITH STATISTICAL ERROR COMPENSATION Federico Pedersini, Augusto Sarti, Stefano Tubaro Dipartimento di Elettronica e Informazione - Politecnico di Milano Piazza L. Da Vinci, 32, 20133 Milano, Italy Tel: +39-2-2399.3647, Fax: +39-2-2399.3413 E-mail: pedersin/sarti/tubaro@elet.polimi.it ABSTRACT Subpixel Edge Localization (EL) techniques are often affected by an error that exhibits a systematic character. When this happens, their performance can be improved through compensation of the systematic portion of the localization error. In this paper we propose and analyze a method for estimating the EL characteristic of subpixel EL techniques through statistical analysis of appropriate test images. The impact of the compensation method on the accuracy of a camera calibration procedure has been proven to be quite significant (44\%), which can be crucial especially in applications of low-cost photogrammetry and 3D reconstruction from multiple views. Paper MV.3 A COMPARISON OF LINEAR AND NONLINEAR SCALE-SPACE FILTERS IN NOISE Richard Harvey, Alison Bosson, J. Andrew Bangham} School of Information Systems, University of East Anglia, Norwich, NR4 7TJ, UK. Tel: +44 1603 593257; fax: +44 1603 593345 e-mail: \{rwh,bosson,ab\}@sys.uea.ac.uk} The properties of two scale-space systems are compared by examining their performance in noise. It is found that in Gaussian noise linear diffusion and a new type of filter called the area sieve have similar performance but in impulsive noise of random amplitude the area sieve is superior. Paper MV.4 A COMPARISON OF CFAR STRATEGIES FOR BLOB DETECTION IN TEXTURED IMAGES Carlos Alberola-Lopez, Jose Ramon Casar-Corredera, Juan Ruiz-Alzola DTSCIT.ETSIT-UVA.C/Real de Burgos s/n. 47011 Valladolid DSSR.ETSIT-UPM. Ciudad Universitaria s/n. 28040 Madrid DSC.EUIT-ULPGC.Campus de Tafira s/n. 35017 Las Palmas de Gran Canaria Tel: +34 83 423262; fax: +34 83 423261 e-mail: carlos@tel.uva.es Traditional CFAR (constant false alarm rate) approaches applied to the detection of objects in images have proved useful in locating small patches on non-stationary backgrounds. However, the topic of detecting arbitrarily large objects by means of these approaches has received less attention. In this paper we make a comparative analysis of the performance of several CFAR strategies applied to the detection and segmentation of blobs in textured images. The difference in the strategies lies in the way the references for the estimation of the parameters of the detector are considered. By treating four detection schemes through MonteCarlo simulation, we show that directional approaches to the target have better results than non-directional ones. The fourth approach, refered to as "gradient-guided", is the most promising philosophy. Paper MV.5 OPTIMIZATION OF SPACEBORNE IMAGING SENSORS WITH AN END TO END SIMULATION SYSTEM Ralf Reulke, Herbert Jahn German Aerospace Research Establishment (DLR) Institute for Space Sensor Technology Rudower Chaussee 5, D-12484 Berlin, Germany Tel.: (+49) 30 69545518, Fax: (+49) 30 69545512, e-mail: Ralf.Reulke@dlr.de, Herbert.Jahn@dlr.de Abstract To optimize a sensor (and a mission), the existing knowl- edge about the scientific problem and about the available technology should be used. That is the objective of the opti- mization concept used here. The optimization concept is based on an error function which compares assumed values with the estimated values of the parameters provided by the experiment. The error on the consideration is a function of some instrument (and mission) parameters which can be optimally chosen by minimizing the error function taking into account the tech- nological (and cost) limitations. For this approach a numerical simulation tool has been developed that allows computer experiments with specific sensor configurations for design and optimization of opto- electronic sensors for specific (well known) applications. Such a concept is important for the development of dedica- ted sensors for well defined remote sensing tasks to obtain the optimal solution of the problem. Paper MV.6 A NON-ITERATIVE APPROACH TO INITIAL REGION ESTIMATION APPLIED TO COLOR IMAGE SEGMENTATION Javier Portillo-Garcia, Carlos Alberola-Lopez* Lorenzo Jose Tardon-Garcia, Juan Ignacio Trueba-Santander SSR-ETSI Telecomunicacion-UPM, Ciudad Universitaria s/n. 28040 Madrid, Spain *TSCIT-ETSI Telecomunicacion. Univ. Valladolid, C/Real de Burgos s/n. 47011 Valladolid, Spain Tel: +34 1 5495700 ext. 206; fax: +34 1 3367350 e-mail: javierp@gtts.ssr.upm.es A non-iterative segmentation approach is developed to generate a fast initial estimation of the layout of the different color textures presented in the original image mainly based on hypothesis testing. Most of the proposed methods have a tremendous computational burden which make them difficult to be implemented in a real-time working processor. We will compare our method with a known iterative clustering algorithm that guides to similar results with much higher computational cost. We present two examples that show similar results and compare the computational cost for each case. Spotty resemblance caused by pixel oriented decision is diminished in both cases by modeling regions as Markov Random Fields. Paper MV.7 ACCURATE 3-D RECONSTRUCTION FROM TRINOCULAR VIEWS THROUGH INTEGRATION OF IMPROVED EDGE-MATCHING AND AREA-MATCHING TECHNIQUES Federico Pedersini, Stefano Tubaro Dipartimento di Elettronica e Informazione (DEI) Politecnico di Milano Piazza L. Da Vinci 32, 20133 Milano, Italy Tel: +39-2-2399-3647, Fax: +39-2-2399-3413 e-mail: pedersin/tubaro@elet.polimi.it ABSTRACT This paper describes a method for obtaining a reliable 3D reconstruction of close-range objects by properly combining edge- and area-based matching techniques. The adopted acquisition system is a set of three calibrated low-cost CCD cameras. By using an accurate camera model and camera calibration, the method is capable of working with any camera setup. The proposed technique has been tested on some real scenes with encouraging results. Some of these experimental results are presented here. Paper MV.8 3D TRACKING OF DEFORMABLE OBJECTS WITH APPLICATIONS TO CODING AND RECOGNITION Juan Ruiz-Alzola, Carlos Alberola-Lopez*, J.R.Casar-Corredera**,Gonzalo de Miguel-Vela** EUIT Telecommunication-ULPGC, 35017 Las Palmas, Spain *ETSI Telecommunication-UVA, 47011 Valladolid, Spain **ETSI Telecommunication-UPM, 28040 Madrid, Spain Tel : +34-28-452862. Fax : +34-28451243 e-mail : jruiz@cibeles.teleco.ulpgc.es ABSTRACT In this contribution we address the problem of motion and structure estimation of objects that fit a deformation model. Our purpose is to provide a suitable input to a recognition system detected at detecting particular shapes and deformation patterns (gestures) of the object. This is accomplished by means of a stereoscopic vision system which first reconstructs 3D tokens -points- from the images; then the tokens are tracked independently in order to obtain an improved estimation of their positions and to keep a correspondence among them in consecutive instants of time. Finally the tokens are matched to an allowed state -shape- of a Finite State Machine which depicts the deformation of the body. Rigid motion is considered to relate the actual tokens positions with the estimated shape. This approach provides with a convenient way to deal with incomplete collections of measurements due to occlusions. Paper MV.9 Determining Hybrid Reflectance Properties and Shape Reconstruction by using Indirect Diffuse Illumination Method *Tae-Eun Kim and *Jong-Soo Choi *, **Department of Electronic Engineering, Chung-Ang University 221 Huksuk-dong, Dongjak-ku, Seoul, 156-756, KOREA E-mail : kte@candy.ee.cau.ac.kr Abstract In this paper, we propose the estimating method of reflectance properties and the recovery of shape by using Normal Sampler and Indirect Diffuse Illumination Method(IDIM). Photometric Stereo Method(PSM) is generally based on the direct illumination. PSM in this paper is modified with indirect diffuse illumination(IDI) and then applied to hybrid reflectance model which consists of two main components; Lambertian and specular reflectance. Under the indirect diffuse illumination, the reflectance properties of natural objects can be determined by using Normal Sampler that has almost all the normal components in the observed direction. The estimated reflectance properties are used to construct reference table. Also, 3-D shape of an object can be recovered from intensity distribution of a pixel and a reference table. In this paper, the reference table is used to recover the 3-D shape of an object and IDI simplifies the limited conditions of reflectance analysis for prior studies without any loss in performance. The proposed method can be applied to various types of surfaces which can be defined by hybrid reflectance. Paper MV.10 THREE-DIMENSIONAL INSPECTION OF PRINTED CIRCUIT BOARDS USING PHASE PROFILOMETRY Luigi Di Stefano and Frank Boland DEIS, University of Bologna, Viale Risorgimento 2, 40136 Bologna, Italy e-mail: ldistefano@deis.unibo.it EEE, University of Dublin, Trinity College, Dublin 2, Ireland, e-mail: fboland@ee.tcd.ie Reconstruction of 3D shape of the solder paste printed on SMT component pads is a major inspection task in the PCB manufacturing process. The paper reports on the use of phase profilometry for this inspection task. In phase profilometry a structured light pattern is projected onto the object and viewed by a camera. Since the imaged pattern is phase-modulated according to the topography of the object, the extraction of phase information from the image enables reconstructing the 3D shape. In this paper two phase-extraction methods, Fourier Transform Profilometry and Signal Domain Profilometry, are compared by means of simulations and experiments. Results show that the Fourier method performs better, yielding neat detection of the elevation with respect to PCB surface associated with solder paste. Paper NLP.2 STEADY-STATE PERFORMANCE ANALYSIS OF THE LMS ADAPTIVE TIME-VARYING SECOND ORDER VOLTERRA FILTER Mounir Sayadi*, Farhat Fnaiech* and Mohamed Najim** *E.S.S.T.T, 5 Av. Taha Hussein 1008, Tunis, Tunisia, Tel (216) 1 392 559, Fax (216)1 391 166 **Equipe signal / image, ENSERB 351 cours de la libration, 33405 Talence cedex, France. Tel (33) 56 84 66 74, Fax (33) 56 84 84 06, email : najim@goelette.tsi.u-bordeaux.fr ABSTRACT: In this paper, the steady-state performance of the Least Mean Square (LMS) adaptive second order Volterra filter, with constant step-size, in a time-varying setting, is analysed. The quantitative evaluation of the steady-state Excess Mean Square Error (EMSE), where the contribution of the gradient misadjustment and the tracking error are well characterized, is established . The optimum step-size for time-varying second order Volterra filter is then given. Thus, we can study the correlation between the Excess MSE and the optimum step-size in one hand and the parameters of the time-varying nonlinear system, in the other hand. Furthermore, the steady-state behavior predicted by the analysis is in good agreement with the experimental results. The adaptive filter was used in a second order Volterra system identification in a non stationary environment. Paper NLP.3 ON BLIND IDENTIFICATION OF QUADRATIC SYSTEMS Panos Koukoulas, Nicholas Kalouptsidis University of Athens. Department of Informatics, Division of Communications and Signal Processing, T.Y.P.A. Buildings, 157 71 Athens, Greece, koukoula@di.uoa.gr, kalou@di.uoa.gr In this paper the blind identification problem of a finite extent quadratic system driven by a sequence of independent and identically distributed random variables is considered. Output cumulants up to the fourth-order are used and solutions are obtained for special cases of quadratic systems. Paper NLP.4 RECURSIVE VOLTERRA FILTERS WITH STABILITY MONITORING Enzo Mumolo, Alberto Carini Dipartimento di Elettrotecnica, Elettronica ed Informatica Universita' di Trieste, Via Valerio 10, 34127 Trieste, Italy Tel/Fax: +39.40.676.3861/3460 e-mail: mumolo@univ.trieste.it Abstract In this paper we describe a sufficient stability condition for a p-th order recursive Volterra filter. Moreover, we show an application of the stability condition to a system identification problem. Paper NLP.5 PARALLEL-CASCADE ADAPTIVE VOLTERRA FILTERS T. M. Panicker*, V. J. Mathews* and G.L. Sicuranza** *Department of Electrical Engineering University of Utah, Salt Lake City, UT 84112, USA, e-mail panicker@ee.utah.edu **DEEI - University of Trieste, Via A. Valerio, 10, 34127 Trieste, Italy, e-mail sicuranza@univ.trieste.it Adaptive truncated Volterra filters using parallel-cascade structures are discussed in this paper. Parallel-cascade realizations implement higher-order Volterra systems as a parallel and multiplicative combination of lower-order Volterra systems. A normalized LMS adaptive filter for parallel-cascade structures is developed and its performance is evaluated through simulation experiments. The experimental results indicate that the normalized LMS parallel-cascade Volterra filter has superior convergence over several competing structures. Paper NLP.6 GENERALIZED ADAPTORS WITH MEMORY FOR NONLINEAR WAVE DIGITAL STRUCTURES Augusto Sarti Dipartimento di Elettronica e Informazione - Politecnico di Milano Piazza L. Da Vinci 32, 20133 Milano, Italy Tel. +39-2-2399.3647, Fax +39-2-2399.3413 sarti@elet.polimi.it Giovanni De Poli Dipartimento di Elettronica e Informatica - Universita di Padova Via Gradenigo 6A, 35131 Padova, Italy Tel. +39-49-827.7631, Fax +39-39-827.7699 Giovanni.DePoli@dei.unipd.it ABSTRACT The problem of modeling a nonlinear resistor in the Wave Digital domain can be seen as that of applying to its nonlinear characteristic the affine transformation that maps Khirchhoff variables into wave variables. When dealing with nonlinear elements with memory, such as nonlinear capacitors and inductors, the above approach cannot be applied, as affine transformations are memoryless. In this paper, a new approach is proposed for modeling nonlinear elements with memory in the wave domain. The method we propose defines a more general class of wave variables and adaptors with memory that, under some conditions, can incorporate the memory'' of a nonlinear circuit and allow us to treat some nonlinear elements with memory as if they were instantaneous. Paper NLP.7 NONLINEAR INTERFERENCE CANCELLATION USING A RADIAL BASIS FUNCTION NETWORK Paul Strauch, Bernard Mulgrew Dept. of Electrical Eng., The University of Edinburgh, Edinburgh EH9 3JL, Scotland, U.K., Tel/Fax: +44 [31] 650 5655 / 650 6554. e-mail: pes@ee.ed.ac.uk Conventional linear filtering techniques cannot suppress interference or noise in the same band as the signal without degrading the signal. However if the corrupting noise arises from a nonlinear low dimensional dynamical system, it is possible to model the noise as a deterministic process rather than a stochastic one. In this paper a combination of linear and nonlinear models are used to separate the linear signal from the nonlinear noise. The normalised gaussian radial basis function (RBF) network is used to model the nonlinear interference. Decimators have been implemented to reduce the computational cost of the RBF network and re-embed the filtered chaos. Paper NLP.8 PERIODICITY RETRIEVAL FROM NONSTATIONARY SIGNALS USING JOINT ORDER STATISTICS Aleksej Makarov Signal Processing Laboratory Swiss Federal Institute of Technology 1015 Lausanne, Switzerland Tel: +41 21 6934623; fax: +41 21 6937600 WWW: http://ltswww.epfl.ch/~makarov e-mail: makarov@ltssg4.epfl.ch Composite signals are nonstationary processes consisting of trend, noise and cyclic components. A cyclic component consists of periodic or almost periodic data. In this paper we present a method based on nonlinear order statistics that evaluates the fundamental period of a cyclic component. This information can be used for decomposition of composite signals. Paper NLP.9 USES OF NONLINEAR MODEL-BASED TIME SERIES ANALYSIS R.J.Martin GEC Hirst Research Centre, Elstree Way, Borehamwood, Herts WD6 1RX, UK R.Martin@hirst.gmmt.gecm.com We shall discuss recent methods of nonlinear time series analysis and discuss irregular sampling and detection of a known chaotic map in noise. We also discuss the analysis of complex periodic vibrations, and present an example using real data from a steam turbine. Paper NLP.10 SIGNAL PROCESSING VIA SYNCHRONIZED CHAOTIC SYSTEMS WITH FEEDBACK CONTROL Alexander K. Kozlov and Vladimir D. Shalfeev Research Institute of Applied Mathematics and Cybernetics, University of Nizhny Novgorod, 10 Ulyanov St., Nizhny Novgorod 603005, Russia e-mail: alex@hale.appl.sci-nnov.ru In this paper we focus on the transmission of information signals via chaotic oscillations. To this end, we consider systems which contain generators with additional control loop and could behave chaotically and which dynamics may be controlled using feedback or directional coupling. Below we discuss three schemes of signal transmission and detection using 1) phase or frequency controlled generators, 2) coupled Chua's circuits with the adaptive parameter control, and 3) directionally coupled generators extracting binary signal from chaos in a presence of noise. New capabilities of conventional control systems for producing and processing chaotic signals are very promising - both for individual use and for implementation in networks. Paper NN.1 FAST SELF-ORGANIZING OF N-DIMENSIONAL TOPOLOGY MAPS Karin Haese, Heinz-Dieter vom Stein University of the Federal Armed Forces Hamburg Signal Processing Holstenhofweg 85 D - 22043 Hamburg Tel.: (040)/65412462 Fax: (040)/6530413 Email: e_haese@unibw-hamburg.de The self-organizing algorithm, proposed in this contribution, is more efficient than the original one, because it starts at a coarse lattice and refines the lattice of the map using spline interpolation at well determined learning steps until a quantization criteria is reached. Therefore the feature map becomes self-growing. The proposed algorithm also includes a hierarchical search for the nearest neighbour. All these enhancements lead to a time complexity of order O(log{N}) of the self-organizing algorithm for a n-dimensional map with N neurons in each dimension. Paper NN.2 BOUNDING THE ERRORS IN CONSTRUCTIVE FUNCTION APPROXIMATION D. Docampo and C.T. Abdallah* ETSIT, Universidad de Vigo, Campus Universitario, Vigo 36207 *EECE Department, University of New Mexico, Albuquerque, NM 87131 USA Tel: +3486 812134; fax: +3486 812116; e-mail: ddocampo@tsc.uvigo.es Abstract In this paper we study the theoretical limits of finite constructive convex approximations of a given function in a Hilbert space using elements taken from a reduced subset. We also investigate the trade-off between the global errorand the partial error during the iterations of the solution. The results obtained constitute a refinement of well established convergence analysis for constructive iterative sequences in Hilbert spaces with applications in projection pursuit regression and neural network training. Paper NN.3 A NOVEL CONSTRUCTIVE NEURAL NETWORK THAT LEARNS TO FIND DISCRIMINANT FUNCTIONS Jose L. Alba and Laura Docio Departamento de Tecnologias de las Comunicaciones Universidad de Vigo, Spain Phone:34-86-812126, Fax:34-86-812116 e-mail:jalba@tsc.uvigo.es , ldocio@tsc.uvigo.es} This paper presents a novel architecture based on a constructive algorithm that allows the network to grow attending to both supervised and unsupervised criteria. The main goal is to end up with a set of discriminant functions able to solve a multi-class classification problem. The main difference with well-known NN-classificators lean on the fact that training is performed over labeled sets of patterns that we call high-level-structures (HLS). Every set contain patterns linked each other by some physical evidence, like neighbor pixels in a subimage or a time-sequence of frequency vectors in a speech utterance, but the membership of every individual pattern in the high-level-structure can not be so clear. This architecture has been tested on a number of artificial data sets and real data sets with very good results. We are now applying the algorithm to classification of real images drawn from the DataBase created for the ALINSPEC project. This system has been developed in connection with ALINSPEC(Automatic Inspection of Alimentary Products): A BRITE-EURAM project partially supported by EEC under contract number BRE2-CT92-0132. Paper NN.5 RBF NETWORKS FOR DENSITY ESTIMATION Lucia Sardo and Josef Kittler Department of Electronic & Electrical Engineering University of Surrey, Guildford, Surrey GU2 5XH United Kingdom A non-parametric probability density function (pdf) estimation technique is presented. The estimation consists in approximating the unknown pdf by a network of Gaussian Radial Basis Functions (GRBFs). Complexity analysis is introduced in order to select the optimal number of GRBFs. Results obtained on real data show the potentiality of this technique. Paper NN.6 Title: OPTIMIZATION OF A NEURAL NETWORK APPLIED TO PULSED RADAR DETECTION. Authors: Diego Andina and Jos L. Sanz-Gonzlez Affiliation: Departamento de Seales, Sistemas y Radiocomunicaciones, ETSI de Telecomunicacin Universidad Politcnica de Madrid, Madrid, Spain. e-mail: andina@gc.ssr.upm.es Abstract: The purpose of this paper is to present the results of the optimization of some features of a Neural Network applied to the binary detection problem. We present how to design the structure and the training sets, and how to modify the BackPropagation algorithm to improve the results of the network for the binary detection task. The Neural Network, so designed, presents an optimal range of pulse integration and a performance very close to the theoretical limits, even under input distributions different from those used for training. Paper NN.7 NEURAL NETWORKS TO PREDICT OZONE POLLUTION IN INDUSTRIAL AREAS P. Arena, S. Baglio, L. Fortuna, G. Nunnari Dipartimento Elettrico, Elettronico e Sistemistico Viale A. Doria,6, 95125, Catania, ITALY e-mail: parena@dees.unict.it ABSTRACT In this paper a novel approach, based on a neural network structure, is introduced in order to face with the problem of pollutant estimation in an industrial area. In particular a short-term prediction (six hours ahead) of the O3 pollutant mean value has been performed. The results obtained show the capability of such structures to model complex chemical reactions heavily dependent on the meteorological conditions and on the typical geographical characteristics. Paper NN.8 MULTI-STAGE NONLINEAR CLASSIFICATION OF RESPIRATORY SOUNDS E. aatay Gler , Blent Sankur *, Yasemin P. Kahya *, and Sarunas Raudys Boazii University-Biomedical Engineering Institute, * Boazii University-Electrical Engineering Department, Institute of Mathematics and Informatics, e-mails: {gulerc, sankur, kahyay}@boun.edu.tr, Sarunas.Raudys@DAS.MII.lt ABSTRACT: The three-class recognition problem of respiratory sounds based on multi-stage decisions is addressed. The method consists of dividing respiratory cycles of patients into phases, and classifying each phase with a separate multilayer perceptron, called the phase expert. Each phase information consists of several time segments and their parametric representation. Expert decisions on phase segments are then combined by a decision fusion scheme, simulating a consultation session. Thus in the first stage of hierarchy one uses signal features to reach segment decisions, while in the second stage one uses decision votes themselves as features inputted into a second classifier. Furthermore a new regularization scheme is applied to the data to stabilize training and consultation. Paper NN.9 NEURAL NETWORK FOR CALCULATING ADAPTIVE SHIFT AND ROTATION INVARIANT IMAGE FEATURES Sabine Kroener Technische Informatik I Technische Universitaet Hamburg-Harburg 21071 Hamburg, Germany Tel/Fax: +49 [40] 7718 2539 / 7718 2911 e-mail: kroener@tu-harburg.d400.de Shift and rotation invariant pattern recognition is usually performed by first extracting invariant features from the images and second classifying them. This poses the problem of not only finding suitable features but also a suitable classifier. Here a structured invariant neural network architecture (SINN) is presented that performs adaptive invariant feature extraction and classification simultaneously. The network is sparsely connected and uses shared weight vectors. As a result features especially well suited for a given application are calculated with a computational complexity of O(N) for N = 2^n input elements. Experiments show the recognition ability of the invariant neural network on synthetic and real data. Paper NN.10 ROI-BASED IMAGE CODING USING MULTIRESOLUTION NEURAL NETWORKS Vassilios Alexopoulos and Stefanos Kollias Division of Computer Science Department of Electrical and Computer Engineering National Technical University of Athens Heroon Polytechniou 9, 15773 Zographou, Greece Tel: +30 1 772 2491; fax: +30 1 772 2459 e-mail: valex@image.ece.ntua.gr, stefanos@cs.ntua.gr In this paper is presented a ROI-based multiresolution coding scheme, whose main importance is that it achieves both high compression ratios and good reconstruction of the images. It uses optimal, in the mean square error sense, analysis and synthesis filters in the most significant areas (Regions of Interest) while conventional ones are used in the rest of the image. A linear autoassociative neural network architecture is proposed to compute the filters for optimal reconstruction of the images based on low resolution approximations of these. The characteristics of optimal filters are examined in 'head and shoulder' videoconferencing images. Paper PAP.1 A CYCLIC COHERENT METHOD FOR WIDEBAND SOURCE LOCATION Giacinto Gelli (1) and Luciano Izzo (2) (1) Seconda Universita' di Napoli, Dipartimento di Ingegneria dell'Informazione, via Roma, 29 I-81031 Aversa, Italy, E-mail: gelli@nadis.dis.unina.it (2) Universita' di Napoli Federico II, Dipartimento di Ingegneria Elettronica, via Claudio, 21 I-80125 Napoli, Italy, E-mail: izzo@nadis.dis.unina.it Abstract: The problem of source location of wideband signals impinging on an array of sensors is addressed. The proposed method exploits the cyclostationarity exhibited by most communication signals to discriminate signals of interest from noise and interfering signals. The new method performs coherent combination of the spatial contributions at different frequencies and exploits signal-subspace properties of the resulting focused matrix. Numerical results show that the proposed technique is superior to existing algorithms and assures good performances also when the signals of interest are fully correlated. Paper PAP.2 COMPARISON OF DOA ESTIMATION PERFORMANCE FOR VARIOUS TYPES OF SPARSE ANTENNA ARRAY GEOMETRIES Y. I. Abramovich[1], D. A. Gray[1], A. Y. Gorokhov[2] and N. K. Spencer[1] [1] Cooperative Research Centre for Sensor Signal and Information Processing (CSSIP), Technology Park, The Levels, South Australia, 5095, Australia Tel: +61 8 302 3328, Fax: +61 8 302 3124, e-mail:{yuri|dgray|nspencer}@cssip.edu.au [2] Departement Signal, Telecom Paris, 46 rue Barrault, 75634, Paris, Cedex 13, France Tel: +33 1 45 81 75 47, Fax: +33 1 45 88 79 35, e-mail:gorokhov@sig.enst.fr Three subclasses of geometries for nonuniform linear antenna arrays with a fixed number of sensors are compared in the sense of maximum possible direction-of-arrival (DOA) estimation accuracy. Cramer-Rao bound analysis is applied to compare the optimal accuracy for each geometry under some fixed source environment. Actual DOA estimation simulations, obtained by recently-introduced algorithms, are used to demonstrate the applicability of Cramer-Rao bound analysis for DOA estimation in these cases. We show that previous attempts to maximise the number of contiguous correlation lags and to avoid missing lags in certain array geometries does not necessarily lead to an improvement in DOA estimation performance. Paper PAP.3 EIGENVECTOR PEELING APPROACH TO COHERENT MULTIPLE SOURCE LOCATION PROBLEM Seenu S. Reddi, Alex B. Gershman Signal Research Lab., Clemons Circle, Irvine, CA Electrical Engineering Dept., Ruhr University, Bochum, Germany e-mail: gsh@sth.ruhr-uni-bochum.de We propose a novel preprocessing scheme, referred to as vector peeling, as an alternate to the conventional spatial smoothing for solving the multiple source location problem involving coherent sources or a rank deficient source covariance matrix. The essence of the technique is to preprocess the signal subspace eigenvectors rather than the covariance matrix as in spatial smoothing. It is shown by analysis and computer simulations that these two approaches are related, and that vector peeling slightly outperforms spatial smoothing when employed with the MUSIC-type DOA estimators. In certain instances, vector peeling offers advantages in terms of computational simplicity and flexibility. The latter is especially true with eigenstructure DOA estimators in adaptive estimation problems, i.e., when the signal subspace eigenvectors are updated using fast adaptive algorithms. Paper PAP.5 NEW GEOMETRICAL RESULTS ABOUT 4-TH ORDER DIRECTION FINDING METHODS PERFORMANCE P. Chevalier, A. Ferreol and J.P. Denis Thomson-CSF-Communications, 66 rue du Foss Blanc, 92231 Gennevilliers, France Tel: 33 1 46 13 26 98 ; Fax: 33 1 46 13 25 55 ABSTRACT : Since a decade, higher order direction finding (DF) methods have been developed for non gaussian signals. However, relatively few papers have been devoted to the performance analysis of these methods. The purpose of this paper is to present a geometrical analysis of the potential performance of these methods trough the new concept of "equivalent array", which makes possible the prediction of some of their performance. Paper PAP.6 PASSIVE IDENTIFICATION OF MULTIPATH CHANNEL Joel Grouffaud, Pascal Larzabal Henri Clergeot LESiR - ENS de Cachan - URA CNRS 1375 61, av. du President Wilson - 94235 Cachan - Francee Tel: +33 1 47 40 27 09 E-mail: joel.grouffaud@lesir.ens-cachan.fr Anne Ferrol Thomson CSF-RGS 66, rue du Foss blanc 92231 Gennevilliers-France ABSTRACT RF transmissions are often done along multipath channel, due to reflections. A physical model of propagating along such a channel is available, and takes into account few parameters as angles of incidence of waves on the array, group delay for each path, Doppler shift, polarisation. In order to compensate Rayleigh fading, a spatio-temporal separation of multipaths is proposed. Usually, this is done by transmitting a training sequence (known), which reduces the data rate. We show in this paper that a passive identification can be performed, using only received signals. Proposed algorithm proceeds in two steps: the first step is a blind deconvolution, and then a parametric estimation of the channel is performed. Many simulations exhibit performances of proposed algorithms. Paper PAP.7 Title: CRITERIA FOR COMPLEX SOURCES SEPARATION Author: Eric Moreau Affiliation: MS-GESSY, ISITV, Universite de Toulon Av. G. Pompidou, BP 56, 83162 La Valette du Var, France e-mail: moreau@isitv.univ-tln.fr Abstract: We consider the problem of sources separation. Two necessary and sufficient conditions involving high-order cumulants are given and proved. Hence, a family of criteria for source separation is obtained. A novel gradient based algorithm is derived in order to optimize the proposed criteria and various computer simulations are presented in order to illustrate the performances of the algorithm. Paper PAP.8 ROBUST BEAMFORMING FOR INTERFERENCE REJECTION IN MOBILE COMMUNICATIONS Jaume Riba, Jason Goldberg and Gregori Vazquez Department of Signal Theory and Communications. Universitat Politecnica de Catalunya. E.T.S.E.Telecomunicacio, Campus Nord, Edifici D5, c/Gran Capita s/n, 08034, Barcelona, Spain. The problem of robust beamformer design in the presence of moving sources is considered. A new technique based on a generalization of the constrained minimum variance beamformer is proposed. The method explicitly takes into account changes in the scenario due to the movement of the desired and interfering sources, requiring only estimation of the desired DOA. Computer simulations show that the resulting performance constitutes a compromise between interference and noise rejection, computational complexity, and sensitivity to source movement. Paper PAP.9 ADAPTIVE ARRAY BEAMFORMING FOR FREQUENCY HOPPPING MODULATION Montse Najar, Miguel A. Lagunas. Department of Signal Theory and Communications, Universitat Politecnica de Catalunya c/. Gran Capita s/n, 08034 BARCELONA, SPAIN Phone:34-3-4017051. Fax: 34-3-4016447. e-mail: najar@gps.tsc.upc.es A new architecture for Array Processing using Frequency Hopping (FH) modulation is addressed in this paper which takes advantage of the knowledge of the frequency sequence at the receiver, requiring neither temporal nor spatial a priori reference. Consequently, the paper deals with a Code Reference Beamformer (CRB). The proposed framework is composed of two parallel processors. The first one, the Anticipative processor, is devoted to predict the scenario at the hop frequency before this frequency is transmitted, providing a fast convergence of the second processor and avoiding the fall of the Signal to Interference plus Noise Ratio (SINR) with the frequency hops. The second one, the On-line processor, provides maximum SINR by applying the optimum beamvector which can be estimated minimizing the Mean Square Error (MSE) at the array output or, directly, maximizing the SINR. Paper PAP.10 ALGORITHMS AND STRUCTURES FOR SOURCE SEPARATION BASED ON THE CONSTANT MODULUS PROPERTY J.R. Cerquides, J.A. Fernandez-Rubio Signal Theory and Communications Department, Polytechnic University of Catalonia, E-mail: ramon@tsc.upc.es We propose two structures and theirs associated algorithms designed to solve the blind source separation problem in the presence of noise and interferences. Both structures exploit the non convexity of the Constant Modulus cost function, finding its multiple local minima. A convergence analysis shows that both schemes achieve the desired solution, separately extracting the sources of interest while rejecting noise and interferences, provided that they do not share the constant modulus property. Paper PAP.11 PARTIALLY ADAPTIVE GENERALIZED SIDELOBE CANCELLER WITH PRESCRIBED ZEROS Zoran M. Saric , Milorad Cetina Mathematical Institute, Kneza Mihaila 35, 11000 Beograd, Yugoslavia ABSTRACT Linear constraints in adaptive beamformer are often used to control its transfer function. In this paper we utilized these constraints to reduce computational cost of the adaptive algorithm. For this aim, two types of constraints were proposed. The first one is that all zeros of the transfer function appear as conjugate-complex pairs lying on the unit circle. The second one is that some zeros have prescribed positions and the adaptation is realized by the rest of zeros. Developed constraints are applied to the generalized sidelobe canceller and used to blocking matrix design. Experiments proved that degradation in performance of the partially adaptive algorithm is a little compared to the full adaptive algorithm.  Paper PAP.12 A REALIZABLE PARALLEL RLS PARAMETER ESTIMATOR F.M.F. Gaston and D.W. Brown Digital Systems and Vision Processing, University of Birmingham, Edgbaston, Birmingham, B15 2TT, UK Tel: +44 (0)121 414 4283; Fax: +44 (0)121 414 4291 e-mail: f.m.gaston@bham.ac.uk In this paper we derive, from a dependence graph, a rectangular parallel architecture for RLS parameter estimation. It has a number of advantages over the traditional triangular structure whilst maintaining the same throughput. These advantages are identical cells, easy expandability for increases in the number of parameters, reduced data flow, useful data is easily extracted and all these properties together make it more attractive for VLSI implementation. Paper PAP.13 WIDEBAND ARRAY PROCESSING USING A PARTITIONED SPECTRAL MATRIX S. Bourennane and M. Frikel C.M.C.S. URA 2053 CNRS, B.P. 52, Quartier Grossetti, 20250 Corte - France bourenna@univ-corse.fr - frikel@univ-corse.fr Abstract - This paper presents a propagator method for high resolution estimation of the angles of arrival of multiple wideband plane waves without eigendecomposition. The technique is based on a partition of the array spectral matrix. The noisy situation is considered and an algorithm to eliminate the noise contribution is given. The results of simulations support the theoretical predictions are presented. Paper PAP.14 FAST ALGORITHM FOR THE WIDEBAND ARRAY PROCESSING USING A TWO-SIDED CORRELATION TRANSFORMATION M. Frikel and S. Bourennane C.M.C.S. URA 2053 CNRS, B.P. 52, Quartier Grossetti, 20250 Corte - France frikel@univ-corse.fr - bourenna@univ-corse.fr Abstract - The purpose of this paper is the passive angular location of the wideband sources using an array of sensors. The improvement of the two-sided correlation transformation (TCT) is proposed, only the signal subspace estimated at each frequency is transformed by focusing matrices such that to obtain the coherent signal subspace for all the analysis band. The simulation results show that the proposed algorithm reduce the computational load compared to the original version TCT. Paper PAS.1 CASCADED ALL-PASS SECTIONS FOR LMS ADAPTIVE FILTERING Authors : H.J.W. Belt H.J. Butterweck Affiliation : Eindhoven University of Technology P.O. Box 513, 5600 MB Eindhoven, The Netherlands Tel: +31 (0)40 2473627, Fax: +31 (0)40 2448375 Email : H.J.W.Belt@ele.tue.nl H.J.Butterweck@ele.tue.nl ABSTRACT: The behaviour of the LMS adaptive algorithm is analyzed for a class of adaptive filters that is based on a cascade of identical N-th order all-pass sections. The well-known tapped-delay-line is a special case of this class. We look at the rate of convergence and the steady-state weight fluctuations. It is shown that in the steady state the weight-error correlation matrix satisfies a Lyapounov equation for sufficiently small values of the step-size. Sometimes a priori knowledge of the unknown reference system is available that can be used to select the N parameters of the all-pass section. In these cases the LMS adaptive filter based on a cascade of identical all-pass sections can outperform the LMS adaptive tapped-delay-line. Paper PAS.2 Title : AN APPROACH TO LMS ADAPTIVE FILTERING WITHOUT USE OF THE INDEPENDENCE ASSUMPTION Author : H.J. Butterweck Affiliation : Eindhoven University of Technology P.O. Box 513, 5600 MB Eindhoven, The Netherlands Tel: +31 (0)40 2473860, Fax: +31 (0)40 2448375 Email : H.J.Butterweck@ele.tue.nl ABSTRACT: Without use of the well-known "independence assumption" an exact analysis of the LMS-type tapped-delay line adaptive filter is provided, valid for small adaptation constants. For arbitrarily coloured excitations, the steady-state weight-error correlation matrix satisfies a Lyapounov equation, which under special conditions admits a closed-form solution. Paper PAS.3 Does Fractionally-Spaced CMA Converge Faster Than LMS ? A. Touzni, I.Fijalkow. ENSEA / ETIS, 95014 Cergy-Pontoise Cedex, France. fax: (33-1) 30 73 66 27 e-mail:touzni,fijalkow@ensea.fr ABSTRACT: This paper addresses the convergence rate study of the Fractionally-Spaced Equalizer updated by Constant Modulus Algorithm (FSE-CMA). By analyzing the average algorithm behavior we compare the FSE-CMA to the FSE-LMS. Although the FSE-CMA is based on a fourth order statistics criterion, we will show for constant modulus input that the algorithm has the amazing property to converge locally twice as fast as FSE-LMS (which requires a training sequence). Furthermore, we will show that the global FSE-CMA transient behavior convergence is accomplished in two steps. Paper PAS.4 Convergence Analysis of a Variable Step-Size Normalized LMS Adaptive Filter Algorithm Lillg QIN, Maurice G.BELLANGER Laboratoire Electronique et Communicalion, CNAM 292, rue St-Martin, 75141 Paris Cedex 03, France Tel. 33 1 40 27 20 82 Fax: 33 1 40 27 27 79. E-mail: qin@cnam.fr Abstract This paper investigates the convergence properties of a variable step normalized LMS (VSNLMS) adaptive filter algorithm. Instead of a fixed step-size used in the conventional normalized LMS algorithm, the step-size of the algorithm under study is updated in each iteration, based on an expression related to the output errors. The variable step-size improves the convergence speed, while sacrificing little in complexity. For an application where the adaptive filter is used to track a time varying channel it is shown that the step-size converges towards its optimum value. Simulation results are presented to support the analysis. Paper PAS.5 A NOVEL GIVENS ROTATION BASED FAST SQR-RLS ALGORITHM Alberto Carini Dipartimento di Elettrotecnica, Elettronica ed Informatica Universita di Trieste, Via Valerio 10, 34127 Trieste, Italy Tel: +39.40.676.7127; Fax: +39.40.676.3460 e-mail: carini@imagets.univ.trieste.it Abstract: A novel Fast RLS Algorithm based on the Givens Rotation and developed from an UD square-root factorization of autocorrelation matrix is discussed. The algorithm presents excellent numerical properties and requires 14 N multiplications and 6 N divisions per sampling interval, where N is the linear filter order. Paper PAS.6 ON THE ADAPTATION OF THE POLE OF LAGUERRE-LATTICE FILTERS T. Oliveira e Silva Universidade de Aveiro / INESC Aveiro 3810 AVEIRO PORTUGAL tos@inesca.pt The main purpose of this paper is to present some experimental results concerning the adaptation of the pole position of the lattice version of the Laguerre filter. Basically, we propose the adaptation of the Laguerre-lattice parameters with the \mbox{GAL-L} algorithm, and the adaptation of the pole with a suitable sign algorithm. An example and suggestions about how to attempt to avoid local minima (with respect to the pole position) are also given. Paper PAS.7 Title: CONVERGENCE BEHAVIOR OF TWO-DIMENSIONAL LEAST-SQUARES LATTICE ALGORITHM Authors: Takayuki NAKACHI , Katsumi YAMASHITA and Nozomu HAMADA Affiliation: Faculty of Science and Technology, Keio University (First and third authors) e-mail: naka@tkhm.elec.keio.ac.jp Faculty of Engineering, University of the Ryukyus (Second author) e-mail: yamasita@lark.ie.u-ryukyu.ac.jp Abstract: In this paper, we propose a two - dimensional (2-D) least-squares lattice (LSL) algorithm for the general case of the autoregressive (AR) model with an asymmetric half-plane (AHP) coefficient support. The resulting LSL algorithm gives both order and space recursions for the 2-D deterministic normal equation. The size and shape of the coefficient support region of the proposed lattice filter can be chosen arbitrarily. Although the 2-D signals of the model support are ordered into a one-dimensional (1-D) array, the ordering of the support signal can be assigned arbitrarily. Finally, computer simulation for modeling a texture image is demonstrated to confirm the proposed model gives rapid convergence. Paper PAS.8 Paper ID: 157 CHAOTIC TIME-SERIES PREDICTION AND THE RELOCATING-LMS (RLMS) ALGORITHM FOR RADIAL BASIS FUNCTION NETWORKS Authors: Afsar SARANLI Middle East Technical University, Ankara, TURKIYE. Tel: +90 312 2104419; fax: +90 312 2101261 e-mail: saranli@rorqual.cc.metu.edu.tr Buyurman BAYKAL Imperial College of Science, Technology and Medicine, London, UK. e-mail: b.baykal@ic.ac.uk ABSTRACT: In this study, the problem of real-time chaotic time-series prediction using Radial Basis Function Networks is addressed. The performance of a number of training methods based either on supervised error correction or on adaptive clustering techniques are investigated. Some performance drawbacks due to their exclusive usage are pointed out and a new algorithm combining their desirable properties is presented. The proposed {\em Relocating-LMS} algorithm is compared with the existing methods on a chaotic time-series produced by the Mackey-Glass Equation and is further tested on a series generated by the Logistic Map function, leading to encouraging results. Paper PAS.9 CYCLOSTATIONARY SPECTRAL ANALYSIS OF SUBBAND ADAPTIVE FILTERS Hideaki Sakai and Noriyuki Hirayama Graduate School of Engineering, Kyoto University Kyoto 606-01, Japan Tel: +81-75-753-5492; fax: +81-75-761-2437 e-mail: hsakai@kuamp.kyoto-u.ac.jp This paper presents cyclostationary spectral analysis of subband adaptive filters. First, the convergent point of the LMS type algorithm is determined. Next, using the spectral theory of cyclostationary processes, the cyclic spectral density matrix of the error signal is derived. Finally, its averaged variance is calculated for typical value of the delay in the desired signal and is compared with the simulation result. Paper PAS.10 SINGLE-LAYER PERCEPTRON BASED COMMUNICATION CHANNEL EQUALISATION WITH LEAST-MEAN-ABSOLUTE-ERROR ADAPTIVE ALGORITHM Changjing Shang, Murray J. J. Holt and Colin F. N. Cowan Department of Electronic and Electrical Engineering Loughborough University of Technology Loughborough, Leicestershire LE11 3TU, United Kingdom Tel: +44 1509 263171, Fax: +44 1509 222854 e-mail: scj@cee.hw.ac.uk, M.J.Holt@lut.ac.uk, C.F.N.Cowan@lut.ac.uk ABSTRACT This paper presents a novel approach to weight adaptation of single-layer perceptron (SLP) based communication channel equalisers, by developing the Least-Mean-Absolute-Error adaptive algorithm using the absolute-error cost function. Theoretical and experimental results are provided and comparisons made between the present algorithm and the traditional back-propagation, Rosenblatt and linear LMS algorithms. This work shows that the proposed algorithm is faster in adapting the weights of the SLP-based equalisers and leads to better estimation performance. Paper PAS.11 SUB-BAND, DUAL-CHANNEL ADAPTIVE NOISE CANCELLATION USING NORMALISED LMS David J. Darlington, Douglas R. Campbell Department of Electrical and Electronic Engineering, University of Paisley, High Street, Paisley PA1 2BE, UNITED KINGDOM. Tel: (+44) 141 848 3428 Fax: (+44) 141 848 3404 email: darl_ee0@helios21.paisley.ac.uk An adaptive noise cancellation scheme for speech processing is proposed. In this, the adaptive filters are implemented in frequency-limited sub- bands, based on a simplified model of the human cochlea. A modification to the basic LMS structure is introduced which guarantees stability and convergence at all times. This modification, a non-recursive normalisation, is used both in a wideband and in a sub-band implementation of the scheme. The effect of this normalisation on the quality of the processed speech is to cause the speech to be distorted, showing that there is no benefit to using normalised LMS in a sub-band scheme, whether the application uses classical or intermittent noise cancellation. Paper PAS.12 ADAPTIVE NOISE CANCELLATION OF DOPPLER SHIFTED SIGNALS: A LINEAR FRAMEWORK Stephan Weiss and Robert W. Stewart Signal Processing Division, Department of Electronic and Electrical Engineering University of Strathclyde, Glasgow G1 1XW, Scotland e-mail: weiss@spd.eee.strath.ac.uk In this paper we investigate the performance of single channel adaptive noise cancellation techniques for situations where the noise signal received by the two microphones cannot be related by a fixed weight canceller's (linear) digital filter due to Doppler shift on the two signals. A mathematical signal model is produced, which shows that the adaptive filter is in fact required to identify a time-varying system which incorporates Doppler shift, and potential rapid variations in signal power as the Doppler producing source passes the filter microphones. We present theory, simulated performance and real world performance for both the least mean square (LMS) and normalised LMS (NLMS) when operating in a Doppler noise environment. Paper PAS.13 UNSUPERVISED SEPARATION OF DISCRETE SOURCES WITH A COMBINED EXTENDED ANTI-HEBBIAN ADAPTATION Zied Malouche and Odile Macchi Laboratoire des Signaux et Systemes, CNRS, Sup'elec Plateau de Moulon 91192 Gif-sur-Yvette Cedex FRANCE Groupement de Recherche TdSI du CNRS e-mail: malouche@lss.supelec.fr} ABSTRACT In the classical methods of unsupervised source separation, the a priori hypothesis is independence of sources. In certain applications, there is some additional knowledge on the sources (statistics, distributions, alphabet...). It is the case with discrete sources with known alphabet. Then we can improve separation. Initialization of adaptation is done according to some known algorithm, e.g. thanks to an extended anti-Hebbian algorithm, provided there are not less sensors than sources. As soon as the separation performance index has reached some preassigned level, a second part which involves the output decision error is introduced in the increment. In a noiseless environment, this method allows complete cancellation of steady state adaptation fluctuations and perfect source recovery. Paper PAS.14 EXTENSION OF A HYPERSTABLE ADAPTIVE LINE ENHANCER FOR TRACKING OF MULTIPLE CISOIDS Mukund Padmanabhan and Petr Tichavsky IBM T. J. Watson Research Center, mukund@watson.ibm.com Academy of Sciences of the Czech Republic, tichavsk@utia.cas.cz A hyperstable ALE for tracking complex cisoids is presented. The ALE incorporates an adaptive IIR filter, with the convergence of the filter being conditional on the overall system being 'passive'. The passivity of the system depends on the location of the input cisoid frequencies, and it is shown that for the case of upto two cisoids, the system is passive for all distinct frequencies. For the case of larger number of cisoids, the system is passive for certain ranges of the cisoid frequencies. Simulations are also given to back up the theoretical results. Paper PAS.15 Title : ANALYSIS OF AN ADAPTIVE IIR FILTER FOR MULTIPATH TIME DELAY ESTIMATION Authors : H. C. So and P. C. Ching Affiliation : Department of Electronic Engineering, The Chinese University of Hong Kong Shatin, New Territories, Hong Kong Tel: 852 2609 8271; fax: 852 2603 5558 e-mail: hcso@ee.cuhk.edu.hk, pcching@ee.cuhk.edu.hk Abstract : A novel adaptive recursive algorithm is proposed for estimating the interpath delay of a radiated signal in a multipath environment. Using LMS-type adaptation, the estimator is computationally efficient and it provides direct measurements of multipath gain and delay on a sample-by-sample basis. The convergence dynamics and variances of system parameters are derived. It is shown that the optimal performance of the estimator approaches the Cramer-Rao lower bound (CRLB) for high signal-to-noise ratio (SNR) conditions. Computer simulations have validated the capability of the method to track time-varying delays accurately. Paper PBP.1 MODIFIED FOURIER TRANSFORM RECOGNITION OF ECHOGRAPHIC IMAGES Paolo Sirotti* Mauro Zanchetti ** Giorgio Rizzatto*** Fulvio Stacul**** * DEEI - University of Trieste, via A. Valerio,10, 34127 Trieste Italy Tel +39 40 6763453; fax +39 40 6763460; e-mail:sirotti@gnbts.univ.trieste.it ** Alcatel Italia S.P.A. str. Monte D'Oro, 14 Trieste Italy Tel +39 40 8322463 *** General Hospital, Servizio di Radiologia, Gorizia Italy Tel +39 481 592244 fax +39 481 535346 **** Cattinara Hospital, Istituto di Radiologia, Trieste Italy Tel +39 40 3994372 fax+39 40 910921 The recognition of echotexture in echographic images may fail due to the distortions introduced by the scan system. We have implemented a rotation and scale invariant recognition method of echographic textures. The significant features assumed to characterise the images are vectors whose components are the values of a modified Fourier transform (MFT) of the images. Our method assures a good reliability and allows a short computation time, also when implemented on small computers. The method has till now been proved over breast and thyroid images, exhibiting a very good discrimination capability. Paper PBP.2 Title: TEXTURAL 3-DIMENSIONAL MULTISCALE ANALYSIS OF MRI VOLUMES OF THE BRAIN Authors: Harry Hatzakis, Stephen Roberts and Ioannis Matalas Affiliation Department of Electrical and Electronic Engineering Imperial College of Science, Technology and Medicine London SW7 2BT, U.K. email: h.hatzakis@ic.ac.uk Abstract: We describe a method for textural feature extraction of MRI volumes of the brain and, based upon those features, a method for classification and assessment of the anatomical malformations of the brain, due to Alzheimer's Disease (AD). In our research, we make the hypothesis that there is enough detectable textural evidence from a 3D analysis of MR images of the brain to detect and identify the earliest structural changes of AD. To uniquely characterise structural malformations we construct a database of statistical information for 3D textures at different scales, using wavelet operators. The major goal at this stage of our research is to explore the inherent constraints imposed by the structure of the texture and its symbolic description. Our representation benefits from a unique method of parameter reduction, which gives an unambiguous description of the textures of the brain in 3D. One of the key attributes of this model is that, in the case of conflicting statements, it generates a low confidence estimate, thus allowing a local measure of reliability. Paper PBP.3 FILTERING BY APPROXIMATED DENSITIES APPLIED TO TEXTURE MODELLING FOR MAMMOGRAPHY Virginie Ruiz* & Anthony G. Constantinides** *ISMRA-ENSI, GREYC CNRS URA 1526, 6 Bd Marchal Juin, 14150 Caen Cedex, France. Tel: +33 31 45 27 05; fax: +33 31 45 26 98; e-mail: v.ruiz@greyc.ismra.fr **Imperial College of Science, Technology and Medicine, Dept Electrical, Electronic Engineering Exhibition Road, London SW7 2BT, UK. Abstract: Many techniques are currently used for breast abnormality location and breast cancer detection, in particular. One find statistical approaches involving second and/or higher order statistics. The applicability of the filtering by approximated densities (FAD) is here demonstrated. The FAD introduced to alleviate limitations due conventional Kalman modelling, is applied to texture modelling for mammography. This application uses the simplest form of FAD involving second order statistics. Paper PBP.4 AN ORIENTED FRACTAL ANALYSIS FOR THE CHARACTERIZATION OF TEXTURE. APPLICATION TO BONE RADIOGRAPHS T. Loussot*, R. Harba*, G. Jacquet**, C.L. Benhamou***, E. Lespessailles***, A. Julien****. *Laboratoire d'Electronique, Signaux, Images, ESPEO, Universit d'Orlans, B.P. 6744, 45067 Orlans, FRANCE TelFax : (33)38.49.45.37  (33)38.41.72.45, E-mail : harba@lesi.univ-orleans.fr **TSI, Universit Jean Monnet, 23, rue Paul Michelon, 42023 St Etienne Cedex 2, France, Tel : (33)77.42.18.77 ***Ple d'activit Rhumatologie, C.H.R. Orlans, 45100 Orlans-La Source, France, Tel : (33)38.51.44.69 ****Ecole Suprieure d'Energie et des Matriaux, 45100 Orlans La Source, France, Tel : (33)38.41.70.66 ABSTRACT: In this communication, we propose an oriented fractal analysis to characterize a texture. A frequency based method is used to measure the H parameter following different directions. The results are displayed on a polar diagram. Its analysis gives coefficients which quantify both the roughness of the texture and its anisotropy. This method is applied to the characterization of trabecular bone architecture by analysis of X-ray films. The whole acquisition process is optimized to obtain a good reproducibility of the results. Two studies show the medical interest of the method. Paper PBP.5 NEW APPROACHES TO ROBUST GAUSSIAN MIXTURE ESTIMATION FOR BRAIN MRI Philippe SCHROETER, Jean-Marc VESIN Signal Processing Laboratory Swiss Federal Institute of Technology CH-1015 Lausanne, Switzerland tel: (+41 21) 693 4622 fax: (+41 21) 693 7600 e-mail: schroep@ltssg4.epfl.ch This paper presents two new methods for robust parameter estimation of mixtures in the context of MR data segmentation. The head is constituted of different types of tissue that can be modeled by a finite mixture of multivariate Gaussian distributions. Our goal is to estimate accurately the statistics of desired tissues in presence of other ones of lesser interest. These latter can be considered as outliers and can severely bias the estimates of the former. For this purpose, we introduce a first method, which is an extension of the EM-algorithm, that estimates parameters of Gaussian mixtures but incorporates an outlier rejection scheme which allows to compute the properties of the desired tissues in presence of atypical data. The second method is based on genetic algorithms and is well suited for estimating the parameters of mixtures of different kind of distributions. Experiments on synthetic and real MR data show that accurate estimates of the gray and white matters parameters are computed. Paper PBP.6 VASCULAR NETWORK TRACKING IN SLO OCULAR FUNDUS IMAGES FOR STATIC AND DYNAMIC PARAMETER EXTRACTION Tomaso Bufalini*, Ada Fort*, Leonardo Masotti*, Riccardo Pini* *Dept. of electronic Engineering - University of Florence Via S. Marta 3 - 50139 Florence - Italy, E-mail uscnd@ingfi1.ing.unifi.it. ABSTRACT In this work a tracking algorithm which extracts the vascular network structure from fundus Scanning Laser Ophthalmoscope (SLO) images is presented. The tracking algorithm is based on a priori knowledge of the vessel structure. It exploits the continuity of radius, position, direction and brightness of a blood vessel and is based on a recursive strategy. First, a main vessel is tracked and its branch points on both sides are identified. Then, the tracking process is applied again to the identified branches. The procedure is repeated till no branch points are found. The output of the algorithm is a structural description of the vascular network consisting of vessel position, radius and curvature. The presented algorithm was developed for images obtained using a contrast agent (fluoroangiography) but was also adapted to images without any contrast agent.The algorithm was tested both on simulated and real images and proved to give accurate measurement of vessel radius and position (mean errors below 1 pixel). Paper PBP.7 IMAGE QUALITY EVALUATION FOR RADIATION DOSE OPTIMIZATION IN CR BY SHAPE AND WAVELET ANALYSES Jianhua Xuan, Tulay Adali, Eliot Siegel, and Yue Wang Department of Computer Science and Electrical Engineering, University of Maryland Baltimore County, Baltimore, MD 21228, USA Tel: (410)455-3521, fax: (410)455-3969, e-mail: {xuanj, adali}@engr.umbc.edu Dept. of Diagnostic Radiology and Nuclear Medicine, Baltimore VA Medical Center, Baltimore, MD 21201, USA ISIS Center, Department of Radiology, Georgetown University Medical Center, Washington, DC 20007, USA It is one of the primary responsibilities of any department of diagnostic radiology to minimize the amount of unnecessary radiation administered to patients during diagnostic procedure. In this paper, we present three effective ways of quantifying the information content of computed radiography (CR) images for radiation dose optimization through shape and wavelet analyses. The experimental results demonstrate that the shape and wavelet analyses can be efficiently used to determine an optimum radiation dosage in computed radiography. Paper PBP.9 MATCHED MEYER NEURAL WAVELETS FOR CLINICAL AND EXPERIMENTAL ANALYSIS OF AUDITORY AND VISUAL EVOKED POTENTIALS V. J. Samar(1,5) H. Begleiter(2) J. O. Chapa(3) M. R. Raghuveer(4) M. Orlando(5) D. Chorlian(2) 1. National Technical Institute for the Deaf, Rochester Institute of Technology, Rochester, NY 14623, phone: 716-475-6338, fax 716- 475-6500, vjsncr@rit.edu 2. Department of Psychiatry, State University of New York Health Science Center, Brooklyn, NY 11203 3. Hanscom Air Force Base, Code ESC/AW, Hanscom AFB01731, Massachussetes 01731 4. Electrical Engineering Department, Rochester Institute of Technology, Rochester, NY 14623 5. Otolaryngology Division, University of Rochester Medical Center, Rochester, NY 14620 The wavelet transform provides a time-scale analysis that permits flexible pattern recognition, component identification, and detection of transients for time-varying neural signals such as the EEG, event-related potentials, neuromagnetic signals, and other neural signals and images. Many future applications to neural signals will benefit from choosing a mother wavelet that mimics neural waveform features. We use a recently developed algorithm to design physiologically realistic orthonormal Meyer wavelets, including 1) a wavelet that matches the prominent IV-V complex of the auditory brainstem evoked response used widely for clinical evaluation of hearing loss, and 2) a wavelet that matches ERPs containing prominent P300 components from control and alcoholic subjects. We also compare the relative naturalness of dyadic decompositions that use matched Meyer wavelets, the Haar wavelet, and Daubechies D4 wavelet. Designer neural wavelets have broad potential to customize and improve neurometric imaging and clinical neurodiagnosis of sensory and cognitive dysfunction. Paper, page 1 Paper, page 2 Paper, page 3 Paper, page 4 PBP.10 DOUBLE TREE DECOMPOSITION OF LUNG SOUNDS E. Ademovic (+*), J.-C. Pesquet (+), G. Charbonneau (*) (+) Laboratoire des Signaux et Systemes, CNRS/Univ. Paris-Sud and GdR-PRC ISIS, ESE, Plateau de Moulon, 91192 Gif sur Yvette, France. (*) Institut d'\'Electronique Fondamentale Universit\'e de Paris-Sud, 91405 Orsay, France. e-mail: ademovic@lss.supelec.fr Abstract: The analysis of respiratory sounds highlights the limits of commonly used techniques as a huge variety of sounds can be observed (stationary or nonstationary and of different durations) which can have themselves a great variability. New approaches have been developed in order to associate the acoustic phenomena to the respiratory flow and volume. We present here another approach only based on the wavelet packet decomposition to segment respiratory sounds. Paper PBP.11 Title: FOETAL ECG EXTRACTION USING BLIND SOURCE SEPARATION METHODS Authors: E. Bacharakis, A. K. Nandi and V. Zarzoso. Affiliation: Department of Electronic and Electrical Engineering, University of Strathclyde, 204 George Street, Glasgow G1 1XW, U.K. e-mail: asoke@eee.strath.ac.uk Abstract: Three methods to achieve Blind Source Separation are applied to the foetal electrocardiogram (ECG) extraction problem: Principal Component Analysis (PCA), Higher-Order Singular Value Decomposition (HOSVD) and Higher-Order EigenValue Decomposition (HOEVD). The first one gives uncorrelated source signals by means of second-order tools, while the last two resort to higher-order statistics of the data signals, so higher-order independence is attained. When tested on real ECG data, the last two produce better results than the former, with the HOEVD yielding the best performance, as expected from the theoretical unfolding. Paper PBP.12 INFLUENCE OF THE SINUSOIDAL AND GAUSSIAN NOISES IN THE ESTIMATION OF THE EEG FRACTAL DIMENSION A.Accardo*, M.Affinito** and M.Carrozzi** * D. E. E. I., Universita' di Trieste, Via A.Valerio 10, 34127 Trieste, Italy Tel: +39 40 6767148; fax: +39 40 6763460 e-mail: accardo@gnbts.univ.trieste.it ** I.R.C.S.S. Osp. Infantile "Burlo Garofolo", Via dell'Istria 65/2, Trieste Tel: +39 40 3785302 ABSTRACT EEG signals corresponding to different psychophysiological conditions can be characterized by their fractal dimension (D). The noises present on the recording can affect the estimation of such a dimension. In this work we analyse the behaviour of two D estimators in case of different kinds (gaussian and sinusoidal) and amplitudes of noise. The EEG fractal dimension seems to be strongly compromised by gaussian noise greater than about 3-4%, of the EEG rms value, while a 50Hz noise of about 10% of the EEG rms signal is necessary to produce estimation errors greater than 10%. A dependence on the sampling frequency of the D estimation is also pointed out. Paper PBP.13 FUZZY - WEIGHTED AVERAGING FOR HIGH-RESOLUTION ECG BASED ON EXPLORATORY DATA ANALYSIS N. Laskaris :1, S. Fotopoulos :2, A. Bezerianos :1, A. Manolis :3 Department of Medical Physics 1 / Physics 2 / Cardiologic clinic 3 University of Patras, GR-26500, Patras, GREECE Tel.: +30 61 996115, FAX: +30 61 997745, email: bezer@upatras.gr ABSTRACT In this work we introduce a method for the enhancement of Late Potentials in the Signal Averaged electrocardiography. The method involves computation of weights prior averaging. Two fuzzy control techniques are proposed for the derivation of weights. The experimental results indicate the contribution of the method to a more reliable prognosis. Paper PBP.14 FAST AND ACCURATE PARAMETER ESTIMATION OF NOISY COMPLEX EXPONENTIALS WITH USE OF PRIOR KNOWLEDGE Leentje Vanhamme Aad van den Boogaart Sabine Van Huffel Katholieke Universiteit Leuven leentje.vanhamme@esat.kuleuven.ac.be In this paper we address the problem of parameter estimation of magnetic resonance spectroscopy (MRS) signals. MRS signals are modeled as complex exponentials in noise. Iterative methods based on an optimisation procedure can be used for the parameter estimation. We examine which functional we have to minimise and which nonlinear least squares algorithms we have to use in order to attain maximum efficiency and robustness. The influence of starting values and prior knowledge is examined. Paper PBP.15 STATISTICAL STUDY OF THE DELAY VARIANCE ESTIMATION FOR THE INDIVIDUAL AND GLOBAL METHODS O. Meste, E. Bataillou, H. Rix Laboratoire I3S-CNRS URA 1376 Bat. 4, Les Lucioles, 250 Av. Albert Einstein Sophia Antipolis 06560 Valbonne FRANCE e-mail: meste@essi.fr When the variance of the delays is assumed to be relevant in a series of recurrent signals, two approaches are encountered. Either each delay is estimated allowing the computation of the sample variance (individual method) or the expected variance is directly estimated (global method). These two approaches are statistically compared using the global method introduced in a previous work and two individual methods: a Averaged Square Difference Function based estimator and a linear system based one. We finally show that the global method exhibits an interesting behaviour mainly due to its unbiasness. Paper PC.1 USING ORTHOGONALIZED VOICE FOR SIMULTANEOUS TRANSMISSION OF VOICE AND DATA M. Goren, O. Tirosh, L. Kishon-Rabin* and D. Wulich Department of Electrical & Computer Engineering, Ben-Gurion University of the Negev. Beer-Sheva 84105, POB 635, Israel. Tel: ++972-7-461537, Fax: ++972-7-472949 e-mail: dov@bguee.bgu.ac.il *School for Communication Disorders, Speech, Language and Hearing, Sackler Faculty of Medicine, Tel-Aviv University, Israel ABSTRACT Voiceband channels are frequently used for data transmission, even though they were not designed for such a use. The reason is very simple; such channels already exist. It is also clear that such channels when used for data transmission can not be used at the same time for voice transmission, and vice versa. However, there are a lot of applications where simultaneous transmission of voice and data through the existing voiceband channel is needed. In this work we propose a method for simultaneous transmission based on orthogonalization of the voice signal. A comprehensive assessment of the orthogonal voice which includes subjective measures shows that the orthogonal signal may have full intelligibility while its quality is only slightly degraded. The MOS for orthogonal voice is in the range 2.5 - 3.9 and depend on the data transmission parameters. Paper PC.2 MODELLING MAN-COMPUTER ORAL DIALOGUE IN NOISY ENVIRONMENT Josef Psutka, Jiri Kepka, Ludek Muller, Zbynek Tychtl University of West Bohemia, Department of Cybernetics, Univerzitni 22, 306 14 PILSEN, Czech Republic e-mail: psutka@kky.zcu.cz ABSTRACT A model of a voice controlled system will be presented in the paper. The behaviour of the system is modelled by a finite state process. The problem domain is supposed to be well bounded and very limited. An isolated word classification method is used for the recognition of users control command or sequence of commands. A speech synthesizer is used to implement acoustic feedback control. As an illustration example its implementation and application for searching and updating database is described. Problems involved in classification and communication between the system and the user in noisy environment are treated. Optimization tradeoffs are proposed. Paper PC.3 AN ADAPTIVE BLIND EQUALISER WITH AUTOMATICALLY CONTROLLED STEP-SIZE Bee Eng Toh and Desmond C McLernon Department of Electronic and Electrical Engineering The University of Leeds Leeds LS2 9JT United Kingdom Tel : 0113 2332075 Int :+44 113 2332075 Fax : 0113 2332032 email : eenbet@sun.leeds.ac.uk ABSTRACT: Although research into blind equalisation has been on-going for more than a decade, the existing blind equalisation algorithms are often inefficient in combating the impairments introduced by mobile communication channels. New and efficient algorithms are hence needed. The performance characteristics of a recently proposed blind clustering technique in the presence of frequency selective fading and Doppler-effects in a mobile communication environment is first studied. A new, fast convergence algorithm is then introduced based on modification of the Super-Exponential (SE) algorithm, followed by a clustering technique with an automatically controlled step-size. Simulations show that the new algorithm converges very fast and can get rid of the constellation rotation problem encountered when applying the SE method to time-varying channels. Paper PC.4 EFFICIENT CLUSTERING TECHNIQUES FOR SUPERVISED AND BLIND CHANNEL EQUALIZATION IN HOSTILE ENVIRONMENTS Sergios Theodoridis and Kristina Georgoulakis University of Athens Department of Informatics TYPA Buildings 15771 Athens Greece Tel : +(301) 7211119 Fax : +(301) 7228981 email : stheodor@di.uoa.gr - kristina@di.uoa.gr In this paper the equalization problem is treated as a classification task. No specific (linear or nonlinear) model is required for the channel or for the interference and the noise. Training is achieved via a supervised learning scheme. Adopting Mahalanobis distance as an appropriate distance metric, decisions are made on the basis of minimum distance path. The pro- posed equalizer operates on a sequence mode and implements the Viterbi searching Algorithm. The robust performance of the equalizer is demonstra- ted for a hostile environment in the presence of CCI and non linearities, and it is compared against the performance of the MLSE and a symbol by symbol RBF equalizer. Suboptimal techniques with reduced complexity are di- scussed. The operation of the proposed equalizer in a blind mode is also considered. Paper PC.5 PERFORMANCE OF AN ADAPTIVE KALMAN EQUALISER ON TIME VARIANT MULTIPATH CHANNELS Tetsuya Shimamura, Colin F.N. Cowan Department of Electronic and Electrical Engineering, Loughborough University of Technology T.Shimamura@lut.ac.uk This paper develops an adaptive equaliser which utilises the Kalman filtering to reconstruct the transmitted sequence in time variant environments. The adaptive Kalman equaliser(AKE) addressed by Mulgrew and Cowan is modified by adopting a channel estimator, coefficients of which are updated by a gradient algorithm with fading memory prediction. By computer simulations, the performance of the AKE is investigated, and shown to be superior to that of the decision feedback equaliser(DFE) involving the adaptation of recursive least squares(RLS) algorithm in the case of a second order Markov communication channel model. Paper PC.6 Title: UNBIASED MMSE DECISION-FEEDBACK EQUALIZATION FOR PACKET TRANSMISSION Authors: Dirk T.M. Slock and Elisabeth de Carvalho Affiliation: Institut EURECOM, 2229 route des Cretes, B.P. 193 06904 Sophia Antipolis Cedex, FRANCE Tel: +33 93002606 Fax: +33 93002627 email:{slock,carvalho}@eurecom.fr Abstract: We derive expressions for the different linear and decision feedback equalizers in burst mode in the multichannel case. Among them we derive the class of unbiased minimum mean squared error equalizers. Optimal burst mode filters are found to be time-varying. Performance comparisons between these equalizers are done in terms of SNR and probability of error: these measures depend on the position in the burst. We study furthermore the performance when symbols are known or not at the edges of the burst and compare it to the continuous processing level. Finally we show that (time-invariant) continuous processing applied to burst mode can be organized to give sufficiently good performance, so that optimal (time-varying) burst processing implementation can be avoided. Paper PC.7 BLIND MAXIMUM LIKELIHOOD SEQUENCE DETECTION OVER FAST FADING CHANNELS David J. Reader and William G. Cowley* Communications Division, Defence Science and Technology Organisation, PO Box 1500, Salisbury, SA, 5108 Telephone: +61 8 259 6588 Facsimile: + 61 8 259 6549 E-Mail: david.reader@dsto.defence.gov.au *The Institute for Telecommunications Research, University of South Australia, The Levels, Pooraka, SA, 5095 Telephone: +61 8 302 3316 Facsimile: + 61 8 302 3873 E-Mail: bill.cowley@unisa.edu.au Maximum a posteriori (MAP) sequence detection for channels with intersymbol interference (ISI) has previously required knowledge of the channel sampled impulse response (SIR). Generally the SIR coefficients are determined via least mean square (LMS) or recursive least squares (RLS) estimation algorithms. For many unguided media channels such as mobile radio and high frequency radio which exhibit a time-varying SIR, these estimators must be adaptive. Adaptive estimators often fail to track adequately and are a major source of detector deterioration. A novel, blind maximum likelihood sequence detection (BMLSD) formulation without the need for external channel SIR estimation is proposed. The BMLSD performance is evaluated via simulation over several fast Rayleigh fading channels, which indicates substantial improvement compared to the conventional MLSD. Paper PC.8 COMBINED MATCHED FILTER/INTERPOLATOR FOR DIGITAL RECEIVERS S.Ries*, M.Th. Roeckerath-Ries** *Universitat-GH-Paderhorn, Abt. Meschede Lindenstr. 53, D-59872 Meschede Tel +49 291 991076; fax +49 291 991040 ** Markische Fachhochschule, Fachbereich Elektrotechnik Haldener Str. 182, D-58093 Hagen Tel +49 2331 987 2368; fax +49 2331 987 2326 ABSTRACT Timing adjustment in digital receivers is usually performed by an interpolator following the matched filter. With a root-cosine pulse with rolloff 0.5, linear interpolation with 2 samples per symbol leads to SNR loss. In this paper, it is shown that the receiver structure can be simplified, and that the SNR-loss can be reduced. This is achieved by the construction of novel strictly timelimited root-Nyquist pulses with good spectral properties. Using these pulses, combination of matched filter and interpolator for timing adjustment in digital receivers with negligible SNRloss up to a BER of 10-6 for BPSK is possible at two samples per symbol. Paper PC.9 BLIND MULTIUSER ADAPTIVE COMBINING FOR ASYNCHRONOUS CDMA SYSTEMS Olga Muoz, Juan A. Fernndez Rubio Dpt. Teoria Senyal i Comunicacions, ETSETB, Universitat Politcnica de Catalunya (UPC) e-mail: olga@gps.tsc.upc.es This paper presents a novel technique to globally estimate and track the direction of arrival (DOA) of different users in an asynchronous CDMA system. The estimates are obtained exploiting the temporal structure of CDMA signals. No training signal nor a priori spatial information is required. The necessary information is extracted directly from the received signals. The proper combining of the overall information present at the receiver after the despreading, jointly with an Eigenvalue Decomposition (EVD), let us estimate the generalized steering vector for each user. Furthermore, a direct iteration method is introduced in our scheme in order to make the array robust to channel variations and to reduce the computational load of the EVD required for each user. Paper PC.10 EVOLUTIONARY ARMA MODELLING FOR AERONAUTICAL COMMUNICATIONS Jamila BAKKOURY, Francis CASTANIE and Daniel ROVIRAS National Polytecnics Institute of Toulouse LEN7/GAPSE,Toulouse, France email: bakkoury@len7.enseeiht.fr This contribution deals with modelling and equalization for the aeronautical channel. This channel is subject to multipath and is characterized by a time-variant impulse response. An evolutionary ARMA model is proposed for such a non stationary channel . Evolutionary ARMA models are presented, they are used to derive a parameter estimator which is based upon an eigenformulation for a minimization criteria. Paper PC.11 TITLE: NEW TECHNIQUES FOR THE BAUD DURATION ESTIMATION AUTHORS: E. E. Azzouz and A. K. Nandi AFFILIATION: Department of Electronic and Electrical Engineering, University of Strathclyde, Glasgow, G1 1XW, U. K. Tel: +44 141 552 4400; Fax: +44 141 552 2487 Email: asoke@eee.strath.ac.uk ABSTRACT: The aim of this paper is to introduce fast and reliable baud duration estimators. This work is concerned with the symbols transitions sequence extraction and the baud duration estimation. The symbols transitions sequence is extracted using one of three methods - the level-crossing method, the derivative method and the wavelet method. Subsequently, the baud duration is estimated by applying the greatest common divisor principles on the symbols transitions difference sequence. Paper PC.12 A Soft Receiver Using Recurrent Networks Lorenzo Favalli*, Alessandro Mecocci**, Rita Pizzi* *Universit di Pavia,- Dipartimento di Elettronica via Ferrata, 1, I-27100 Pavia (PV) Italy; Tel: +39-382-505923; fax: +39-382-422583; e-mail: lorenzo@comel1.unipv.it **Universit di Siena,- Facolt di Ingegneria; via Roma, 77, I-53100 Siena (SI), Italy tel: +39-577-2636041 fax: +39-577-263602; e-mail: mecocci@comel1.unipv.it Abstract. Two different neural network architectures have been used to realize a non-linear adaptive receiver for GSM signals. Using the well-established backpropagation technique we firstly built a recurrent network which has been trained considering different channels corrupted by ISI, fading and Doppler. The network has shown better performances than the a classic coherent receiver. A second recurrent architecture, based on a partially supervised Self Organizing Map, has been proposed in order to perform an effective real time learning . Paper PC.13 SELF CALIBRATING LOW IF DIGITAL IMAGEREJECTION RECEIVER FOR MOBILE COMMUNICATIONS Jos M. Pez-Borrallo , Francisco J. Casajs Quirs, Santiago Zazo * ETSI Telecomunicacin, Universidad Politcnica de Madrid, Spain Phone: 341-3367280, Fax: 341-3367350, email: paez@gaps.ssr.upm.es * Universidad Alfonso X El Sabio, Villanueva de la Caada, Madrid, Spain ABSTRACT Here we present and develop a receiver capable of capturing two RF channels at the same time with a single RF front end and only one IF stage. The idea is to use a low IF digital image rejection receiver that can separate two adjacent RF channels with a negligible cochannels image interference. We analyze two procedures of compensating, in the IF range, any gain and phase misadjustment generated in the RF mixing section that could produce some residual images in any of the channels. The first one needs the help of an internal reference or pilot signal whereas the second one implements a blind procedure that only needs the current working signals. Paper PC.14 CLASSIFICATION OF LINEAR MODULATIONS BY MEAN OF A FOURTH-ORDER CUMULANT Denys Boiteau* and Christophe Le Martret** * CESTA, 37 av. du Gnral de Gaulle, 35170 Bruz, France, e-mail: cesta@broceliande.galeode.fr ** Centre d'Lectronique de l'ARmement, 35170, Bruz, France, e-mail: lemartre@celar.fr In this paper, we present a new linear modulation classification method based on a fourth-order cumulant of the stationary signal. Under some hypothesis, this method can be applied to carrier-modulated or baseband signals and doesn't need the knowledge of the signal to noise ratio. An example of classification is given for 4 PSK vs. 16 QAM modulations. Theoretical performance are approximated and compared to simulation results. The system achieves more than 90 % of correct classification for only 500 transmitted symbols and a signal to noise ratio of 0 dB. Paper PC.15 SOME NEW ARQ PROTOCOLS FOR PERSONAL COMMUNICATION SYSTEMS Alessandro Andreadis, Giuliano Benelli, Andrea Garzelli School of Engineering, University of Siena Via Roma, 56, 53100 Siena, Italy Tel: +39 577 263601; fax: +39 577 263602 benelli@unisi.it Mobile communication channels are frequently plagued by severe noise and disturbances such as multipath fading and doppler effects that severely degrade performance. Among the automatic-repeat-request (ARQ) protocols used to improve the communication channel reliability, the stop-and-wait (SW) is positively characterized by simple implementation and negatively by low throughputs. This work describes the application of some new SW protocols that retain the simple implementation of the classical SW schemes, while reducing the transmitter's wait state time to increase throughput. The performance of the modified SW protocols, derived through computer simulations, is shown to be comparable to that of more complex ARQ protocols. Paper PDE.1 PERFORMANCE OF AN OPTIMAL MULTIPLICATIVE JUMP DETECTOR BASED ON THE CONTINUOUS WAVELET TRANSFORM Marie CHABERT, Jean-Yves TOURNERET and Francis CASTANIE ENSEEIHT/GAPSE National Polytechnics Institute of Toulouse chabert@len7.enseeiht.fr Additive and multiplicative abrupt changes in random signals have been studied in many applications. In segmentation theory, the detection of these additive abrupt changes allows the determination of stationary parts of signals. In radar images, multiplicative abrupt jumps have been used to model speckled'' signal: these multiplicative jumps correspond to object edges on piecewise constant backgrounds. The Continuous Wavelet Transform (CWT) has shown nice properties for the detection of abrupt additive jumps. The paper studies the problem of abrupt multiplicative jump detection using the CWT. The time-scale plane Neyman-Pearson test is studied and its performance is evaluated. Paper PDE.2 LINE SPECTRUM PAIRS IN PATTERN RECOGNITION Jean-Yves TOURNERET and Mounir GHOGHO ENSEEIHT/GAPSE, National Polytechnics Institute of Toulouse 2 rue Camichel, 31071 Toulouse, France email: tournere@len7.enseeiht.fr The optimal Bayesian Classifier is often difficult to implement because of its complexity. For Gaussian parameters, the Bayes decision rule reduces to a simple centroid distance rule. However, the centroid distance rule fails for non-Gaussian parameters with non-convex probability density functions (p.d.f.). This paper studies some statistical properties of Line Spectrum Pairs (LSP). These statistical properties can be used to study the convexity of LSP point clusters in pattern recognition applications.  Paper PDE.3 CFAR DETECTOR FOR BACKGROUND NOISE WITH TWO-PARAMETER DISTRIBUTION G. de Miguel Vela, J. J. Martnez Madrid *, J. I. Portillo Garca Dep. Seales, Sistemas y Radiocomunicaciones ETSI Telecomunicacin - Universidad Politcnica de Madrid Ciudad Universitaria s/n, 28040 - MADRID (SPAIN) * Universidad Alfonso X el Sabio Avenida de la Universidad, Villanueva de la Caada (MADRID - SPAIN) ABSTRACT: We present a double-parameter CFAR with very reasonable losses and low computational complexity. Its basic architecture has been conceived from tail extrapolation theory. The detector uses a detection threshold, set from the measured PFA which is obtained with an auxiliary threshold (pseudothreshold), lower than the final detection threshold. Starting from the basic scheme, a CFAR detector for Weibull clutter has been designed; both, the pseudothreshold control mechanism and the correction of the basic extrapolation equation are described. Paper PDE.4 ASYMPTOTIC PERFORMANCE ANALYSIS OF THE SINGLE-CYCLE DETECTOR P. Rostaing, E. Thierry, T. Pitarque I3S UNSA rostaing@alto.unice.fr The paper deals with the analytical performance of the single-cycle detector, which is based on the cyclostationary properties of the signal to be intercepted. The Receiver Operating Characteristics (ROC) are derived theoretically, in discrete time, by using the asymptotic complex normality and covariance expressions of the sample average estimator of the cyclic-covariance when some mixing conditions'' are verified. Performance analysis of the single-cycle detector is evaluated for a cyclostationary signal observed in a background of stationary, zero-mean, white Gaussian noise. A numerical example for interception of a Binary-Phase-Shift-Keying (BPSK) signal is considered. Paper PDE.5 USE OF FOURTH-ORDER STATISTICS FOR NON-GAUSSIAN NOISE MODELLING: THE GENERALIZED GAUSSIAN PDF IN TERMS OF KURTOSIS A. Tesei, and C.S. Regazzoni DIBE, University of Genoa Via all'Opera Pia 11A, 16145 Genova, Italy Tel: +39 10 3532792; fax: +39 10 3532134 e-mail: tale@dibe.unige.it ABSTRACT In this paper non-Gaussian noise modelling is addressed. HOS-based parametric pdf models are investigated in order to provide realistic noise modelling by means of easy and quick estimation of needed parameters. Attention is focused on the generalized Gaussian pdf. This model, generally depending on a real theoretical parameter c, difficult to estimate from data, is proposed expressed in terms of the fourth-order parameter kurtosis b2 by introducing the analytical relationship between c and b2. The model is compared with well-known pdfs and used in the design of a LOD test. Paper PDE.6 DISCRETE HMMs FOR CLASSIFICATION OF MIXTURES OF SIGNALS Christophe Couveur, Vincent Fontaine, Henri Leich Service de Theorie des circuits et de traitement du signal Faculte Polytechnique de Mons, B-7000 Mons (Belgium) e-mail couvreur@thor.fpms.ac.be The concept of mixtures of discrete HMMs (MDHMM) is introduced. The application of MDHMMs to the classification of mixtures of signals is described. The optimal decision rule is presented. Alternative algorithms with reduced computational load are proposed: a simplified decision statistic is defined and sub-optimal search methods are discussed. The performance of the various algorithms are analyzed on Monte-Carlo simulations. Paper PDE.7 MODEL ORDER SELECTION IN UNKNOWN CORRELATED NOISE: A SUPERVISED APPROACH P.Costa, J. Grouffaud, P. Larzabal and H. Clergeot [M LESIR-ENS Cachan, IJRA CNRS D 1375, 61, av. du Pdt Wilson, 94235 CACHAN cedex France tel. +33 147 40 27 09 E-mail pascale.costa@lesir.ens-cachan.fr ABSTRACT The purpose of this paper is to propose the design and the use of a Neural Network for model order selection. The proposed neural network learns from real life situation by constructing an imput/output mapping (for detection) which brings to mind the notion of non parametric statistical inference. Such a strategy can improve performances of traditional tests relying on Iinearity, stationarity and second order statistics. We focus on the case where the noise covariance matrix is unknown but is a band matrix. This paper includes simulations which show improvements obtained by supervised approach. Paper PDE.8 EQUALIZER EVALUATION IN INTEGRATED DATA AND CHANNEL ESTIMATION Luca D'Ambrosio SYSFER Quality System Srl via Feltrino, 65128 Pescara - Italy Rossano Marchesani ALCATEL TELSPACE DED/STAS 5, rue Noel Pons, 92734 Nanterre - France Marina Ruggieri Universita' di L'Aquila, Dipartimento d'Ingegneria Elettrica 67040 Poggio di Roio, L'Aquila - Italy ABSTRACT - Per-Survivor Processing is a general approach which includes in the survivors of the Viterbi Algorithm trellis, the relative estimation of unknown parameters; this expensive method better approximates the optimum decoder in certain conditions. The method is applied to the case of a typical HF channel and a simplification is proposed, based on a per survivor equalizer, to be employed when selective fading is present. This solution, although increasing the per survivor cost, greatly reduces the number of states of the Viterbi decoder. Paper PDE.9 A MODIFIED FILTERBANK FOR TRACKING MULTIPLE SINUSOIDAL SIGNALS H.W. Sun, A. Yardim and G. D. Cain School of Electronic & Manufacturing Systems Engineering University of Westminster 115 New Cavendish Street London W1M 8JS, England Tel: +44 171 911 5083 Fax: +44 171 580 4319 e-mail: yardim@cmsa.westminster.ac.uk ABSTRACT A modified adaptive filterbank structure is presented to track multiple sinusoids which is based on the resonator-in-a-loop filterbank structure [1]. The advantages of this configuration over the previous resonator-in-a-loop filterbank structure are two-fold: its better enhanced transfer function characteristics, and the easier and more accurate determination of its signal-to-noise enhancement ratio. The simulation results using both the filterbank structures are presented and confirm the improved performance of the new filterbank. Paper PDE.10 WIGNER TRANSFORM INSTANTANEOUS PHASE ESTIMATOR Tomasz P. Zielinski Department of Instrumentation and Measurement Technical University AGH Al. Mickiewicza 30, 30-059 Krakow, Poland Tel: (+4812) 17 28 41; fax: (+4812) 17 39 72 e-mail: tzielin@uci.agh.edu.pl ABSTRACT. Computation of an instantaneous phase shift between two real-value signals by means of the Wigner transform is proposed. It is pointed out that the new method is about 37.5% faster than the Fourier transform one while having the similar dynamic accuracy and noise sensitivity for signals with high SNR. Paper PDE.11 SOURCE SEPARATION USING SECOND ORDER STATISTICS Ulf Lindgren, Henrik Sahlin and Holger Broman Department of Applied Electronics Chalmers University of Technology S-412 96 Gothenburg, Sweden E-mail:lindgren@ae.chalmers.se, salle@ae.chalmers.se, holger@ae.chalmers.se It is often assumed that blind separation of dynamically mixed sources can not be accomplished with second order statistics. In this paper it is shown that separation of dynamically mixed sources indeed can be performed using second order statistics only. Two approaches to achieve this separation are presented. The first approach is to use a new criterion, based on second order statistics. The criterion is used in order to derive a gradient based separation algorithm as well as a modified Newton separation algorithm. The uniqueness of the solution representing separation is also investigated. The other approach is to use System Identification. In this context system identifiability results are presented. Simulations using both the criterion based approach and a Recursive Prediction Error Method are also presented. Paper PDE.12 CONSTRAINED DECONVOLUTION: A GAME THEORY APPROACH IN AN H-inf SETTING Edgard SEKKO, Gerard THOMAS L.A.G.E.P. U.P.R.E.S.-A C.N.R.S. Q 5007 Universite Claude Bernard Lyon I et CPE- Lyon Bat 721, 43, Bd du 11 novembre 1918, 69622 Villeurbanne Cedex, FRANCE e-mail: sekko @lagep.univ-lyon1.fr In this paper we solve the constrained deconvolution problem by state space approach in an H-inf setting. The problem addressed is the design of a nonlinear estimator that guarantees H-inf performance on infinite horizon for the estimation error by using the Game Theory technic. The method proposed is useful in cases where the statistics of the disturbance and the noise signal are not completely known. We used the technic proposed to estimate heat production rate from the knowledge of the temperature. Paper PDE.13 Title APPLICATION OF THE STRUCTURED TOTAL LEAST NORM TECHNIQUE IN SPECTRAL ESTIMATION Authors Hua Chen, Sabine Van Huffel and Dirk van Ormondt Affiliation Electrical Engineering Department, Katholieke Universiteit Leuven, Belgium Tel: +32 16 321703 Fax: +32 16 321986 Email: sabine.vanhuffel@esat.kuleuven.ac.be Abstract In the problem of estimating parameters of exponentially damped sinusoids, an improved variant of Kung's method, called HTLS and based on the use of a Hankel data matrix, the singular value decomposition and the total least squares technique, has been proposed and proven to be accurate and efficient. In this paper, a more accurate estimator HTLN is proposed. It starts from the same Hankel data matrix, but uses a new technique, called structured total least norm, prior to the HTLS estimator. This technique computes the solution of a structured overdetermined linear system, AX=B, with possible errors in both A and B. The better accuracy of the STLN and HTLN techniques is shown by means of computer simulations. Paper PDE.14 A SHORT WAY TO COMPUTE WT FROM STFT Corneliu Rusu Technical University of Cluj-Napoca Str. Baritiu Nr.26-28, RO-3400 Cluj-Napoca, Romania Tel: +40 64 196285; Fax: +40 64 194831 e-mail: c.rusu@utcluj.ro ABSTRACT Wavelet transforms are often related to Fourier transforms due to their similitude and to underline some applications where wavelet theory should be superior over Fourier analysis. Many researchers have identified that the wavelet transform (WT) maps a function analogous to the short-time Fourier transform (STFT) that has a changing size. The increase numbers of FFT dedicated devices and of previous STFT databases request for a new approach between STFT and WT. This is one of the goals of the present paper. It also suggests a possible relation between the energetic representation of these transforms. The scalogram yields a graphical representation of the signal's energy distribute over the time-scale plane, as a spectrogram distributes the energy over the time-frequency plane. So, an interesting problem, the mapping between the spectrogram and the scalogram is derived. Few examples and some computational considerations are also provided. Paper PDE.15 SUBBAND DECOMPOSITION BASED ON THE HILBERT TRANSFORM APPLIED TO RADAR IMAGING S. Rouquette, Y. Berthoumieu, and M. Najim Equipe Signal et Image de l'ENSERB et GDR-134, CNRS BP 99, 33402 Talence Cedex, FRANCE Tel: +33 56846140; fax: +33 56848406 e-mail: steph@goelette.tsi.u-bordeaux.fr In this paper, we propose an approach to improve high-resolution frequency estimation for narrow-band planes. This approach is based on a signal preprocessing combined with a high-resolution method to increase the accuracy of frequency estimation. The preprocessing step is a Subband Decomposition Based on the Hilbert Transform (SDBHT) [1] for one and two-dimensional signals. This improvement is achieved by using an empirical criterion to determine the number of waves of the signals derived from the SDBHT technique. Simulation examples show the performances of this criterion. Then, we apply SDBHT method and empirical criterion to radar imaging. Paper PFT.1 DAMPED SINUSOIDAL SIGNAL RECONSTRUCTION USING HIGHER-ORDER CORRELATIONS Diego P. Ruiz, Mara C. Carrin, Antolino Gallego. Dpto. Fsica Aplicada, Facultad de Ciencias, Universidad de Granada, 18071 Granada, Spain. Tel/Fax: +58-243229/+58-243214, E-mail: druiz@goliat.ugr.es Abstract In this paper the reconstruction of deterministic damped sinusoidal signals from a one-dimensional slice of their multiple correlations is analyzed. Signal correlations are estimated using a new higher-order correlation estimator, which allows the exponentially damped structure of the signal to be maintained in any horizontal slice of correlations. This characteristic is of utmost importance for the subsequent application of a linear method to estimate the signal parameters and thus reconstruct the signal. Simulations results show that the correlation-based approach gives better reconstructed signals than data-based methods (KT method) when colored noise contaminates the signal. Paper PFT.2 THE SYNTHESIS OF A HIGH ORDER DIGITAL BANDPASS FILTERS WITH TUNABLE CENTRE FREQUENCY AND BANDWIDTH ) A lexander A . Petrovsky Belorussian State University of Informatics and Radioelectronics 6, P.Brovky Str., 220027, Minsk, Republic of Belarus Tel: +375 172 2312910; fax: +375 172 2310914 e-mail: palex@micro.rei.minsk.by ABSTRACT In this paper. described tunable digital bandpass filters whereby the centre frequency and bandwidth can be independently related to the multiplier coefficients, which permit simple frequency response adjustment by varying the coefficients values. The bandpass filters proposed here have a cascade form and are composed of several second-order recursive bandpass sections with identical characteristics. The methods for the direct computation of the number of second-order filters in the cascade form, adjustable parameters and designing filter bank are shown in this paper. The design equations strate the true parametric tuning ability of the circuit. By cascading a few such circuits, a complete parametrically adjustable digital frequency responsed equalizer may be realized. It does not require precomputing the multiplier coefficient values for all designed equalizer settings. Paper PFT.3 HIGHER-ORDER STATISTICS FOR QAM SIGNALS: A COMPARISON BETWEEN CYCLIC AND STATIONARY REPRESENTATIONS Pierre Marchand and Denys Boiteau CEPHAG-ENSIEG URA CNRS 346, BP 46, 38402 Saint Martin d'Hres Cedex, France TEAMLOG, "Le Grand Sablon", 4 av. de l'Obiou, 38700 La Tronche, France CESTA, 37 av. du Gnral de Gaulle, 35170 Bruz, France For a cyclostationary signal, the cumulant-based cyclic tricorrelation (fourth-order correlation) at cycle frequency zero should not be confused, in the general case, with the cumulant-based tricorrelation of the same signal after stationarization. The reasons for this unusual assertion are detailed; as an illustration, we show that if QAM signal classification is impossible using their fourth-order cyclic statistics, classification is however possible if a stationary modelling is adopted. Remarks on the estimation of both cyclic and stationary temporal cumulants are provided and consequently, the skip between the cyclic and the stationary models is enlightened. Theoretical expressions of cyclic and stationary tricorrelations are derived and computer simulations confirm the results. Paper PFT.4 MODULATED FILTER BANKS : A FOLDING APPROACH Mohamed Gharbi*, Frederic Nicot+, Marc Georges Gazalet* and Francois-Xavier Coudoux* *: Institut d'Electronique et de Microelectronique du Nord U.M.R C.N.R.S. 9929 Universite de Valenciennes et du Hainaut Cambresis BP 311 Le Mont Houy 59304 Valenciennes Cedex (France) e-mail : gharbi@univ-valenciennes.fr +: NORTEK, 21, rue Elisee Reclus, 59650 Villeneuve d'Ascq (France) Cosine Modulated Filter Banks (MFB) have been widely studied [MAL92, KOI92, MAU94] and are successfully used in signal and image processing. A perfect reconstruction factorization of filter banks based on cosine modulation of a linear phase prototype filter of length L=2KM has been proposed in [MAL92, KOI92]. This factorization leads to paraunitary MFB where the analysis and synthesis filters are the same. In this paper, with the use of the discrete folding operator introduced in [AUC92], we extend this factorization to the L=NM case with a more general modulation matrix. If N is even, the MFB is either paraunitary or biorthogonal. While if N is odd, the MFB is biorthogonal. Paper PFT.5 CORRECTION OF RESIDUAL PHASE DISTORTIONS IN SEISMIC DATA Sabeur Mansar* and Fransois Glangeaud** * TOTAL, TEP/DE/CST/RTS, 78470 Saint Remy les Chevreuses, France **CEPHAG/ENSIEG. BP 46. 38402 Saint-Martin-d'Heres Cedex. France ABSTRACT The residual wavelet on a processed seismic section is often not zero phase despite all efforts to make it so. Phase distortions arise for a variety of reasons. In this paper we deal with phase distortions which arise during the seismic processing. Constant phase rotation have been used to try to correct phase distortions. We present here two new methods for phase distortion correction. The first is based on Higher Order Statistics and can handle frequency-dependent phase distortions. The second is based on the Continuous Wavelet Transform and can handle time-varying frequency-dependent phase distortions. The application of the two methods on synthetic and real traces has shown their efficiency. Paper PFT.6 AN ALGORITHM FOR ROBUST STABILITY OF DISCRETE SYSTEMS M. BARRET (1) and M. BENIDIR (2) (1) SUPELEC, Campus de Metz, 2 rue E. Belin, 57070 Metz, France Tel: (33) 87 74 99 38, Fax: (33) 87 76 95 49, e-mail: Michel.Barret@supelec.fr (2) Universite Paris-Sud, L2S-ESE, Plateau de Moulon, 91192 Gif-sur-Yvette, France Tel : (33) 1 69 85 17 17, Fax: (33) 1 69 85 12 34, e-mail: benidir@lss.supelec.fr In many applications, digital recursive filter coefficients have no distinct values, therefore the test of an entire family of polynomials is required in order to be sure of the filter stability. The edge theorem by Bartlett, Hollot and Lin states that a polytope is stable, if and only if, the exposed edges are stable. In this paper, this last condition is transformed into an equivalent one, that can be tested in a finite number of arithmetic operations and from which an algorithm is derived. It is shown that the condition, which has been established, is optimum i.e., it can neither be avoided, nor simplified. Paper PFT.7 GENERALIZED TIME-FREQUENCY DISTRIBUTIONS AND APPLICATIONS M. BENIDIR and A. OULDALI Universite Paris-Sud, L2S-Supelec, Plateau de Moulon, 91192 Gif-sur-Yvette, France Tel : (33) 1 69 85 17 17, Fax: (33) 1 69 85 12 34, e-mail: benidir@lss.supelec.fr A decomposition of the derivatives of order k of a polynomial is proposed in terms of the translated versions of the polynomial. This result allows us to introduce generalized time-frequency distributions for studying polynomial phase signals with constant amplitude in order to determine the degree and the coefficients of the corresponding phase. Relationships between these distributions and the already known polynomial distributions are established. Statistical properties of the proposed distributions are studied and their application for estimating the instantaneous frequencies in multiple chirp signals are discussed. Paper PFT.8 THE PROPERTIES OF FLOATING POINT SINGLE QUANTIZATION INCLUDING UNDER- AND OVERFLOW F. Hartwig and A. Lacroix University of Frankfurt, Institute of Applied Physics D - 60325 Frankfurt am Main, Robert - Mayer - Strae 2-4 ABSTRACT The quantization of floating point numbers is well investigated for situations where no under- or overflow occurs [1-3]. In this paper results are presented including these cases for the quantization of uniform, gaussian and sinusoidal distributed numbers. For underflow two different cases are considered: (1) no unnormalized mantissas occur and numbers whith magnitudes lower than a certain limit are set to zero, (2) unnormalized mantissas are used in the underflow region which leads to a behaviour similar to that of fixed point quantization. It can be seen that different slopes of the SNR vs. S curves in the underflow regions characterize the utilization of normalized or unnormalized mantissas. In the overflow-region it is assumed that saturation is utilized, which means that numbers with magnitude greater than a certain limit are set to fixed overflow values. Paper PFT.9 IDENTIFICATION AND PREDICTION OF NONLINEAR SYSTEMS USING ORTHONORMAL FUNCTIONS Iain Scott, Bernard Mulgrew Department of Electrical Engineering, The University of Edinburgh, The King's Buildings, Mayfield Road, Edinburgh EH9 3JL, Scotland. Tel: +44-131-650 5565, Fax: +44-131-650 6554, E-mail: is@ee.ed.ac.uk Abstract In a recent paper Mulgrew [1] proposed a nonlinear filtering structure which utilises a set of orthonormal expansions to model nonlinear dynamical systems. Provisional results were presented for a simple 1--dimensional system. In this paper we extend the analysis of this structure to multi--dimensional filtering, and examine the application of the orthonormal structure for nonlinear system identification and communications channel equalisation. The link between the choice of Fourier basis functions and popular kernel probability density estimation techniques is examined. Paper PFT.10 title: A NEW LEAST SQUARES-BASED APPROACH FOR FAST LEARNING IN RECURRENT NEURAL NETWORKS} authors: R.Parisi, E.D.Di Claudio, A.Rapagnetta and G.Orlandi affiliation: INFOCOM Dept.- University of Rome La Sapienza, via Eudossiana 18, 00184, Rome, Italy email: parisi@infocom.ing.uniroma1.it abstract: In this paper a new approach to learning in recurrent neural networks is presented. The method proposed is based on the descent of the error functional in the space of the linear part of the neurons (neuron space approach). A linear system is solved for the weights following a Recursive Least Squares criterion at each step of the learning process. This approach, w.r.t. traditional gradient-based algorithms, guarantees better performances from the point of view of the speed of convergence and the numerical robustness. Paper PFT.11 ON THE DETERMINATION OF THE OPTIMAL CENTER AND SCALE FACTOR FOR TRUNCATED HERMITE SERIES T. Oliveira e Silva and H. J. W. Belt Universidade de Aveiro / INESC Aveiro 3810 AVEIRO PORTUGAL tos@inesca.pt and Eindhoven University of Technology The Netherlands H.J.W.Belt@ele.tue.nl Signals that are fairly concentrated in time or space can often be conveniently described by a truncated Hermite series. The rate of convergence of such series depends on the center and scale factor of the Hermite functions. In this paper we present some results concerning the determination of the optimal values of these two important parameters. We address the problem of the approximation of one-dimensional functions defined in the continuous interval$(-\infty,\infty)$. Paper PFT.12 On the approximation of nonbandlimited signals by nonuniform sampling series Paulo Jorge S. G. Ferreira Departamento de Electronica e Telecomunicacoes / INESC Universidade de Aveiro 3810 Aveiro Portugal E-mail: pjf@inesca.pt The classical WKS sampling theorem is a central result in signal processing, but it applies to band-limited signals only. For many purposes, this class of signals is too narrow. For example, the signals that occur in practice are invariably of finite duration, or time-limited, and often have discontinuities. Clearly, such signals cannot be band-limited. We consider the problem of approximating such signals, or other signals not necessarily band-limited, using sampling series. We do not assume that the sampling instants are regularly distributed, in order to account for errors due to jitter. To the best of our knowledge, the problem of obtaining nonuniform sampling approximations for signals not necessarily band-limited, despite its practical interest, has not been addressed in the literature. In this work we introduce a method that leads to sampling approximations with the required properties. It is shown that the sampling sums considered are capable of approximating a wide class of signals, with arbitrarily small L-2 and L-infinity errors. Paper PFT.13 A Running Walsh-Hadamard Transform Algorithm and Its Application to Isotropic Quadratic Filter Implementation G. Deng and A. Ling School of Electronic Engineering, La Trobe University Bundoora, Victoria 3083 Australia Phone: +61 3 479 2036; Fax: +61 3 471 0524 Email: d.deng@ee.latrobe.edu.au Abstract Two problems associated with adaptive isotropic quadratic filters are the computational complexity and the speed of convergence. This paper presents a transform domain implementation scheme to solve these problems. A new implementation of the filter using the Walsh-Hadamard transform (WHT) is described. A running WHT (RWHT) algorithm is also proposed to reduce the computational cost. Theoretical analysis shows that the number of operations of the WHT implementation (using the RWHT) is considerably less than that of the direct implementation. The advantage of using the WHT implementation is illustrated by modelling a real nonlinear system. Results show that the WHT implementation converges significantly faster than the direct implementation. Paper PI.1 A/D CONVERSION WITH FUZZY MEMBERSHIP FUNCTION Giuseppe Di Cataldo (1), Valentino Liberali (2), Franco Maloberti (2), and Gaetano Palumbo (1) (1) Dipartimento Elettrico Elettronico e Sistemistico (2) Dipartimento di Elettronica Universita di Catania Universita di Pavia Viale Andrea Doria 6 Via Ferrata 1 95125 Catania, Italy 27100 Pavia, Italy Phone +39.95.339535 Phone +39.382.505205 Fax +39.95.330793 Fax +39.382.505677 E-mail: gdicata@ns2.cdc.unict.it, E-mail: valent@ipvsp4.unipv.it, gpalumbo @ ns2.cdc .unict.it franco @ franco.unipv.it ABSTRACT This paper presents a general architecture of an A/D converter whose input-output transfer characteristic has the shape of a typical fuzzy membership function. Indeed, the proposed A/D converter performs a fuzzification operation from input to output through a programmable trapezoidal function. The proposed architecture requires a conventional A/D converter, a comparator, some inverters and switches, thus allowing to save silicon area while maintaining good flexibility and programmability. Moreover, it does not depend on the converter architecture and can be applied to every A/D converter. Paper PI.2 TITLE : MEMORY ASPECTS IN SIGNAL PROCESSING AND HLS TOOL : SOME RESULTS AUTHORS : J.L.Philippe, D.Chillet, O.Sentieys, J.P.Diguet AFFILIATION : LASTI-ENSSAT 6 rue de kerampont 22300 LANNION FRANCE eMail : philippe@enssat.fr ABSTRACT : The digital signal processing system design consists in four synthesis phases which concern the processing, the control, the memory and the communication units. Today, many tools enables us to produce the processing unit. However, in many applications, the hardware solution may be challenged by the number and complexity of memories. This paper proposes a design methodology of the memory units for algorithms restricted by a real time constraint. The original nature of our approach, is due to the fact that it proposes a global memory solution for a transfer sequence computed by the synthesis tools, like GAUT. (CAD Tools, High Level Synthesis, ASIC Design, Digital Signal Processing, Memory Synthesis) Paper PI.3 A 650 MHz Pipelined MAC for DSP Applications using a New Clocking Strategy F. Fraternali, G. Masera, G. Piccinini, M. Zamboni Politecnico di Torino - Dipartimento di Elettronica Corso Duca degli Abruzzi 24 - I10129 TORINO - Italy A 8x8 bit multiplier and accumulator unit for high speed applications is presented in this paper. The multiplier archlitecture is directly derived from the Baugh and Wooley algorithm, with some modifications, to reduce area and latency while the accumulator section is distributed along the multiplier structure. In this way the accumulator's latency is hidden in the multiplier's one. A new clocking strategy ha.s been used for the design of the four stages pipelined a.ccumulator cell, based on a full adder with partial feedback. The unit is synthesized in a 0.7 micron N well CMOS technology. A one phase dynamic logic (True Single Pha.se Clocking- TSPC) has been adopted and the transistors widths had been sized by using an optimization algorithm achieving a clock frequency of 650 MHz with a latency of 36 clock cycles. Paper PI.4 DESIGN OF A FAST AND AREA EFFICIENT FILTER Pontus strm, Peter Nilsson and Mats Torkelson Dept. of Applied Electronics, University of Lund, Box 118, 22100 Lund, Sweden e-mail: Pontus.Astrom@tde.lth.se This paper shows how to optimize full custom, fixed coefficient filters to gain both in area and speed. The idea is to trade the filter order with the coefficient length and thus reduce the delay and the size of the multipliers. The result is a smaller design that needs fewer clock cycles per sample compared to a minimum order filter. Measurements on the manufactured chips verified a speed gain of 26% and a size reduction of 20%. Paper PI.5 A Real Time ISO/MPEG2 Multichannel Encoder C. Costantini, G. Parladori, M. Stanzani Alcatel Corporate Research Centre Via Trento, 30, 20059 Vimercate (MI), Italy Tel: +39 39 686.4976; fax: +39 39 686.3587 e-mail: ccostantini@tlt.alcatel.it ABSTRACT We describe the characteristics of a real time MPEG2 Multichannel Encoder based on a multi-DSP board. We discuss both the architectural base and the algorithm real time implementation. Due to the flexibility of the described card, realisations of other audio processing algorithms are also possible. A brief description of these implementations is also given. Paper PI.6 PROGRAMMABLE BIT-SERIAL REED-SOLOMON ENCODERS S.T.J.Fenn, M. Benaissa, D.Taylor & J. Luty School of Engineering, The University of Huddersfield, Queensgate, Huddersfield, HD1 3DH, U.K. e-mail: S.T.J.Fenn@hud.ac.uk ABSTRACT In this paper the design of programmable bit-serial Reed-Solomon encoders is considered using the traditional Berlekamp multiplier. It is suggested that there are certain advantages to be gained by deriving the generator polynomial of the code using combinational logic, or equivalently using look-up tables, rather than using an iterative LFSR based approach. The use of the recently proposed Berlekamp-like bit-serial multiplier is also considered and shown to demonstrate a number of potential advantages over the traditional Berlekamp multiplier in Reed-Solomon encoders. Paper PI.7 RATIONAL APPROXIMANT ARCHITECTURE FOR NEURAL NETWORKS F.M. Frattale Mascioli & G. Martinelli Dip. INFO-COM, Universit di Roma "La Sapienza" via Eudossiana, 18 - 00155 Roma - Italy Tel: +39 6 44585488/9; fax: +39 6 4873300 e-mail: mascioli@infocom.ing.uniroma1.it ABSTRACT A novel approach is proposed for overcoming the multiple minima problem, present in the learning of a supervised neural network. It allows to connect rational function approximations to neural networks and is based on the use of a truncated Fourier expansion for determining: 1) the architecture; 2) the parameters of the net, avoiding local minima in an efficient way. Paper PI.8 A NEW STRUCTURE FOR VIDEO-RATE 2D SC FIR FILTERS G.M. Cortelazzo, E. Malavasi, A. Gerosa, A. Neviani, A. Baschirotto t Dipartimento di Elettronica ed Informatica, Univerista di Padova via Gradenigo 6/A, 35131 Padova, Italy Tel: +39 49 8277827; fax: +39 49 8277699 e-mail: corte@dei .unipd. it t Dipartimento di Elettronica, Universita di Pavia via Ferrata 1, 27100 Pavia, Italy ABSTRACT Since Switched Capacitor (SC) circuits operate with discrete-time analog signals, it becomes rather attractive revisiting traditional video operations in the attempt of replacing systems currently implemented digitally with SC circuits, wherever possible. Indeed substituting the digital part for an analog one leads to substantial improvements with respect to power and area characteristics . This kind of circuit variations poses a number of challenging problems mainly related to the fact that the circuit clocks are at video-rate. This work describes possible solutions to these problems exemplified by the SC realisation, by standard monolithic CMOS technology of a 2-D low-pass filter designed for picture-in-picture (PIP) resizing. Paper PI.9 PARALLEL IMPLEMENTATION OF IMAGE CODING USING WAVELET TRANSFORM: SYNDEX SOFTWARE ENVIRONMENT APPLICATION Christophe Cudel+, Bertrand Vigouroux++ +LAM, Equipe Image LTI - IUT de TROYES - BP 396 - 10026 TROYES CEDEX - FRANCE e-mail: cudel@altern.com ++IUT d'ANGERS BP 2018 - 49016 ANGERS CEDEX - FRANCE e-mail: bertrand.vigouroux@univ-angers.fr ABSTRACT: This work is a contribution to Adequation between Algorithm and Architecture. It presents an example of application made with SynDEx, a software environment to implement signal processing or automatic algorithms on multi-processor network. This communication shows that a Conditionned Data Flow Graph used for modelising an algorithm, is enough to do an implementation on multi-processeur network. Paper PI.10 GFLOPS COMPUTER: AN IMAGE PROCESSING PARALLEL ARCHITECTURE Dominique Houzet, Abdelkrim Fatni IRIT-ENSEEIHT-INP, 2 rue Camichel, 31071 Toulouse, France Tel: +33 6158 83 18; fax: +33 61 58 82 09 houzet@enseeiht.fr Abstract The real-time Image and signal processing applications, such as vision, image synthesis, HDTV, signal processing, neural networks, require both computing and input/output power. The GFLOPS project is dedicated to the study of all the aspects concerning the design of such computers. Its aim is to develop a parallel architecture as well as its software environment to implement those applications efficiently. The proposed architecture supports up to 512 processor nodes, connected over a scalable and cost-effective network at a constant cost per node. GFLOPS-2 is a single-user machine which is designed to be used as a low-cost parallel co-processor board in a desk-top work station. Paper PI.11 A NOVEL SORTING ALGORITHM AND ITS APPLICATION TO A GAMMA-RAY TELESCOPE ASYNCHRONOUS DATA ACQUISITION SYSTEM Alberto Colavita(*), Enzo Mumolo(**), Gabriele Capello(**) (*) Microprocessor Laboratory, ICTP-INFN, Via Beirut 31, 34100 Trieste, Italy (**) Dipartimento di Elettrotecnica, Elettronica ed Informatica DEEI, Universita' di Trieste, Via Valerio 10, 34127 Trieste, Italy Tel/Fax: +39.40.676.3861/3460 E-mail: mumolo@univ.trieste.it Abstract In this paper we present a novel parallel sorting algorithm, highly suited for VLSI implementation, which works through a cascade of elementary sorting units and leads to a scalable architecture. The paper describes the applications of such device to the asynchronous data acquisition for a gamma ray telescope. Paper PI.12 A FAST ALGORITHM FOR MORPHOLOGICAL EROSION AND DILATION C. Jeremy Pye and J. A. Bangham School of Information Systems, University of East Anglia, Norwich, NR4 7TJ, United Kingdom. Tel/Fax: +44 0 1603 456161/453345 Email: cjp@sys.uea.ac.uk, ab@sys.uea.ac.uk This paper describes a new algorithm for performing erosion and dilation which is suitable for flat line-segment structuring functions, and which has a computational complexity that is independent of the structuring function size. Unlike other proposed algorithms, the computation time required by this method is directly proportional to the number of extrema within the signal being processed. This makes it particularly suitable for signals and images that have large and slowly varying segments. Paper PI.13 COMBINED BLOCK CODE AND DIVERSITY OVER A RAYLEIGH FADING CHANNEL WITH SOFT DECISION DECODING Authors: S.B.Hashimi and R.A.Carrasco Electronic Group, School of Engineering, PO.Box 333 Staffordshire University, Beaconside, Stafford ST18 0DF, (U.K) email: b.hashim@staffs.ac.uk r.carras@staffs.ac.uk ABSTRACT Three new Block Code Diversity combing schemes for digital transmission over a Rayleigh fading channel have been proposed. It has been shown that improved performance in term of probability of error versus signal to noise ratio has been obtained using soft-decision Viterbi decoding together with diversity techniques. Simulation results have been included to verify the performance of the proposed schemes. Paper PI.14 Title HERMES : AN OBJECT-ORIENTED MULTITASKING SYSTEM FOR CONCURRENT DIGITAL SIGNAL PROCESSING APPLICATIONS Authors Antonios Anagnostopoulos and Georgios Kouroupetroglou Affiliation Division of Communication and Signal Processing, Department of Informatics, University of Athens, Athens GR 15784, Greece e-mail: koupe@di.uoa.gr Abstract This paper presents the design and implementation of the PC-based multitasking system HERMES, which supports the development of concurrent Digital Signal Processing (DSP) applications using object-oriented programming techniques under the MS-DOS operating system. The signal abstractions by Objects of the HERMES system and its architecture are described along with the software framework that enables the realization of highly reusable and modular code for rapid production of DSP tools and applications that can cooperate in real-time. Paper PIC.1 A Mathematical Model for Coding Hand-drawn Letters Masaru KAMADA, KenYa YONEZAWA and Yuji ABE Department of Computer and Information Sciences, Ibaraki University, Hitachi, Ibaraki 316 Japan e-mail: kamada@cis.ibaraki.ac.jp A Japanese letter can be regarded as a collection of short strokes related to each other by the movement of pen-tip in the air. This dynamical relationship is an important factor in the impression of hand-drawing. A three-dimensional model of drawing letters by hand is proposed in this paper based on the principle to minimize the variation of force operating on the tip subject to certain constraints. The solution is a mixture of fifth and fourth degree splines in the horizontal direction, and a trigonometric series modulated by linear functions under the paper surface or a fifth degree polynomial in the air in the vertical direction. Several examples of approiximation of real letters by this function are presented. Paper PIC.2 COMPARISON OF MEAN-SQUARE AND ABSOLUTE VALUE DISTORTION MEASURES IN FRACTAL CODING OF STILL IMAGES F.C. Cesbron and F.J. Malassenet Georgia Tech Lorraine The European Platform of the Georgia Institute of Technology 2-3, rue Marconi F-57070 Metz, France Tel: (33) 87 20 39 39; fax: (33) 87 20 39 40 e-mail:fcesbron@georgiatech-metz.fr, fjm@georgiatech-metz.fr Fractal coding often blurs or smoothes images. In particular, edges are poorly coded due to the choice of the distortion measure. Indeed, mean square error possesses no edge preserving property. The solution proposed in this article is to code using fractals and to introduce the L_1 norm or the absolute value distortion measure. The visual quality of the coded images with the L_1 norm is improved. However, the computational complexity of the coder is drastically increased. It may be kept low by preprocessing using an edge detection scheme to select the pertinent measure. The reconstruction algorithm remains the same. Paper PIC.3 FRACTAL CODING OF IMAGE SEQUENCE USING EXTENDED CIRCULAR PREDICTION MAPPING Chang-Su Kim, Rin-Chul Kim and Sang-Uk Lee School of Electrical Engineering, Seoul National University e-mail: cskim@claudia.snu.ac.kr This paper proposes a novel algorithm for fractal coding of color image sequence, based on the extended CPM (Circular Prediction Mapping). In the extended CPM, each range block is approximated by a domain block in the adjacent frame, which is of the same size as the range block. Therefore the proposed domain-range mapping is similar to the block matching algorithm in the motion compensation techniques, and we can exploit the temporal correlation in moving image sequence effectively. Also we show that fast decoding is possible, since the decoder requires about 1 multiplication and 3 additions per pixel for each Y, U, V components. The computer simulation results on real image sequences demonstrate that the proposed algorithm provides very promising performance at low bit-rate. Paper PIC.4 EMBEDDED ZERO-TREE CODING OF IMAGES EMPLOYING SOFT-THRESHOLDING Frank Mueller and Klaus Illgner Institut fuer Elektrische Nachrichtentechnik Aachen University of Technology (RWTH), 52056 Aachen, Germany e-mail: {mueller,illgner}@ient.rwth-aachen.de Improvements of embedded zero-tree wavelet (EZW) coding by employment of soft-thresholding in the wavelet domain are reported. By proper adjustment of the thresholds, the attainable PSNR can be slightly improved (around 0.3 dB). Moreover, soft-thresholding can improve the visual appearance of the coded images, especially in the very low data rate case. The quality in the early stages of progressive image transmission can be improved by reduction of annoying artifacts resulting from coarse quantization of the wavelet coefficients. Adaptation of the thresholds to the "current" width of the quantization intervals preserves the embedded bit stream property of the coder. Paper PIC.5 EMBEDDED IMAGE CODING BASED ON LAPLACIAN PYRAMIDS WITH QUANTIZATION FEEDBACK Bruno Aiazzi, Stefano Baronti, Franco Lotti Nello Carrara Research Institute on Electromagnetic Waves IROE - CNR Via Panciatichi, 64 - 50127 Florence, ITALY e-mail: baronti@iroe.fi.cnr.it Luciano Alparone Department of Electronic Engineering, University of Florence Via S. Marta, 3 - 50139 Florence, ITALY e-mail: alparone@cosimo.ing.unifi.it ABSTRACT In this paper, a multi-layer SNR-scalable error-bounded image encoder is achieved in the framework of Laplacian pyramids with quantization noise feedback, by exploiting an entropy-minimizing optimum quantization strategy, a content-driven decision rule based on an L-infinity activity measure, and multistage quantizers to progressively upgrade quality at full scale. The resulting scheme yields intermediate versions with scale and SNR both increasing, and a further SNR scalability on the full resolution, with possibly lossless reconstruction, thereby expediting interactive browsing of remote data bases of images of any sizes and wordlength. The proposed encoder outperforms JPEG which does not possess all the above mentioned attractive characteristics. Paper PIC.6 ERROR DETECTION AND CONCEALMENT IN JPEG IMAGES Mourad ABDAT, Ziad ALKACHOUH and Maurice G. BELLANGER CNAM, 292 rue Saint-martin,75141 PARIS CEDEX 03, FRANCE Tel. : +33 1 40 27 20 82, Fax. : +33 1 40 27 27 79 e-mail: abdat@cnam.fr Even with use of restart intervals, some residual errors remain in the decoded JPEG images after transmission. In order to improve the image quality, robust decoding techniques are useful. First, we propose error detection techniques, then error compensation and concealment techniques for the damaged blocks. Depending on the entropy coding and on the neighbourhood template, improvements between 3 and 7 dB in terms of Peak-to-peak Signal-to-Noise Ratio (PSNR) are provided by robust decoders with respect to conventionnal JPEG decoders, under bit error rates around and less than 10^-4. Paper PIC.7 DUAL SET ARITHMETIC CODING AND ITS APPLICATIONS TO IMAGE CODING Bin Zhu, Enhui Yang, and Ahmed H. Tewfik Department of Electrical Engineering, University of Minnesota Minneapolis, MN 55455, USA email: binzhu, ehyang, tewfik@ee.umn.edu Arithmetic coding is usually implemented in fixed precision. Such an implementation cannot efficiently code sources, such as image coding algorithms, that locally produce a small fraction of a large alphabet of symbols. In this paper, we propose a novel approach to overcome this inefficiency. The proposed algorithm uses dual symbol sets: a primary symbol set that contains the symbols that have occurred in the recent past and a secondary symbol set that contains all other symbols. Both sets are dynamically adapted to the local statistics. We summarize an analysis of the proposed approach and describe the results that we have obtained by applying it to images. Paper PIC.8 ON THE DESIGN OF AN IMAGE COMPRESSION SCHEME BASED UPON A PRIORI KNOWLEDGE ABOUT IMAGING SYSTEM AND IMAGE STATISTICS C.H. Slump, F.J. de Bruijn, P.J.A. Hagendoorn University of Twente, Dept. of Electrical Engineering Lab. for Network Theory, P.O. Box 217 7500 AE Enschede, the Netherlands tel.:+31 53-4892094 fax.:+31 53-4891060 e-mail: c.h.slump@el.utwente.nl ABSTRACT This contribution is about the design of an image compression scheme for near loss-less image compression of a restricted class of images and a specific application. The images are digital diagnostic X-ray images of the coronary vessels of the human heart. This paper proposes a novel compression scheme with a compression ratio of 8 - 10 with preservation of the diagnostic image quality. Central in our approach is the amount of information a trained and highly skilled observer i.e. the cardiologist is able discern at a given exposure and thus quantum noise level. The physics of the image detection process together with the a priori knowledge of the imaging system are the basis of the image statistics. Relevant elements of the human visual system complete the stochastic characterization of imaging process whereon the compression scheme is based. Paper PIC.9 OBJECT-SCALABLE DYNAMIC CODING OF VISUAL INFORMATION Corinne Le Buhan, Emmanuel Reusens and Touradj Ebrahimi Signal Processing Laboratory Swiss Federal Institute of Technology CH-1015 Lausanne, Switzerland e-mail: lebuhan@ltssg4.epfl.ch This paper describes an extension of a dynamic video coding scheme to provide object scalable functionalities. As a particular instance of the dynamic coding concept, the coding scheme considered here jointly optimizes video data partition and representation modes. Indeed, as there exists no universal video coding method, dynamic coding insures the choice of the most efficient technique (for instance DCT, fractal or motion compensation) for each data segment. These data segments are themselves optimally partitioned within the original frame (respectively region of interest). An optimization algorithm achieves this joint data partition/representation modes selection to yield the best rate/distortion compromise within the available set of possible solutions, under a rate or distortion constraint. Such a dynamic coding algorithm designed for low bitrates was proposed to MPEG-4 first set of tests in November 1995. This paper describes the corresponding object scalable coding scheme. Paper PIC.10 A NEW CONTOUR SIMPLIFICATION FILTER FOR REGION-BASED CODING V. A. Christopoulos, C. A. Christopoulos*, J. Cornelis, A. N. Skodras** Vrije Universiteit Brussel, VUB-ETRO (IRIS), Pleinlaan 2, 1050 Brussels, Belgium *Ericsson Telecom AB, HF/ETX/MN, S-126 25 Stockholm, Sweden **University of Patras, Electronics Laboratory, Patras 26110, Greece Tel: +32 2 6292982 fax: +32 2 6292883 e-mail: vschrist@etro.vub.ac.be In region-based (RB) coding, the image is divided into a number of various-shaped regions, which are then treated as objects and coded. Experimental results show that a bottle-neck in RB coding at very low bitrates is the amount of shape information, represented by the contours separating the regions, which has to be coded. This paper presents a new filter for contour simplification. The filter reduces the number of contour points by an average of more than 30%. A simplification example shows that its use does not affect subjective image quality. Paper PIC.11 INTERBLOCK REDUNDANCY REDUCTION USING QUADTREES Marcos Faundez Zanuy, Xavier Domingo Reguant Department of Signal Theory and Communications (UPC) e-mail: marcos@gps.tsc.upc.es This paper applies the quadtree structure for image coding. The goal is to adapt the block size and thus to increase the compression ratio (without reducing SNR). Also, the computational time is not significatively increased. It has been applied to Block Truncation Coding of still images, and motion vector coding (interframe). An inter/intraframe application is also discussed. We propose a method based on block compression with small block size, and the clustering of blocks whenever they represent the same information. Paper PIC.12 VECTOR QUANTIZATION CLUSTERING USING LATTICE GROWING SEARCH Dorin Comaniciu Caip Center, Rutgers University Frelinghuysen Rd., P.O.Box 1390 Piscataway, NJ 08855-1390 USA e-mail: comanici@caip.rutgers.edu Cristina Comaniciu Dept. of Applied Electronics Polytechnic University of Bucharest 313 Spl. Independentei, 77206 Bucharest Romania e-mail: ccoman@pcnet.pcnet.ro ABSTRACT: In this paper we introduce a non-iterative algorithm for vector quantization clustering based on the efficient search for the two clusters whose merging gives the minimum distortion increase. The search is performed within the K-dimensional cells of a lattice having a generating matrix that changes from one step of the algorithm to another. The generating matrix is modified gradually so that the lattice cells grow in volume, allowing the search of the two closest clusters in an enlarged neighborhood. We call this algorithm Lattice Growing Search (LGS) clustering. Preliminary results on 512 x 512 images encoded at 0.5 bits/pixel showed that the LGS technique can produce codebooks of similar quality in less than 1/10 of the time required by the LBG algorithm. Paper PIC.13 ON THE SIZES OF VORONOI CELLS IN ENTROPY-CONSTRAINED VECTOR QUANTIZATION Stephan F. Simon Institut f. Elektrische Nachrichtentechnik Rheinisch-Westf. Technische Hochschule (RWTH) Aachen, 52056 Aachen, Germany Tel: +49-241-807677, Fax: +49-241-8888196 E-mail: simon@ient.rwth-aachen.de ABSTRACT Voronoi cells for vector quantization subject to an entropy constraint are considered. It is shown that the constraint on the output entropy leads to a weaker and even vanishing dependency of the Voronoi cell's volume on the probability density function. Using some simplifying assumptions like linearization of a small part of the n-dimensional input space and modeling of the cell shapes as hyperspheres leads to an analytic expression of the quotient of the volumes of two neighboring Voronoi cells. The results confirm the use of entropy coded lattice vector quantizers with optimized reproduction vectors in cases of vanishing dependency and may in other cases be exploited for the design of vector companders to be used in conjunction with lattice vector quantization. Paper PIC.14 GENERALIZED GAIN-SHAPE VECTOR QUANTIZATION FOR MULTISPECTRAL IMAGE CODING Gerardo R. Canta, Luigi Paura, Giovanni Poggi Dipartimento di Ingegneria Elettronica Universit\a di Napoli via Claudio 21, 80125 Napoli, Italy Tel: +39 81 7683151 Fax: +39 81 7683149 E-mail: poggi@nadis.dis.unina.it This paper proposes a new encoding scheme that generalizes the Gain-Shape Vector Quantization technique and takes advantage of the distinctive features of multispectral images to encode them at very low bit rate, with a satisfactory reproduction quality and low complexity. Each codevector is obtained as the Kronecker product of a gain codevector and a shape codevector, which reduces both the memory requirements and codebook design complexity. Besides, the encoding complexity is also greatly reduced by resorting to a fast encoding algorithm. Paper PIC.15 Region Based KLT for Multispectral Image Compression Gabriel Fernandez and Craig M. Wittenbrink(*) Signal Processing Laboratory Swiss Federal Institute of Technology CH-1015 Lausanne, Switzerland fernandez@lts.epfl.ch (*)Computer Engineering & Information Sciences University of California, Santa Cruz Santa Cruz, CA 95064, USA craig@cse.ucsc.edu In this paper we present a new approach of spectral decorrelation for multispectral image compression. It is based on the merging of two main tendencies such as the use of KLT as spectral decorrelator and object based image coding schemes. The use of the principal component in multispectral imagery is described and used to perform a multispectral segmentation. This segmentation is taken as the basis for a specific spectral decorrelation for each segmented class. The resulting eigenimages present lower variance than classical KLT approaches, leading to better compression ratios. Paper PIP.1 SEGMENTATION OF COLOR STILL IMAGES USING VORONOI DIAGRAMS Susumu Itoh and Ichiro Matsuda Science University of Tokyo 2641 Yamazaki Noda-shi, Chiba Prefecture, 278 JAPAN Tel: +81 471 24 1501, ext. 3711; fax: +81 471 24 9367 e-mail: itoh@itohws01.ee.noda.sut.ac.jp This paper proposes a new segmentation method based on Voronoi diagrams in order to develop efficient region-oriented coding for color still images. The method disposes generators according to local activity of a color image, and modifies their positions so that boundaries between Voronoi regions can run parallel to the principal contours in the image. Since a Voronoi diagram is uniquely determined by only positions of generators, the method can efficiently represent region-shapes. Moreover it can segment images quite freely, because there is in general no limitation about disposition of generators. Simulation results indicate that the method realizes better segmentation even at a low coding rate than a conventional method. Paper PIP.2 TEXTURED IMAGES SEGMENTATION BY A MULTIRESOLUTION MORPHOLOGICAL DECOMPOSITION METHOD A. Ploix, V. Chen, P. Leclere, M. Roussel LAM - Equipe de Troyes - LTI - IUT de TROYES BP 396 - 10026 TROYES CEDEX - FRANCE Tel: (33) 25.42.46.43; fax: (33) 25.42.46.43 ABSTRACT This contribution deals with the textured images segmentation. The model exploits morphological operators and order filters properties. A morphological decomposition filters bank is built to isolate elementary patterns by decomposing the textural image characteristics. The 1 and 2 order statistic moments and the gradient means are computed in order to select the best feature component image which allows to perform the image segmentation. The method is illustrated by a real image randomly textured. Paper PIP.3 UNSUPERVISED TEXTURE SEGMENTATION USING 2-D AR MODELING AND A STOCHASTIC VERSION OF THE EM PROCEDURE Claude Cariou, Kacem Chehdi LASTI - Groupe Image Ecole Nationale Suprieure de Sciences Appliques et Technologie BP 47 - 6, rue de Kerampont 22305 Lannion Cedex - France Tel: (+33) 02 96 46 50 30 ; Fax: (+33) 02 96 37 01 99 claude.cariou@enssat.fr, kacem.chehdi@enssat.fr The problem of textured image segmentation upon an unsupervised scheme is addressed. Until recently, there has been few interest in segmenting images involving possible complex random texture patterns. It is also a fact that most unsupervised segmentation techniques generally suffer from the lack of information about the correct number of texture classes. Therefore, this number is often assumed known a priori. On the basis of the so-called SEM (Stochastic Expectation Maximisation) algorithm, we try to perform a reliable segmentation without such prior information, starting from an upper bound for the number of texture classes. The image model first assumes an autoregressive (AR) structure for the class-conditional random field, and in a further step, a Markovian structure of the region process. The application of this method on a textured mosaic is presented. Paper PIP.5 A MORPHOLOGICAL ALGORITHM FOR PHOTOMOSAICKING Francisco P. Araujo Jr. and Neucimar J. Leite IC-UNICAMP Cx. Postal 6065 13081-970 Campinas - SP, Brazil e-mail: neucimar@dcc.unicamp.br We define a morphological algorithm to combine two overlapping images into a single one by a process named photomosaicking. By means of a very powerful morphological operation, namely, the watershed transformation, the method described here considers global information of a correlation image to obtain a seam which is connected, irregular and, thus, more realistic than those defined by the existing methods. Paper PIP.6 APPLYING MULTI-ANGLED PARALLELISM TO SPANISH TOPOGRAPHICAL MAPS. Josep-Maria Cusco, Marcos Faundez. Departament de Teoria del Senyal i Comunicacions. ETSE Telecomunicacio (Universitat Politecnica de Catalunya). c/ Gran Capita, s/n, E08034 Barcelona, Spain. e-mail: marcos@gps.tsc.upc.es Multi-Angled Parallelism (MAP) is a method to recognize lines in binary images. It is suitable to be implemented in parallel processing and image processing hardware. The binary image is transformed into directional planes, upon which, directional operators of erosion-dilation are iteratively applyed. From a set of basic operators, more complex ones are created, which let to extract the several types of lines. Each type is extracted with a different set of operations and so the lines are identified when extracted. In this paper, an overview of MAP is made, and it is adapted to line recognition in Spanish topographical maps, with the double purpose of testing the method in a real case and studying the process of adapting it to a custom application. Paper PIP.7 SYNTHESIS-BY-ANALYSIS OF COMPLEX TEXTURES Patrizio Campisi , Alessandro Neri , Gaetano Scarano Electronic Engineering Dept., University of Rome III via della Vasca Navale 84, I-00146 Rome, Italy Tel:+39.6.5517.7004, Fax:+39.6.5579.078, email: neri@infocom.ing.uniroma1.it INFOCOM Dept., University of Rome "La Sapienza" via Eudossiana 18, I-00184 Rome, Italy Tel:+39.6.4458.5500, Fax:+39.6.4873.300, email: gaetano@infocom.ing.uniroma1.it A technique for unsupervised texture synthesis by analysis is presented. It is based on a stochastic approximation of a textured field obtained by nonlinearly transforming a complex white Gaussian random field. The nonlinear transformation is constituted by two linear filters connected by a complex hard-limiter. The identification of the texture model is performed by means of a Bussgang blind deconvolution algorithm exploiting a generalization to the complex case of the Van Vleck rule. After a theoretical discussion of the method typical examples are provided. Paper PIP.8 CORK PORES AND DEFECTS DETECTION BY MORPHOLOGICAL IMAGE ANALYSIS Fernando Lopes, Helena Pereira (1) Francesco G.B.De Natale, Frank Tintrup, Daniele D.Giusto, Gianni Vernazza (2) (1) Departamento de Engenharia Florestal Instituto Superior de Agronomia Universidade Ticnica de Lisboa, Portugal (2) Dept. of Electrical and Electronic Engineering Universita di Cagliari, Italy e-mail: vernazza@diee.unica.it ABSTRACT The paper presents an application of rank filters to the problem of autometed visual inspection of materials. The aim of the system was to verify the quality of cork planks through the detection, classification and statistical quantification of pores and defects present in the acquired samples. The techniques adopted are a combination of morphological operators, applied to appropriate masks adaptively determined, and rank-order functions. Paper PIP.9 TEXTURES DISCRIMINATION ENHANCEMENT BY FUSION WITH SECOND AND FOURTH ORDER STATISTICS Carlos Avils-Cruz, Anne Gurin-Dugu INPG-TIRF , 46, Avenue Flix Viallet, F-38031 Grenoble cedex email : aviles@tirf.inpg.fr ABSTRACT In this paper, second and fourth order statistical moments are used to segment fine grain textures. The fusion of the moments is made through different implementations (serial or parallel strategies) and different formalism (Bayes and Evidence theory). A comparative study of the classification performance is presented and interpreted. Paper PIP.10 Title : GRAPH MATCHING BY RELAXATION TECHNIQUE Authors : Seong Hak Cheong and Sang Uk Lee Affliation : School of Electrical Engineering, Seoul National University, Seoul, 151-742, Korea Email : shcheong@phoenix.dwe.co.kr Abstract : In this paper, we describe a hybrid relaxation approach to a graph matching problem, by combining both the discrete and continuous relaxation techniques. Compatibility coefficient, critical factor for both relaxation techniques, is defined in terms of nodes and arcs attributes, and the distance measure between graphs is defined as the inner product of the probability vector and the compatibility vector. The discrete relaxation is used as a preprocessing step to determine the initial matching probabilities, and in the continuous relaxation stage, the final matching probabilities are computed by the gradient projection method, Experimental results show that the proposed algorithm is robust to the corruption of the topologies of the graphs, and the matching probabilities converges very rapidly, alleviating an enormous computational load required for the relaxation process. Paper PIP.11 USING COLOR DISTRIBUTION TO EFFECTIVELY QUERY IMAGE DATABASES B. Barolo, I. Gagliardi, R. Schettini Istituto Tecnologie Informatiche Multimediali (ITIM), Consiglio Nazionale delle Ricerche (CNR) Via Ampere 56, 20131 Milano, Italy e-mail: centaura@itim.mi.cnr.it ABSTRACT We present here an effective image retrieval strategy based on the fuzzy evaluation of color image similarity. In this method both the query and the database images are displayed in device-independent space with a limited palette of perceptual significance. Image color distributions are represented by histograms, and a suitable similarity measure between histograms is also defined in order to model the perceptual similarity between their different colors. Experimental results on a database of some 200 images are reported. Paper PIP.12 MODIFIED SIGMA FILTER FOR PROCESSING IMAGES CORRUPTED BY MULTIPLICATIVE AND IMPULSIVE NOISE Vladimir V. Lukin*, Nikolaj N. Ponomarenko*, Pauli S. Kuosmanen**, and Jaakko T. Astola** *Dept 507, Kharkov Aviation Institute, Chkalova St 17, 310070, Kharkov, Ukraine **Signal Processing Laboratory, Tampere University of Technology, P.O.Box 553, FIN-33101 Tampere, Finland e-mail: lukin@mmds.kharkov.ua, pqo@cs.tut.fi, jta@cs.tut.fi A new modification of sigma filter is proposed and tested in this paper. This modification is suitable for processing images corrupted by Gaussian multiplicative and impulsive noises and it avoids some typical disadvantages of the standard sigma filter while possessing robust properties. The test images include both simulated and real radar images. It is seen that the proposed modification provides improved speckle suppression efficiency and less bias for homogeneous regions of images. Paper PIP.13 EDGE-PRESERVING SMOOTHING BY ADAPTIVE NONLINEAR FILTERS WITH LAYERED NEURAL NETWORKS Mitsuji Muneyasu, Yuji Wada and Takao Hinamoto Faculty of Engineering, Hiroshima University 1-4-1 Kagamiyama, Higashi-Hiroshima, Hiroshima 739, Japan e-mail: muneyasu@ecl.sys.hiroshima-u.ac.jp A new type of edge-preserving smoothing filters to be applied to the images corrupted with impulsive and white Gaussian noise is developed. This filter is based on the weighted mean filter having two kinds of coefficients for impulsive and Gaussian noises, respectively. These coefficients can be varied adaptively by some kinds of local features in the window. The layered neural networks are used for the implementation of the proposed filter. The coefficients of the proposed filter can adapt itself to the nature of an image by the learning of networks. The result of the simulation is demonstrated the effectiveness of the proposed technique. Paper PIP.14 2-D ADAPTIVE PIECEWISE-LINEAR FILTER FOR IMAGE ENHANCEMENT V. Pahor, G. Ramponi, G.L. Sicuranza D.E.E.I., University of Trieste via A. Valerio, 10, 34127 Trieste, Italy Tel: +39 40 6767140; fax: +39 40 6763460 e-mail: pahor@imagets.univ.trieste.it A two-dimensional adaptive nonlinear filter, called 2-D FIR-PWL filter is introduced for noise cancellation from images. It is based on the cascade of a linear FIR filter and a piecewise-linear interpolating function. Experimental results show a very good behaviour of the filter, which outperforms in many application examples the Sigma filter both in terms of visual quality and numerical results. Paper PIP.15 ADAPTIVE WEIGHTED D-ALPHA FILTER I. ISSA, Ph. BOLON Laboratoire d'Automatique et de MicroInformatique Industrielle LAMII/CESALP - Universit de Savoie - B.P 806 74016 Annecy Cedex (France) (CNRS G1047 - Information-Signal-Image) e-mail: {issa, bolon}@esia.univ-savoie.fr In this paper we propose a new adaptive weighted d-alpha filter. The filter is adaptive regarding noise amplitude distribution, orientation of structures and anisotropy measures. The filter coefficient are chosen according to structure orientation and anisotropy measures. a value is chosen according to the result of local noise distribution and anisotropy coefficient estimations. Some experimental results on synthetic and natural images are presented. Results are compared with those of adaptive filters such as the adaptive trimmed mean. Paper PSO.1 MANEUVERING TARGET MOTION ANALYSIS USING BSPLINE REPRESENTATION Laurent Deruaz Thomson-Sintra ASM, BP 157 06903 Sophia-Antipolis cedex, FRANCE Target motion analysis (TMA) for a rectilinear source movement (RSM) has been intensively studied in the last ten years. But difficulties still exist, especially when source heading or speed changes are within the same time as the conventional TMA convergence time. This paper is concerned with a new method of batch TMA for maneuvering sources using a non-linear least-squares fit between the whole set of measurements and a BSpline trajectory representation. It provides a good way to globally estimate both the instants of maneuvers and their number with an experimentally robust model order selection method. This work includes tests on actual data from at-sea recordings. Paper PSO.2 3D TRACKING SONARS WITH HIGH ACCURACY OF RANGE MEASUREMENTS FOR AUTONOMOUS MOBILE ROBOT NAVIGATION Angelo M. Sabatini ARTS-Lab, Scuola Superiore Sant' Anna Via Carducci, 40, 56127 Pisa, Italy Tel: +39-50-883207; fax: +39-50-883215 e-mail: ANGELO@HELIOS . SSSUP . IT ABSTRACT An array of in-air sonar sensors using correlation techniques for range estimation is developed for accomplishing object identification and location in the 3D space; the intended applications are mainly in the field of autonomous mobile robot navigation. A major emphasis in this paper is given to the concept of the baseband equivalent receiver which is proposed for designing digital correlators of low complexity. Thanks to the combination of analog multiplexing and second order bandwidth sampling techniques, the baseband equivalent receiver we propose proves to be a valuable concept for designing a novel class of tracking sonar devices. Paper PSO.3 DATA ASSOCIATION AND TRACKING FROM ACOUSTIC DOPPLER AND MAGNETIC MEASUREMENTS Gilles Dassot, Claire Chichereau, Roland Blanpain LETI (CEA - Technologies Avances) DSYS -- CEA - Grenoble - 17, rue des martyrs -- 38054 Grenoble Cedex 9 - France -- Tel: +33 76 88 36 12; fax: +33 76 88 51 59 -- e-mail: dassot@cea.fr This paper is devoted to the localisation problem of acoustic-magnetic sources moving in straight line at constant speed. Our technique is based on the association of Acoustic Doppler and Magnetostatic Methods. The objective of this study is to achieve localisation with only one sensor performing both frequency and magnetic measurements. The set of possible location is shown to be a circle since no angular information is available. The subsequent developments describe an Extended Kalman Filter with a linear observation equation to perform maximum performance in case of poor initialisation. The filter convergence is actually ensured when tested with simulated signals. A small residual bias on the velocity estimate is however noticed due to the non linearity of the prediction equation. Paper PSO.4 SOURCE LOCALIZATION WITH OVERLAPPING CW INTERCEPTS USING MULTIPATH MODELING Pierre Blanc--Benon Thomson-Sintra ASM, BP.157, 06903 Sophia-Antipolis Cedex, France This paper addresses the problem of passively locating a CW pulse emitter without a priori information concerning the pulse duration or the number of overlapping paths being intercepted. It differs from a previous approach by Manickam, since it consists on jump on-line detections, with a single path being concerned each time. Basically, the method relies on a non-linear ML estimation of the sinusoid parameters and a jump detector based on forward- backward estimation residuals to compute the individual times of beginning -ending for each path. Using the sound speed profile, a non-linear (NL) least-squares fit between the measured time-delays and the guessed ones enables to locate the source in both range and depth. Monte-Carlo simulation in a deep Mediterranean like channel demonstrates the capability of the method for various signal to noise ratio. The Cramer-Rao bounds of the range-depth estimation are computed by using an analytic modelization of the ray propagation. At last, a 1-hour recorded signal experiment proves the at-sea efficiency of the method for a source located in the 20-30 km ranges of the deep Mediterranean channel. Paper PSO.5 TIME DELAY ESTIMATION IN A MULTIPATH CONTEXT Pierre COMON, Bruno EMILE, and Georges BIENVENU I3S-CNRS, 250 av. Albert Einstein, F-06560 Valbonne and Thomson-Sintra ASM, B.P. 157, F-06903 Sophia-Antipolis Cedex comon@asm.thomson.fr, emile@alto.unice.fr A second-order blind deconvolution algorithm is utilized to improve on interception and classification procedures. It consists of applying the subspace decompostion algorithm described by Moulines et al. to several portions of the observation received on a single sensor, and then of estimating the source signal cleaned from its interferences caused by the multipath propagation. Asymptotic performances are lastly analyzed in terms of mean and variance of the estimated filters. Paper PSO.6 WIDEBAND INVERSE FILTERING TO IMPROVE ACTIVE SONAR DETECTION IN BACKGROUND REVERBERATION P. Delachartre, D. Vray, N. Ma, A. Bacelar, G. Gimenez, Y. L. Ma* CREATIS, Research Unit associated to CNRS (UMR #5515) and affiliated to INSERM, Lyon INSA 502, 69621 Villeurbanne cedex (France) e-mail: delachartre@creatis.insa-lyon.fr *Northwesten Polytechnical University 710072 Xi'an (P. R. China) The problem of detecting a known signal in background reverberation with an estimated reverberation spectrum is addressed. In our approach, the prewhitener is a wideband inverse filter estimated from a large data base of reverberation spectra. Simulations and experimental results are presented in the context of detecting a target lying on the seafloor with a wideband transducer. The proposed detector is compared to an AR prewhitener. The results indicate that the proposed detector is well suited for our wideband application. Paper PSO.7 ACCURATE FISH POSITION AND ORIENTATION PARAMETERS CORRELATED TO WIDEBAND ECHO : A NEW APPROACH FOR CLASSIFICATION OF FISH SPECIES Bacelar A., Neyran* B., Delachartre P., Vray D., Gimenez G., CREATIS - Research mixed unity 5515 of CNRS and affiliated to INSERM, INSA 502 - 69621 Villeurbanne cedex (France) Tel : (33) 72 43 81 48; Fax : (33) 72 43 85 26 e-mail : alexis.bacelar@creatis.insa-lyon.fr * team of Lyon I university This work deals with the correlation between high accurate geometric parameters of a free- swimming fish in a tank, obtained by image processing, with the associated 20-140 kHz wideband sonar echo acquired simultaneously. Variations in terms of Target Strength and normalized spectral energy are studied for three species of fish, perch, roach and char, according to different geometric parameters. Classification of the three species is implemented. Paper PSO.8 RADAR SIGNAL EXTRACTION USING CORRELATION LANON Fabienne1-2, HILLION Alain1, SAOUDI Samir1 1 ENST-Bretagne, Dpartement Signal et Communication, BP 832, 29285 BREST CEDEX, e-mail : fabienne.lancon@enst-bretagne.fr e-mail : alain.hillion@enst-bretagne.fr e-mail :samir.saoudi@enst-bretagne.fr 2 THOMSON-CSF, DIVISION RCM, Centre lectronique de Brest, 10 Avenue 1re DFL, 29283 BREST CEDEX, Fax : (33)98312763, Tel : (33)98312705, e-mail : fabienne.F.L.lancon@rcm.thomson.fr In this paper we present a post-integration processing in order to improve sensitivity of electronic support measure (ESM) receivers. Correlation methods take advantage of periodic character of radar signals. In such case, autocorrelation and cross-correlation improve detection of signals with high repetition frequency. Furthermore, since the extraction of radar parameters is necessary to identify received signals, we study three types of estimators : straightforward method, interpolation method and maximum likelihood one. Simulation studies with realistic models and real signals are carried out to validate performances of such processing. With a view to implanting correlation functions, some architectures are studied. The choice of a method is of interest since we need a lot of samples to be integrated. To conclude, as radar ESM receiver requires most information on received signals, enhancement of sensitivity thanks to correlation method is of great interest. Paper PSO.9 Performance Indicators of the Correlation Process for Non Ambiguous Doppler Frequency Estimation in Multiple PRF Radars Christophe BERENGUER and Gerard ALENGRIN Universite de Technologie de Troyes LM2S - GSI 13, Bd Henri Barbusse BP2060 10010 TROYES cedex - FRANCE Tel : +(33) 25 71 46 08 Fax : +(33) 25 82 02 75 E-mail : berenguer@univ-troyes.fr and Universite de Nice Sophia-Antipolis Labo. I3S - URA CNRS 1376 41, Bd Napoleon III 06041 NICE cedex Tel : +(33) 93 21 79 56 Fax : +(33) 93 21 20 54 E-mail : alengrin@unice.fr This communication investigates the performance of alias-free Doppler frequency estimation in multiple Pulse Repetition Frequency radar systems. Three performance indicators are proposed for Doppler ambiguity resolution algorithms based on the use of a correlation interval of given width : probabilities of correlation, of false correlation and of false measurement. Under the assumption of Gaussian errors on the ambiguous frequencies estimates for each PRF, closed forms (function only of the interval width, the PRF values and the estimation variance on the ambiguous frequencies) are derived for these indicators. Some examples of the expected behavior of MPRF systems obtained with these indicators are presented and discussed. Paper PSO.10 SURVEILLANCE RADAR WAVEFORM FITTED FOR ANTI-STEALTHNESS AND FOR COUNTER-COUNTERMEASURES: EVALUATION. N. Gonget*, P.Y. Arques*#, L. Martinet* * DCN - CTSN / LSA / TTS. B.P. 28, 83800 Toulon Naval - France. Tel: (33) 94162114 Fax: (33) 94162281 # ISITV, Universit de Toulon et du Var, B.P.32, 83957 La Garde cedex - France. Tel: (33) 94142000 ABSTRACT In the present context, the naval surveillance radar have to face important progress of both stealthness of the targets and electronic countermeasure (ECM) techniques. In order to resolve simultaneously the two problems of stealth targets and ECM techniques, we have proposed a new solution [1]. This one can be applied to the naval surveillance radar and its main characteristic is the use of a random waveform. We have presented the reception system and the simulation allowing to prove its validity [2]. In this paper, after a recall on the proposed simulation of the waveform, we discuss the performances of this naval surveillance radar waveform. Paper PSO.11 OPTIMAL WAVEFORM SELECTION FOR TARGET CLASSIFICATION Sameh m. Sowelam and Ahmed H. Tewfik Department of Electrical Engineering University of Minnesota Minneapolis, MN 55455 email: ssowelam@ee.umn.edu, tewfik@ee.umn.edua This paper studies the design of a set of outgoing radar signals to discriminate between two target classes. We model the reflectivity function of each target by a two-dimensional stochastic process to account for uncertainties and propagation effects. The signals are selected to minimize the expected number of transmissions that are needed to guarantee a given confidence level in the classification decision. We argue that this goal can be achieved by selecting the signals that maximize the {\em Kullback-Liebler information number} between the two target classes. We illustrate our approach with a particular model. We show that for this model, the optimal set of waveforms can be designed off-line and depends on both the statistics of the reflectivity functions of the targets in both classes and the observation noise level. Paper PSO.12 EXPERIMENTAL VERIFICATION OF A GENERALIZED MULTIVARIATE PROPAGATION MODEL FOR IONOSPHERIC HF SIGNALS Y. Abramovich C. Demeure A. Gorokhov Odessa State Polytechnic University av. Shevchenko 1, 270044, Ukraine Tel +38 0482 288644 THOMSON-CSF Division Communication 66 rue du Fosse Blanc, 92231, Gennevilliers, France. Tel: (33)1-46132113; Fax (33)1-46132555 Telecom Paris, Dept. Signal 46 rue Barrault 75634 Paris Cedex 13 FRANCE New stochastic model for HF signal, received by the multisensor antenna array is presented for ionospheric propagation channel. The model introduces spatial fluctuations that are observed by the receiving antenna array, along with the Doppler frequency fluctuations. The new description generalizes the existing models and collapses into the perfectly validated scalar Watterson model for the single sensor reception. The proposed model is stimulated by practical attempts to improve the performance of HF radiosystems, and has been validated by the set of experimental transmissions from Coloumier (France), received by the antenna array in Odessa (Ukraine). Experimental results demonstrate a good compliance with the introduced model. Paper PSO.13 GROUND CLUTTER DETECTION AND ELIMINATION FOR DUAL-POLARIZED WEATHER RADAR USING MULTIPARAMETER THRESHOLDS Liu, Li Radio Engineering Department South China University of Technology Guangzhou, 510641, P.R.China e-mail: ecliliu@scut.edu.cn V. N. Bringi Electrical Engineering Department Colorado State University Fort Collins, CO 80523, USA e-mail: bringi@lance.colostate.edu ABSTRACT In this paper we described the ground clutter effects on polarimetric radar parameter estimations using non-spectral approach. A simple but efficient technique for detecting and eliminating ground clutter effect on polarimetric radar measurements using multiparameter thresholds is derived based on tremendous data processing and analysis. Some typical examples are given for illustration and interpretation. Paper PSO.14 TIME-FREQUENCY ANALYSIS OF LIDAR SIGNAL TO OBTAIN GRAVITY WAVES CHARACTERISTICS Franck Molinaro, Hassan Bencherif, Miloud Bessafi Laboratoire de Physique de l'Atmosphere, Universite de la Reunion 15 Av. Rene Cassin, BP 7151, 97715 Saint Denis cedex 9, France Tel: (262)93-82-53 Fax: (262)93-81-66 Mail : molinaro@univ-reunion.fr ABSTRACT The Lidar is a laser beam which sent vertically monochromatic pulses in the atmosphere. The analysis of the back scattered light provides information about the vertical temperature evolution versus height. Temperature perturbations are associated with gravity waves phenomenon which play a major role in the middle atmosphere dynamics. The aim of the study is to identify characteristics of these particular waves above Reunion island with an usual parametric time-frequency tool. A comparison is made for two representative periods. Paper PSO.15 COHERENCE ESTIMATION OF INTERFEROMETRIC SAR IMAGES Fabio Gatelli, Andrea Monti Guarnieri, Claudio Prati. Dipartimento di Elettronica - Politecnico di Milano Pzza. L. da Vinci, 32, 20133 Milano. Italy Tel: +39-2-23993585 Fax: +39-2-23993413 e-mail: monti@elet.polimi.it Abstract Usual coherence estimation in SAR\ interferometry is a time consuming task since an accurate estimation of the local frequency of the interferometric fringes is required. In this paper a fast algorithm for generating coherence maps, mainly intended to data browsing, is presented. The proposed estimator is based on the speckle similarity of coherent SAR data and is, thus, independent of the fringes frequency. Advantages with respect to the usual estimates are achieved in terms of computational costs (up to$100\$ times
lower), robustness (the estimator presented is not affected by possible
local frequency estimation errors) and flexibility (the estimator can be
applied both to complex and to detected images). The statistical properties
of the frequency independent estimator are given in the stationary case. A
preprocessing technique that reduces the degradions due to
non-stationarities is then shown.
Paper

PSP.1
EXTENDED SPECTRAL SUBTRACTION

Pavel Sovka & Petr Pollak & Jan Kybic

Czech Technical University, Faculty of Electrical Engineering
CTU FEL K331, Technicka 2, 166 27 Praha 6, Czech Republic
Tel: (+42 2) 2435 2291
Fax: (+42 2) 2431 0784
E-mail: [sovka,pollak]@feld.cvut.cz

This paper describes a new method for one channel noise
suppression system which overcomes the typical disadvantage
of one channel noise suppression algorithms - the impossibility
of noise estimation during speech sequence. Our method is
the combination of Wiener filtering and spectral subtraction.
The noise can be successfully updated even during the speech
sequences and that is why  there is no need of the voice
activity detector.
Paper

PSP.2
NOISE REDUCTION OF SPEECH SIGNALS USING THE RANK-REVEALING ULLV DECOMPOSITION

Peter S. K. Hansen, Per Christian Hansen(1), Steffen Duus Hansen and
John Aasted Sorensen

Department of Mathematical Modelling, Section for Digital Signal Processing
Technical University of Denmark, DK-2800 Lyngby, Denmark
E-mail: pskh@imm.dtu.dk, sdh@imm.dtu.dk and jaas@imm.dtu.dk
(1)UNI-C, Technical University of Denmark, DK-2800 Lyngby, Denmark
E-mail: Per.Christian.Hansen@uni-c.dk

A recursive approach for nonparametric speech enhancement is
developed. The underlying principle is to decompose the vector space
of the noisy signal into a signal subspace and a noise subspace.
Enhancement is performed by removing the noise subspace and estimating
the clean signal from the remaining signal subspace. The decomposition
is performed by applying the rank-revealing ULLV algorithm to the
noisy signal. With this formulation, a prewhitening operation becomes
an integral part of the algorithm. Linear estimation is performed
using a proposed minimum variance estimator. Experiments indicate that
the approximative method is able to achieve a satisfactory quality of
the reconstructed speech signal comparable with eigenfilter based
methods.
Paper

PSP.3

Speech Enhancement Using a Wiener Filtering Under Signal Presence Uncertainty

A. AKBARI AZIRANI - R. LE BOUQUIN JEANNS - G. FAUCON

Laboratoire du Traitement du Signal et de l'Image - Universit de Rennes 1
Bt. 22 - Campus de Beaulieu - 35042 RENNES CEDEX - FRANCE
Regine.Lebouquin@univ-rennes1.fr

Abstract
Noise reduction is a key-point of speech enhancement systems in hands-free communications.
A number of techniques have been already developed in the frequency domain such
as an optimal short-time spectral amplitude estimator proposed by Ephraim and Malah
including the estimation of the a priori signal-to-noise ratio. This approach reduces
significantly the disturbing noise and provides enhanced speech with colorless residual
noise. In this paper, we propose a technique based on a Wiener filtering under uncertainty
of signal presence in the noisy observation. Two different estimators of the a priori
signal-to-noise ratio are tested and compared. The main interest of this approach
comes from its low complexity.
Paper

PSP.4

IMPROVED SPECTRAL SUBTRACTION FOR
SPEECH ENHANCEMENT

Y. Malca and D. Wulich
Department of Electrical & Computer Engineering,
Ben-Gurion University of the Negev.
Beer-Sheva 84105, POB 635, Israel.
Tel: ++972-7-461537, Fax: ++972-7-472949,
e-mail: dov@bguee.bgu.ac.il

ABSTRACT
The spectral subtraction approach has become almost standard in speech
enhancement because it is relatively easy to understand and implement.
The major drawback of the spectral subtraction method is that it leaves
residual noise with annoying noticeable tonal characteristics referred
to as musical noise. For low SNR the perceived effect of the "musical noise"
is close to that of the additive noise.
In the present work we propose to reduce the musical noise by applying
the output of a standard spectral subtractor to a constrained high order
notch filter which suppresses the "musical noise". The filtration process
distorts the speech signal. It is possible to reduce the level of distortion
if the speech signal is preprocessed properly before it is contaminated by
the noise. It will be demonstrated that the proposed method is superior to
the standard spectral subtraction specially for low SNR. A comprehensive
listening test indicated that for segmental SNR= -12dB, 77% of the listeners
strongly preferred the proposed approach over the usual spectral subtraction
approach.
Paper

PSP.5
A SINGLE MICROPHONE NOISE CANCELLER BASED ON

M. Gabrea, E. Mandridake and M. Najim
Equipe Signal et Image, ENSERB and GDR-134, CNRS
BP 99, 33 402 Talence, FRANCE
email: najim@goelette.tsi.u-bordeaux.fr

This paper deals with the problem of Adaptive Noise
Cancellation (ANC) when only corrupted speech signal
with an additive Gaussian white noise is available for
processing. We propose a new method based on adaptive
Kalman filtering. All the approaches based on the Kalman
filter proposed in the past, in this context, operate in two
steps: they first estimate the noise variance and the
parameters of the signal model and secondly estimate the
speech signal. The approach presented in this paper gives
an alternative to these approaches since it does not require
the estimation of the noise variance. The noise variance
estimation is a part of the Kalman gain calculation. For
optimizing the Kalman gain we have reformulated and
adapted, to the single-microphone ANC problem, the
approach proposed in control by R. K. Mehra.
Paper

PSP.6
TWO MICROPHONES SPEECH ENHANCEMENT SYSTEM BASED ON A
DOUBLE FAST RECURSIVE LEAST SQUARES (DFRLS) ALGORITHM

M. Gabrea*, E. Mandridake*, M. Menez+, M. Najim* and A. Vallauri++
* Equipe Signal et Image, ENSERB and GDR-134, CNRS
BP 99, 33 402 Talence, France
+ LASSY-I3S Nice, France
++ Texas-Instruments, Villeneuve-Loubet, France
email: limby@goelette.tsi.u-bordeaux.fr

In this paper a symmetric feedback implementation scheme of a
two microphones speech enhancement is presented. We consider
the coupling systems modelled as a linear time-invariant Finite
Impulse Response (FIR) filters and propose a new recursive-based
adaptive filter solution to enhance the noisy speech . The optimum
filter weight adaptation is based on a Double Fast Recursive Least
Squares (DFRLS) algorithm. This approach can be extended for a
subclass of signal separations where the direct link is stronger than
the interference link in the both channels. A comparative study
with other adaptive algorithms shows the superiority of the
DFRLS in SNR performance improvement.
Paper

PSP.7
Signal Restoration of Broad Band Speech Using Nonlinear Processing

Hiroshi Yasukawa

NTT Optical Network Systems Labs.
1-2356 Take, Yokosuka, 238-03 Japan
Tel: +81-468-59-3016;	Fax: +81-468-55-1283
e-mail: yasukawa@exa.onlab.ntt.jp

ABSTRACT
This paper describes a new system that can enhance the quality of speech
signals that have been severely band limited during transmission. We have
already proposed a spectrum widening method that utilizes aliasing in
sampling rate conversion with digital filtering for spectrum shaping.
This paper proposes a quite simple method by adding spectrum in the higher
band using nonlinear processing. Implementation procedures are clarified,
and its performance is discussed. It is shown that the proposed method
offers good performance in terms of spectrum distortion characteristics.
Paper

PSP.8
Adaptive Digital Filtering For Signal Reconstruction Using Spectrum Extrapolation

Hiroshi Yasukawa

NTT Optical Network Systems Labs.
1-2356 Take, Yokosuka, 238-03 Japan
Tel: +81-468-59-3016;	Fax: +81-468-55-1283
e-mail: yasukawa@exa.onlab.ntt.jp

Abstract
This paper describes adaptive filtering for signal reconstruction. The speech quality
enhancement system by the spectrum extrapolation of the band limited signals is
discussed. In telephone communication, the spectrum extrapolation which employs
aliasing processing is widely known. In this paper a new implementation using adaptive
methods is proposed. This method introduces frequency domain adaptive digital filtering
to broaden band limited signals into wide band signals. Implementation of the system
and its performance are discussed.
Paper

PSP.9
COMBINATION OF TWO-CHANNEL SPECTRAL SUBTRACTION AND ADAPTIVE WIENER
POST-FILTERING FOR NOISE-REDUCTION AND DEREVERBERATION

Matthias Doerbecker, Stefan Ernst

Institute of Communication Systems and Data Processing,
Aachen University of Technology, 52056 Aachen, Germany
e-mail: matthias@ind.rwth-aachen.de

In this contribution a novel structure for the enhancement of speech signals
disturbed by acoustic noise is presented which is based on Spectral
Subtraction. The Spectral Subtraction technique is combined with a novel
estimator for the noise power spectrum which takes advantage of the employment
of a second microphone. Due to the extension to a two-microphone system the
Spectral Subtraction can be used to reduce realistic, non-stationary noise
sources.  Additionally, the performance of the system is further improved by
the application of a post filter adapted according to Wiener filter
techniques.  As a result, the proposed speech enhancement system provides a
significant suppression of noise in realistic situations as well as a
reduction of room reverberation.


Paper

PSP.10
LIP MOVEMENTS SYNTHESIS USING TIME DELAY NEURAL NETWORKS

Sergio Curinga, Fabio Lavagetto, Fabio Vignoli

D.I.S.T. - University of Genova
Via Opera Pia 13A, 16145 GENOVA
E-mail: sergio@dist.dist.unige.it

Abstract
A method exploiting the audio-visual correlation of speech
in order to estimate the lip and mouth movements is presented.
Its applications are in the field of aids and services for elderly
people, in videotelephony, in cartoons and movie dubbing. Notice
that lip movements synthesis does not imply speech recognition and
that the mouth shape is not only specified by the phoneme currently
uttered but it also depends on some past and future speech information.
In order to take into account this temporal correlation, and considering
the constraint of computational effectiveness, the Time Delay Neural
Networks (TDNNs) seem to be the most appropriate analysis tool in
comparison with methods like Markov Models, which are more resource
consuming.
Paper

PSP.11
SPEECH SEGMENTATION USING MULTILEVEL HYBRID FILTERS

Marcos Faundez, Francesc Vallverdu

Department of Signal Theory and Communications UPC
e-mail: marcos@gps.tsc.upc.es

A novel approach for speech segmentation is proposed, based on Multilevel
Hybrid Filters with the following features:
- An accurate transition location
- Good performance in noisy environments (gaussian and impulsive noise)
The proposed method is based on spectral changes, with the goal of segmenting
the voice into homogeneous acoustic segments.
This algorithm is being used for phonetically segmented speech coder with
successful results.
Paper

PSP.12

Joao Manuel Rodrigues
Ana Maria Tome

Departamento de Electronica e Telecomunicacoes / INESC
3810 AVEIRO, PORTUGAL
Tel: +351-34-370500; Fax: +351-34-370545
e-mail: jmr@inesca.pt

This paper presents a new audio compression algorithm that includes a
nonuniform filter bank, gain-adaptive logarithmic quantizers, arithmetic
entropy coding and an explicit psychoacoustic model to adapt the quantization
according to perceptual considerations.  Unlike existing perceptual coders,
already quantized samples, not on the original signal.  We discuss the
applied to perceptual coding.
Paper

PSP.13
Title : SAMPLE-BY-SAMPLE GAIN ADAPTIVE CELP CODING OF WIDEBAND AUDIO

Authors : Man-Tak Chu and Cheung-Fat Chan

Affiliation :	Department of Electronic Engineering
City University of Hong Kong
83, Tat Chee Avenue,  Hong Kong
email : eecfchan@cityu.edu.hk
fax : (852) 27887791

ABSTRACT
--------
This  paper  presents  a  high quality wideband audio coder based on a low delay code excited
linear predictive  (LD-CELP) model where the excitation gain is adapted in a sample-by-sample
manner.  The  proposed  coder employs a backward adaptive predictor which introduces no extra
delay to the system. A simple gain adaptive control is utilized to perform a sample-by-sample
gain adaptive excitation model. In other words, the proposed coder exploits the advantages of
the  LD-CELP  and ADPCM coding. This coder can provide transparent quality audio signals at a
bitrate of 1.5 bits/sample.
Paper

PSP.14
SPLIT-BAND LD-CELP WIDEBAND SPEECH CODING AT 24 KBIT/S

Andrea Santilli(*), Aurelio Uncini(**), Francesco Piazza(**)

(*) AETHRA S.r.L.  60020 Palombina (AN), Italy
(**) Dip. Elettronica ed Automatica, Univ. of Ancona, 60131 Ancona, Italy
phone: +39 71 220 4453    fax: +39 71 220 4464
e-mail: upfm@eealab.unian.it

Nowaday 7 Khz wideband speech coding requires at least 48 kbit/s as it still depends
on the ITU standard G.722. CELP coders have been developed for wideband systems achieving
high quality speech coding at rates from 16 kbit/s to 32 kbit/s as the wideband LD-CELP at 32 kbit/s.
In this paper, a new split-band LD-CELP wideband coder at 24 kbit/s is proposed and its performance
and complexity are compared with those of the already known wideband LD-CELP.
Paper

PSP.15

INNOVATION CODING WITH A CROSS-CORRELATED QUANTIZATION NOISE MODEL

Soeren Vang Andersen, Morten Olesen, Soeren Holdt Jensen, and Egon Hansen

CPK, Aalborg University, Fredrik Bajers Vej 7, DK-9220 Aalborg OEst, Denmark.
E-mail: sva@cpk.auc.dk

We present the use of a cross-correlated quantization noise model
in the recently proposed Kalman innovation speech coding
scheme. Computer simulations and
informal listening tests indicate that the incorporation of a
cross-correlated noise model yields an
improvement in both SNR and perceptual quality when compared to a
uncorrelated noise model.
Paper

PSR.1
MINIMUM CLASSIFICATION ERROR TRANSFORMATIONS
FOR IMPROVING SPEECH RECOGNITION SYSTEMS

Angel de la Torre, Antonio M. Peinado, Antonio J. Rubio,
Jose C. Segura, Victoria E. Sanchez

Dpto. de Electronica y Tecnologia de Computadores
e-mail atv@hal.ugr.es

Signal representation is an important aspect to be taken into account
for pattern classification. Recently, discriminative training methods
have been applied to feature extraction for speech recognition.
In this paper, we apply the Minimum Classification Error estimation to
train the parameters of a feature extractor. This feature extractor is
a linear transformation of the original representation space.
The new representation of the speech signal makes easier the
recognition task and the performance of the different tested
recognizers is improved as the experimental results show.
Paper

PSR.2
TOWARDS SUBBAND-BASED SPEECH RECOGNITION

Herv Bourlard (1,3)
Stphane Dupont (1)
Hynek Hermansky (2,3)
Nelson Morgan (3)

(1) Facult Polytechnique de Mons - TCTS
31, Bld. Dolez, B-7000 Mons, Belgium
Email: bourlard,dupont@tcts.fpms.ac.be
(2) Oregon Graduate Institute, Portland, OR, USA
(3) Intl. Computer Science Institute, Berkeley, CA, USA

In the framework of hidden Markov models (HMM) or hybrid HMM/Artificial
Neural Network (ANN) systems, we present a new approach towards
speech recognition.
The general idea is to split the whole frequency band (represented
in terms of critical bands) into a few subbands on which different recognizers
are independently applied and then recombined at a certain speech unit level
to yield global scores and a global recognition decision.
The preliminary results presented in this paper show that such an
approach, even using quite simple recombination strategies, can yield
at least comparable performance on clean speech while providing significantly
better robustness in the case of speech corrupted by narrowband noise.
Paper

PSR.3
NONLINEAR DISCRIMINANT ANALYSIS WITH NEURAL NETWORKS FOR SPEECH RECOGNITION

Vincent Fontaine, Christophe Ris, Henri Leich

Faculte Polytechnique de Mons --- TCTS
31, Bld. Dolez, B-7000 Mons, Belgium
Tel : + 32 65 374176 - Fax : + 32 65 374129
e-mail: {fontaine,ris,leich}@tcts.fpms.ac.be

Linear Discriminant Analysis (LDA) has been applied successfully to speech
recognition tasks, improving accuracy and robustness against some types of
noise. However, it is well known that LDA suffers from some weaknesses if
the distributions are not unimodal or when the mean of the distributions
are shared.
In this paper, we propose to take advantage of the nonlinear discriminant
properties of the Artificial Neural Networks (ANN) in the task of reducing
the dimensionality of the input space, leading to a nonlinear discriminant
analysis.
Paper

PSR.4
ROBUST SPEECH RECOGNITION USING FUZZY MATRIX QUANTISATION,
NEURAL NETWORKS AND HIDDEN MARKOV MODELS

Professor C S Xydeas and Lin Cong

Speech Processing Research Laboratory, Electrical Engineering Division,
School of Engineering, University of Manchester, Dover Street, Manchester,
M13 9PL, UK, Tel/Fax: +44[161]2754511/2754528, E-Mail: c.xydeas@man.ac.uk

Abstract
In this paper a new approach to robust speech recognition using Fuzzy
Matrix Quantisation, Hidden Markov Models and Neural Networks is presented
and tested when speech is corrupted by car noise. Thus two new robust
isolated word speech recognition (IWSR) systems called FMQ/HMM and
FMQ/MLP, are proposed and designed optimally for operation in a variety
of input SNR conditions. The schemes and associated system training
methodologies result into a particularly high recognition performance
at input SNR levels as low as 5 and 0 dBs.
Paper

PSR.5
LOCALLY RECURRENT NEURAL NETWORKS FOR EFFICIENT REALIZATION OF A SPEECH
RECOGNIZER

Klaus Kasper, Herbert Reininger, Dietrich Wolf, and Harald Wuest

wuest@apx00.physik.uni-frankfurt.de

The computational complexity of speech recognizers based on fully connected
recurrent neural networks, i.e. the large number of connections, prevents
a hardware realization.
We introduced locally connected recurrent neural networks in order to
keep the properties of recurrent neural networks and to reduce the connectivity
density of the network.
A special form of feature presentation and output coding is developed
which reduces the computational complexity and allows learning of long-term
dependencies.

By applying all these methods a locally recurrent neural network results,
which has only one third of the weights as a fully connected recurrent
network.
Thus, with this concept a speech recognition system can be realized on
a single VLSI-Chip.
Paper

PSR.6
TEXT-INDEPENDENT OFF-LINE WRITER RECOGNITION USING NEURAL NETWORKS

D. A. Valkaniotis, J. Sirigos, N. Fakotakis and G. Kokkinakis
Wire Communications Laboratory, University of Patras, 26500 Patras, Greece
Tel: +33 61 991722;  fax: +33 61 991855
e-mail : valkan@wcl.ee.upatras.gr

ABSTRACT
In this paper we present a text-independent off-line writer recognition system based
on multi-layer perceptrons (MLPs). The system can be used for both identification
and verification purposes. It was tested on a population of 20 writers with non-correlated
training and test specimens. The mean error for identification was 3.5% while error
rates as low as 0.5% were achieved on specimens with more than 25 characters. For
verification the mean error was 1.2% (2.22% false rejection, 0.18% false acceptance)
considering a minimum of 15 characters per test specimen. These error rates are
comparable to those achieved by classical methods while the response of the system
is substantially faster.
Paper

PSR.7
SEGMENTAL LVQ3 TRAINING FOR PHONEME-WISE TIED MIXTURE DENSITY HMMS

Mikko Kurimo

Helsinki University of Technology, Neural Networks Research Centre
Rakentajanaukio 2 C, FIN-02150, Espoo, FINLAND
tel: +358 9 451 3266
fax: +358 9 451 3277
email: mikko.kurimo@hut.fi

The system trains speaker dependent, but vocabulary independent, phoneme
models for the recognition of Finnish words.
The Learning Vector Quantization (LVQ) methods are applied to
increase the discrimination between the phoneme models.
A segmental LVQ3 training is proposed to substitute the LVQ2
based corrective tuning as a parameter estimation method.
The experiments indicate that the new method can provide
the corresponding recognition accuracy, but with less training and more
robustness over the initial models.
Experiments to up-scale the current system by introducing context
vectors and larger mixture pools show up to 40 % reduction of
recognition errors compared to the earlier results.
Paper

PSR.8

Title :        THIRD-ORDER CUMULANT-BASED WIENER FILTERING ALGORITHM
APPLIED TO ROBUST SPEECH RECOGNITION

Authors :      Josep M. SALAVEDRA, Javier HERNANDO

Affiliations : Universitat Politecnica de Catalunya.
c/ Gran Capita s/n. 08034-BARCELONA. SPAIN.
Tel/Fax: +34-3-4017404 / 4016447 .
E-mail: mia@gps.tsc.upc.es

ABSTRACT :

In previous works [5], [6], we studied some speech enhancement algorithms based
on the iterative Wiener filtering method due to Lim-Oppenheim [2], where the AR
spectral estimation of the speech is carried out using a second-order analysis.
But in our algorithms we consider an AR estimation by means of cumulant analysis.
This work extends some preceding papers due to the authors: a cumulant-based Wiener
Filtering (AR3_IF) is applied to Robust Speech Recognition. A low complexity approach
of this algorithm is tested in presence of bathroom water noise and its performance
is compared to classical Spectral Subtraction method. Some results are presented
when training task of the speech recognition system (HTK-MFCC) is executed under
clean and noisy conditions. These results show a lower sensitivity to the presence
of water noise when applying AR3_IF algorithm inside of a speech recognition task.
Paper

PSR.9
COMPARISON OF SEVERAL PREPROCESSING TECHNIQUES FOR
ROBUST SPEECH RECOGNITION OVER BOTH PSN AND GSM NETWORKS

Chafic Mokbel, Laurent Mauuary, Denis Jouvet and Jean Monn

France Tlcom - CNET / LAA / TSS / RCP
2 av. Pierre Marzin, 22307 Lannion cedex, France
e-mail: mokbel(jouvet, monne)@lannion.cnet.fr

ABSTRACT
In this paper several preprocessing techniques used to improve speech
recognition performance are compared over both PSN and GSM networks.
Recognition experiments are conducted on a digit database in a speaker-
independent isolated-word mode in order to evaluate the performances under
within- and cross-network (PSN and GSM) conditions. Two classes of preprocessing
techniques are distinguished depending on whether they deal with additive
ambient noise or convolved perturbations. The first class preprocessing
techniques are based on spectral subtraction. In the second class, the low
frequencies of cepstral trajectories are eliminated in order to reduce convolved
disturbances. Blind equalization adaptive filtering has been proposed to reduce
channel effects. In this study, channel equalization and speech enhancement
techniques are combined and compared. Different recording conditions may be
integrated in order to increase robustness. This is done during the training
phase using HMM models with variable parameters. Recognition results are
analysed as a function of recording conditions.
Paper

PSR.10
CONSISTENT SUBSETS
IN SPEECH RECOGNITION SYSTEMS

Stefan Grocholewski
Institute of Computing Science, Poznan University of Technology
Piotrowo 3a, 60-965 Poznan, Poland
Tel: i48 (0)61 782 373; fax: +48 (0)61 771 525
grocholew@poznlv.tup.edu.pl

ABSTRACT

In the paper the method of the transformation of
the learning samples into their representatives is
presented. The proposed algorithm combines the
features of the neural nets approach, i.e. the
representatives lie near the boundaries separating
the classes, and cluster seeking approach - each
representative corresponds to the group of
elements lying close to each other. By using the
consistent subset the drawbacks of those
approaches (cluster can comprise samples from
different classes; the sophisticated network is not
appropriate in the regions where the classes
overlap) can be avoided in some cases. Several
applications in the area of speech recognition are
presented.
Paper

PSR.11
VOCABULARY INDEPENDENT ACOUSTIC-PHONETIC MODELING
FOR CONTINUOUS SPEECH RECOGNITION

L. Fissore (+), P. Laface (*), G. Micca (+), F. Ravera (+)

(+) CSELT - Centro Studi e Laboratori Telecomunicazioni
Via G. Reiss Romoli 274 - I-10148 Torino, Italy
E-Mail fissore@cselt.stet.it
(*) Dipartimento di Automatica e Informatica - Politecnico di Torino
Corso Duca degli Abruzzi 24 - I-10129 Torino, Italy
E-Mail laface@polito.it

This paper investigates the problem of defining the acoustic-phonetic unit
set for flexible vocabulary continuous speech recognition systems.
As an alternative to the classical modeling approach with biphones and
triphones, a set of stationary/transitory state units is defined that is
limited enough in number as to represent a closed set trainable once and for
all. A major benefit of these units is that inter-word transitions can easily
be taken into account.
We show that a system employing these new units favorably compares  with
respect to a baseline recognizer with Continuous Density Hidden Markov Models
of context-dependent biphones and triphones, selected through a minimal
occurrence criterion within the training database.
Paper

PSR.12
ASYNCHRONOUS INTEGRATION OF AUDIO AND VISUAL SOURCES
IN BI-MODAL AUTOMATIC SPEECH RECOGNITION

+Paul Del'eglise, Alexandrina Rogozan and Mamoun Alissali
LIUM, University of Maine
++Av. Olivier Messiaen, BP 535, 72017 Le Mans Cedex, France
Tel: +33 43.83.37.70; Fax: +33 43.83.33.66
e-mail:   deleglise@lium.univ-lemans.fr

This paper presents our work on the integration of visual data in
automatic speech recognition systems. We particularly aim at solving
two problems:

o classifiation differences for the modeling of acoustic information
(phonemes) and visual information (visemes);
o the phenomena of anticipation and retention of visemes on the
corresponding phonemes.

We developed and tested three systems, each dealing with one or
both problems and proposing a different integration strategy.  The
comparison of system performances show that some of the solutions
we propose give satisfactory results, and suggest that further work on
some others would lead to more performance improvement.

Paper

PSR.13
Title NEW TIME-FREQUENCY DERIVED CEPSTRAL COEFFICIENTS FOR AUTOMATIC
SPEECH RECOGNITION

Authors Hubert Wassner, Gerard Chollet

Affiliation IDIAP (wassner@idiap.ch, chollet@idiap.ch), ENST
(chollet@sig.enst.fr)

Abstract

The goal is to improve recognition rate by optimisation of Mel
Frequency Cepstral Coefficients (MFCCs): modifications concern the
time-frequency representation used to estimate these coefficients.
There are many ways to obtain a spectrum out of a signal which differ
in the method itself (Fourier, Wavelets,...), and in the normalisation.
We show here that we can obtain noise resistant cepstral coefficients,
for speaker independent connected word recognition.The recognition
system is based on a continuous whole word hidden Markov model. An
error reduction rate of approximately 50\% is achieved. Moreover
evaluation tests demonstrate that these results can be obtained with
smaller databases: halving the training database have small effects on
recognition rates (which is not the case with traditional MFCCs).
Paper

PSR.14
RECOGNITION OF PHONEMES FROM ESTIMATION ERRORS

L Baghai-Ravary and S W Beet

Department of Electronic and Electrical Engineering,
The University of Sheffield, Mappin Street, Sheffield, S1 3JD, UK.
Tel: (+44 ) 114 282 5409; Fax: (+44) 114 272 6391
Email: l.baghai-ravary@shef.ac.uk, s.beet@shef.ac.uk

Speech recognition systems generally use delta and delta-delta (velocity and acceleration) coefficients to
characterise the dynamics apparent in frame-based representations of speech. These coefficients can be
thought of as the errors of simple predictors.
This paper describes the use of error coefficients derived from more advanced (and accurate) forms of
prediction and interpolation. Both overall recognition accuracy and the detailed confusions observed are
compared with those of the traditional methods. The task used is speaker-independent phoneme
recognition using a subset of the TIMIT database, and four different speech representations. The error
coefficient performance on this task appears to be directly related to the robustness of the estimator used,
with the best of the new methods out-performing delta-delta coefficients by around 10%.
Paper

PSR.15

Words on Lips: How to Merge Acoustic and Articulatory Informations to
Automatic Speech Recognition

Regine Andre-Obrecht, Bruno Jacob, Christine Senac
IRIT- CNRS UMR 5055 - Universite Paul Sabatier
118, route de Narbonne, 31062-Toulouse CEDEX, France
e-mail: obrecht@irit.fr

Our work deals with the classical problem of merging heterogeneous and
asynchronous parameters. It's well known that lip reading improves the
speech recognition score, specially in noisy conditions;  so we study
more precisely the modeling of acoustic and articulatory parameters to
propose new Automatic Speech Recognition systems. We use a segmental
pre-processing, a robust unit "the pseudo-diphone" and we compare a
global HMM and a master-slave HMM. We confirm through experiments the
importance of labial features in clean and noisy environment.
Paper

REI.1

JOINT INTERPOLATION, MOTION AND PARAMETER ESTIMATION
FOR IMAGE SEQUENCES WITH MISSING DATA

Simon J. Godsill and Anil C. Kokaram

Signal Processing and Communications Laboratory, University of Cambridge
e-mail: {sjg,ack}@eng.cam.ac.uk

This paper presents a new scheme for interpolation of missing data in
image sequences, an important problem in many areas including archived
motion picture film and digital video. A unified framework for image data
modelling and motion estimation is adopted which is based on 3-dimensional
autoregressive (3DAR) models with motion correction. A fully Bayesian
methodology is implemented using the Gibbs Sampler, a method which allows
for joint estimation with respect to all of the unknowns, including the motion field.
Paper

REI.2
TITLE :
DETECTION AND REMOVAL OF LINE SCRATCHES

AUTHOR :
Anil Kokaram

AFFILIATION :
Signal Processing and Communications Group,
Engineering Department, University of Cambridge,
Trumpington St., Cambridge CB2 1PZ, England.
Tel: +44 1223 332767; Fax: +44 1223 332662
email: ack@eng.cam.ac.uk

ABSTRACT :
Line scratches are a common problem in archived motion pictures. They are
caused by the abrasion of the film material as it passes through the
projection mechanism. This paper presents a technique for detecting and
removing these line artefacts. The method employs a model of the line profile
for detection and the 2D Autoregressive model (2D AR) of the image for
interpolation.

KEYWORDS : Image Reconstruction, Line Finding, Hough Transform, Gibbs Sampling,
Autoregressive modelling, Bayesian Estimation.


Paper

REI.3
CURVED SURFACE RECONSTRUCTION USING MONOCULAR VISION

William Puech and Jean-Marc Chassery

TIMC-IMAG laboratory, Institut Albert Bonniot,
Domaine de la Merci, 38706 LA TRONCHE Cedex France,
Tel: 76549484; fax: 76549414
e-mail: William.Puech@imag.fr, Jean-Marc.Chassery@imag.fr

ABSTRACT:
In monocular vision, a priori knowledge is necessary to
perform 3D reconstruction. This paper describes how to
evaluate two out of six external parameters of a camera
in order to project an image on a curved surface
(generalized cylinder). The final aim consists of reconstructing
the model of the surface. Afterwards, with this model
we can derive a flat representation of the scene
without any distortions due to the projective geometry.
In this work based on one projected view of the scene, we
develop two methods to detect the projection of the revolution
axis of the curved surface. With this axis, we can then
extract the external parameters of a camera. The first one
is based on the derivation of a polynomial function and the
second one is based on the detection of the common normal between curves.
Paper

REI.4
IMAGE SEQUENCE RESTORATION FOR REMOVING SPACE-VARIANT MOTION BLUR

Kwan Pyo Hong, Dong Wook Kim, and Joon Ki Paik

Department of Electronic Engineering, Chung-Ang University
221 Hunksuk-Dong, Dongjak-Ku, Seoul, 156-756, Korea
Tel:+82-2-820-5300; Fax:+82-2-825-1584
e-mail: paikj@video1.ee.cau.ac.kr

An image restoration algorithm for removing motion blur, which occurs in
an image sequence or moving pictures, is proposed. More specifically, the
proposed iterative restoration algorithm adaptively reduces nonuniform
motion blur by using motion vector information from consecutive image
fields.  Motion vectors are estimated based on the well known block
match ing algorithm, and the corresponding blur model is embodied into
the point spread function, which is used to implement the iterative image
restoration algorithm.
A blur model modification method is also proposed to reduce artifacts on
the boundary area between objects with different blur patterns
Paper

REI.5
COHERENT MODEL-BASED OPTICAL RESOLUTION AND SNR

A.J. den Dekker

Delft University of Technology, Department of Applied Physics
Lorentzweg 1, 2628 CJ Delft, The Netherlands
Tel: +31 15 2781823;  Fax: +31 15 2784263
e-mail: dekker@tn.tudelft.nl

In this paper a new parameter estimation based criterion for two-point
resolution is proposed. Unlike the classical resolution criteria, the
new criterion takes account of noise and systematic errors. A resolution
limit in terms of the observations is derived. This limit depends on the
point spread function used and the degree of coherence supposed. For statistical
observations the probability of resolution as a function of the SNR is
derived. This probability can be used as a performance measure in the
assessment of optical instruments.
Paper

REII.1
DISCRETE B-SPLINE FUNCTIONS

(Koichi ICHIGE)
Doctoral Program in Engineering,
University of Tsukuba, Tsukuba, Ibaraki 305 Japan
e-mail: ichi@fmslab.is.tsukuba.ac.jp
Department of Computer and Information Sciences,
Ibaraki University, Hitachi, Ibaraki 316 Japan

A simple discrete version of B-splines is proposed.  The proposed discrete
version has different values from B-splines at the discrete points, but
it is proven that the proposed discrete version tends to B-splines when
the sampling interval goes to zero.  They can be evaluated more quickly
than the former discrete B-splines, only by RRS digital filters.
Paper

REII.2
2-D NEURAL HYBRID FILTERS USING ADAPTIVE WINDOWS AND LAYERED MEDIAN
FILTERS

Mitsuji Muneyasu, Takahiro Maeda and Takao Hinamoto

Faculty of Engineering, Hiroshima University
1-4-1 Kagamiyama, Higashi-Hiroshima 739, Japan
e-mail: muneyasu@ecl.sys.hiroshima-u.ac.jp

A new structure of 2-D neural hybrid filters composed of the
cascade connection of layered median filter, 2-D linear filter with
adaptive windows, and a neural network is developed.
The proposed filter can be used for edge-preserving smoothing of an
image under the mixed noise environment such that both white Gaussian noise
and impulsive noise exist.
The layered median filter section is composed of the cascade
connection of 1-D median filters which select the median value of
3 points.
The window sizes of 2-D linear filters are chosen so as to prevent
edges in the output image from degrading.
The parameters of the neural network are adjustable by using a learning
algorithm to adapt itself to the property of an image to be processed.
An experimental result is shown to illustrate the effectiveness of the
proposed filter.
Paper

REII.3
Title:
VECTOR MEDIAN-VECTOR DIRECTIONAL HYBRID FILTER FOR COLOR IMAGE RESTORATION

Authors:
Moncef Gabbouj and Faouzi Alaya Cheikh

Affiliation:
Signal Processing Laboratory, Tampere Univ. of Technology
P. O. BOX. 553, 33101 Tampere, Finland
Tel: + 358-31-365 3967; Fax: + 358-31-365 3967
moncef@cs.tut.fi;  faouzi@cs.tut.fi

Abstract:
In this paper we propose a new approach for multichannel signals and image
processing. This new scheme is similar to the VDF's approach, in
the way it decomposes the filtering process into direction estimation and
magnitude estimation of the output vector. While the VDF performs these two
stages sequentially; our filtering approach may execute the two stages in
parallel. This parallel structure eliminates the distorting effect of the
magnitude processing stage on the  direction estimated in the first step. And it
reduces the required time of the overall processing to the time corresponding
to the most demanding task. A further speedup factor is gained over the VDF
approach, since our algorithm does not use sorting at any stage. Simulation
results show the effectiveness of the proposed scheme in color image
restoration.
Paper

REII.4
REGULARIZED IMAGE DECONVOLUTION IN A WAVELET SCHEME.

Jean-Louis Burdeau** and Rmy Prost*, Member EURASIP

*CREATIS, Research Unit Associated to CNRS (#C5515) and
Affiliated to INSERM, Lyon, France.
INSA 502, 69621 VILLEURBANNE Cedex France.
E-mail remy.prost@creatis.insa-lyon.fr

** INT, Signal and Image Dpt, 9 rue Charles Fourier 91011 EVRY Cedex France
and CREATIS, Lyon, France.
E-mail burdeau@int-evry.fr

ABSTRACT

This paper addresses the  problem of deconvolution in a multiresolution scheme.
It results a deconvolution problem at each level of resolution. The Miller regularized
approach is used and the normal equations are solved using a constrained iterative
algorithm. Simulations show the advantages of this approach.
Paper, page 1
Paper, page 2
Paper, page 3
Paper, page 4

REII.5
A Blind Deconvolution Algorithm for Simultaneous
Image Restoration and System Characterisation.

M. Razaz and D. Kampmann-Hudson

School of Information Systems
University of East Anglia Norwich, UK
Email: mr@sys.uea.ac.uk, dmh@sys.uea.ac.uk

The restoration of a blurred image in a practical
imaging system is critically dependent on the
system point spread function. Measurement of the
point spread function is often a difficult and time
consuming process, and the measurement
environment itself is somehow artificial. Also, it is
frequently the case that an observed image and the
point spread function are not measured
simultaneously under the same conditions. An
iterative blind deconvolution algorithm is
presented here which is capable of restoring an
image without the need for an exact estimate of the
point spread function. The ideal image and the
point spread function can be estimated
simultaneously by imposing appropriate a priori
constraints. Typical experimental results are
presented and discussed.
Paper

REII.6
MULTIRESOLUTION IMAGE DECOMPOSITION WITH COMPLEX STEERABLE PYRAMIDS

G. Jacovitti*, A. Manca*,  A. Neri**

* INFOCOM Dpt., University of Rome La Sapienza, Via Eudossiana 18, 00814 Rome, Italy
** Electronics Engineering Dept., University of Rome III, Via Vasca navale 84, 00146 Rome, Italy
e-mail: neri@infocom.ing.uniroma1.it

Abstract
In this contribution we present a steerable pyramid based on  complex wavelets named Circular Harmonic Wavelets
(CHW), suited for multiscale feature-based representations. The Circular Harmonic Pyramid (CHP) performs a local
windowed Fourier analysis in polar co-ordinates around any point of the image. After a survey  on the general
properties of the CHP, we illustrate the application of the CHP to the classical problem of image restoration against
Paper

REII.7
AN ALGORITHM FOR RECONSTRUCTING POSITIVE IMAGES FROM NOISY DATA

Geoffrey de Villiers

DRA Malvern,
Malvern,
Worcestershire,
WR14 3PS, U.K.
Tel: +44 (0)1684 894750;  fax: +44 (0)1684 896502
e-mail gdv@signal.dra.hmg.gb

In this paper we describe a novel method for finding non-negative solutions
to linear inverse problems. Such problems include image reconstruction where
one is required to deconvolve a known point spread function from the image
to produce a clearer image. The method described here is related to the truncated
singular function expansion for solving linear inverse problems. The method
consists of choosing the non-negative solution with minimum 2-norm whose singular
function expansion agrees with the truncated singular function expansion solution
in its first N terms. The fact that only the first N singular function coefficients,
which are easily derived from the data, are used gives the method robustness
with respect to noise and the method is not computationally very demanding.

Published with the permission of the Controller of Her Majesty's Stationery
Office.
Paper

REII.8

Lionel Beaurepaire, Kacem Chehdi

E.N.S.S.A.T, 6 Rue de Krampont,
BP 447, 22305 Lannion cedex, France
Tel: 96-46-50-30; fax: (33) 96-37-01-99
e-mail: beaurepa@enssat.fr, chehdi@enssat.fr

This paper deals with the problem of identifying the nature of the noise
from the observed image in order to apply the processing or analysis algorithm,
whichever is the most appropriate. Here, we restrict ourselves to additive
and multiplicative noises. To identify these two kinds of noises, we propose
a new approach consisting of characterizing each class and thus, each
degraded image by a vector of five parameters. These parameters are obtained
from local statistics computed on homogeneous regions of the image.
Paper

REII.9
A MULTIRESOLUTION SPECKLE REDUCTION ALGORITHM
WITH APPLICATION TO SAR IMAGES

Carmela Galdi (1), John J. Soraghan (2)

(1) Dipartimento di Ingegneria Elettronica,
Universit degli Studi di Napoli "Federico II",
via Claudio 21, 80125 Napoli, Italy.
Tel: +39 81 7683200; fax: +39 81 7683149
(2) Signal Processing Division,
Dept. of Electronic and Electrical Engineering,
University of Strathclyde
204 George Street, Glasgow, G1 1XV, Scotland, U.K.
Fax: +44-141-5522487
e-mail: jjs@spd.eee.strath.ac.uk

Synthetic Aperture Radar images are the representation
in range and azimuth coordinates of the signal received
by a radar system exploring a portion of the earth surface.
The speckle reduction technique presented in this paper takes
advantage of the knowledge of the statistical model of the
backscattered signal to design a wavelet thresholding scheme,
appropriate for this particular type of noise.
Before the application to actual images, the algorithm validity
has been tested by comparison with the Wiener filter, performed
on random sequences generated according to the backscattering
statistical model.
Paper

REII.10
SAR IMAGES RECONSTRUCTION VIA PHASE RETRIEVAL

T. Isernia(l-2), V. Pascazio(3), R. Pierri(4), G. Schirinzi(2)

(l)Dipartimento di Ingegneria Elettronica - Universita di Napoli Federico 11
via Claudio, 21 - 80125 Napoli, Italy
tel: +39-(0)81-7683512; fax: +39-(0)81-5934448; e-mail: isernia@dieO03.dis.unina.it

(2)Istituto per l'Elettromagnetismo e i Componenti Elettronici - Consiglio Nazionale delle Ricerche
via Diocleziano, 328 - 80124 Napoli Italy
tel : +39-(0)81-5707999; fax: +39-(0)81-5705734; e-mail: schiri@irecel.irece.na.cnr.it

(3)Istituto di Teoria e Tecnica delle Onde Elettromagnetiche - Istituto Universitario Navale
via Acton, 38 - 80133 Napoli, Italy
tel: +39-(0)81-5513976; fax: +39-(0)81-5512884; e-mail: pascazio@naval.uninav.it

(4)Dipartimento di Ingegneria dell'Informazione - Seconda Universita di Napoli
via Roma, 29 - 81031 Aversa (CE), Italy
tel : +39-(0)81-5044035; fax: +39-(0)81-5045804; e-mail: pierri@uxing2.sunap.it

ABSTRACT

A new method to accurately reconstruct a Synthetic Aperture
Radar complex image starting from phase errors atfected raw
received data is presented. It is based on a phase retrieval
algorithm, and the unknown complex reflectivity is found by
minimising a proper functional using the partial phase infonnation
cfuried out by the phase corrupted raw data as the initial guess of
an iterative procedure. The method, which is capable of
compensating for both 1-D and 2-D phase errors, has been
validated on real data.
Paper

SAS.1
WAVEFORM INTERPOLATION TECHNIQUE
FOR TEXT-TO-SPEECH SYNTHESIS

Mikel Larreategui and Rolando A. Carrasco
School of Engineering, Staffordshire University
Beaconside, PO 333, ST18 ODF, Stafford, UK.
TEL: +44 1785 353366; FAX: +44 1785 353552
e-mail: mikel@staffs.ac.uk

ABSTRACT

The waveform interpolation (WI) technique has recently
been proposed by Kleijn [5][6] for speech coding
applications. However, there are no known published works
in the open literature concerning the application of the WI
method for high-quality text-to-speech (TTS) synthesis.
The original contribution of this paper is to study and
evaluate the performance of the WI technique in the context
of TTS systems.
Paper

SAS.2
IMPROVED PHONOTACTIC ANALYSIS IN AUTOMATIC LANGUAGE IDENTIFICATION

Jiri Navratil

Department of Communication and Measurement
Technical University of Ilmenau
P.O.Box 0565, 98684 Ilmenau, Germany
Tel: +49 3677 69 1145; fax: +49 3677 69 1195
e-mail: jiri.navratil@e-technik.tu-ilmenau.de

This paper presents a method for phone-dependent weighting within
phonotactic models in automatic language identification.
Based on statistical analysis of the phonetic-recognizer behaviour,
a phone confidence measure is derived and used to weight the
bigram probabilities during testing. The confidence corresponds to
the expected decoding stability of individual phones.
The proposed method was shown to improve the system performance consistently
on a three-language task. The best improvement of the error rate was
from 8.4% to 1.8% for the 45-second utterances.
Paper

SAS.3
AUTOMATIC LANGUAGE IDENTIFICATION: USING INTONATION AS A DISCRIMINATING FEATURE

V.F. Leavers, K. Wiehler, C.E. Burley

Electrical Engineering Division, Manchester University, Dover Street, Manchester,
M13 9PL, England, vfl@ipg.ph.kcl.ac.uk

Current research into automatic language identification systems
sees the problem as being related to speaker independent
speech recognition and speaker identification. In particular,
speaker indentification methods appear to outperform all
other methods and the incorporation of prosodic information has
contributed only marginally to their success.
This is a counterintuitive result suggesting that perhaps
the brute-force application of standard available pattern recognition methods
is inappropriate, not least because it ignores the linguistic cues that human
beings use so easily and efficiently. It has been proposed that
an attempt to rank parameter extraction with respect to a taxonomy of
linguistic complexity would give results more in keeping with our own
abilities to discriminate between various languages.
For example, the pressure of discrimination concerning grossly
different languages such as Mandarin Chinese and English would be low
compared to that associated with an attempt to distinguish between two
quite similar languages such as Dutch and German. The present work aims to
differentiate between the two broadest groups, tone and stress, using parameters
which best model the linguistic differences between those groups.
In particular, the supra-segmental feature of intonation is modelled as a
memory effect which can be measured using the Hurst exponent.
Paper

SAS.4
PROSODY GENERATION BY MEANS OF A SYNTACTIC APPROACH AND ITS APPLICATION
IN A TEXT TO SPEECH SYSTEM

Enzo Mumolo,  Massimo Teia
Dipartimento di Elettrotecnica, Elettronica ed Informatica
Universita' di Trieste, Via Valerio 10, 34127 Trieste, Italy
Tel/Fax: +39.40.676.3861/3460
e-mail: mumolo@univ.trieste.it

Abstract

An algorithm for modeling and generating prosody from a written text is
described in this paper. Among the several speech processing areas which
could benefit of this algorithm, in this paper we have dealt with text
to speech synthesis (TTS). An experimental evaluation of the algorithm
has been carried out and it has been shown that the naturalness of the
produced speech has greatly improved.
Paper

SAS.5
A TEXT-TO-SPEECH SYSTEM FOR THE SLOVENIAN LANGUAGE

Jerneja Gros, Nikola Pavesic, France Mihelic

Faculty of Electrical Engineering, University of Ljubljana

e-mail: jerneja.gros@fer.uni-lj.si

A text-to-speech system, capable of synthesising continuous
Slovenian speech from an arbitrary input text
is described.
The TTS system is based on the concatenation
of basic speech units, diphones, using the TD-PSOLA technique,
and no special hardware is required.
The input text is transformed into its spoken equivalent
by a series of modules. These modules,
constituting the TTS system are described in detail.
Finally, the quality of synthesised speech is assessed in terms of
acceptability and intelligibility.
Paper

SAS.6
SPEAKER RECOGNITION BASED ON A WEIGHTED ACOUSTIC DISCRIMINATION

Carmen Garcia-Mateo, Leandro Rodriguez-Linares

Departamento de Tecnologias de las Comunicaciones
Phone:34-86-812133, Fax:34-86-812116
e-mail:carmen@tsc.uvigo.es, leandro@tsc.uvigo.es}

ABSTRACT
We combine multiple-mixture single-state Markov models with phonetic classification
in order to improve the performance of a speaker recognition system.
Three broad phonetic classes: voiced frames, unvoiced frames and transitions,
are defined. We design speaker templates by the parallel connection of the
weighted outputs of three single state HMM's.
Each model corresponds with a distinct sound class  and the output weights
take into account the perceptual influences across phonetic classes.
The preliminary results show that this novel architecture outperforms its counterpart
without phonetic classification.
Paper

SAS.7
TITLE:          SPEAKER RECOGNITION WITH ARTIFICIAL NEURAL NETWORKS AND
MEL-FREQUENCY CEPSTRAL COEFFICIENTS CORRELATIONS

AUTHORS:        Roberto Amilton Bernardes Soria, Euvaldo F. Cabral Jr.

AFFILIATION:    University of Sao Paulo - DEE/EPUSP
Laboratory of Communication and Signals - LCS
CAIXA POSTAL 8174, Sao Paulo, SP, 01065-970, Brazil

ABSTRACT:
The problem addressed in this paper is related to the fact that classical statistical approach for speaker
recognition yields satisfactory results but at the expense of long length training and test utterances. An
attempt to reduce the length of speaker samples is of great importance in the field of speaker recognition
since the statistical approach, due to its limitations, is usually precluded from use in real-time applications.
A novel method of text-independent speaker recognition which uses only the correlations among MFCCs,
computed over selected speech segments of very-short length (approximately 120ms) is proposed. Three
different neural networks - the Multi-Layer Perceptron (MLP), the Steinbuch's Learnmatrix (SLM) and the
Self-Organizing Feature Finder (SOFF) - are evaluated in a speaker recognition task. The ability of
dimensionality reduction of the SOFF paradigm is also discussed.

Paper

SAS.8
IMPROVED VOCAL TRACT MODEL FOR SPEECH SYNTHESIS

Minsheng Liu, Arild Lacroix
Institut fur Angewandte Physik; University of Frankfurt
Robert-Mayer-Str.2-4; D-60325 Frankfurt am Main,Germany
e-mail:Liu@iap.uni-frankfurt.de, Lacroix@iap.uni-frankfurt.de

Speech synthesis of nasal and non-nasal speech sounds are studied on the
basis of an improved model where a nasal tract is included in the vocal tract.
The transfer function of the model is analysed. Because of the closure of the
oral tract, the three-port adaptor at the  velum is reduced to a two-port
adaptor, so that the model  parameters can be estimated by  inverse filtering
from the  speech signal. Moreover this method is applied to investigate
nasalization of vowels.
Paper

SAS.9
VOWEL-NON VOWEL CLASSIFICATION OF SPEECH USING AN MLP AND RULES

John Sirigos,		john@wcl.ee.upatras.gr
Vassilis Darsinos,		darsinos@wcl.ee.upatras.gr
Nikos Fakotakis,		fakotaki@wcl.ee.upatras.gr
George Kokkinakis,	gkokkin@wcl.ee.upatras.gr

Wire Communications Laboratory, University of Patras, 26500 Patras, Greece

ABSTRACT
In this paper we present a high precision speaker independent vowel/non vowel classifier
based on a simple feed forward MLP (Multi Layer Perceptron) and several rules. RASTA-PLP
analysis of the speech signal resulting to mel-cepstral coefficients and a formant
tracking method are used in order to provide the feature vectors for the MLP. To
train and test the system we used a part of the TIMIT database. The results indicate
that the performance of this classifier for speaker independent vowel classification
is approximately 97.25% so it can be favorably used for speaker recognition or speech
labeling purposes.
Paper

SAS.10
A WAVELET REPRESENTATION EVALUATION
FOR STOP-CONSONANTS CLASSIFICATION

Christophe Gerard, Marc Baudry, Alexandrina Rogozan
L.I.U.M., University of Le Mans
Avenue O. Messiaen, B.P. 535, Le Mans 72017 Cedex, France
Tel:  +33 4383 32 21; Fax:  +33  43 8335 65
E-mail:   gerard@lium.univ-lemans.fr

ABSTRACT
Regarding Short Time Fourier Transform based methods, stop-consonants representation
could be improved using the wavelet transform. After presenting our framework, we
describe the wavelet parameterization and the classification method. Stop consonants
are represented with pseudo-cepstral wavelet based parameters computed on a single-burst-neighbourhood-20
ms frame. Non-parametric nearest neighbours method is used. Evaluation is speaker-independent
; 1593 stop-consonants extracted from TIMIT database are evaluated. Results are
described and discussed comparatively to MFCC's (Mel Frequency Cepstrum Coefficients).
It appears that, in our field of research, wavelet gives equivalent classification
percentages. The first thing which was pointed out, is the necessity to build an
elaborated-wavelet-based-representation to get significant improvements.
Paper

SC.1
AN ATM SPEECH CODEC WITH IMPROVED RECONSTRUCTION OF LOST CELLS

Kai Clver

Institut fr Fernmeldetechnik, Technische Universitt Berlin
Einsteinufer 25, D-10587 Berlin, Germany
telephone: +49 30 314-24581; fax: +49 30 314-25799
e-mail: cluever@ftsu00.ee.tu-berlin.de

A speech codec for ATM networks is presented which includes
ATM adaptation layer functions, a voice activity detection,
and a new method for the reconstruction of lost cells. As the
cell assembly already requires a relatively high buffering
delay, only algorithms are applied which introduce small
additional delays. The reconstruction of lost cells is based
on an analysis of the LPC and pitch parameters of the speech
signal. The new waveform substitution method considerably
reduces the speech quality impairment caused by cell loss.

Paper

SC.2
MULTIMODE SPECTRAL CODING OF SPEECH FOR SATELLITE COMMUNICATIONS

Amitava Das* and Allen Gersho**

*Qualcomm Inc., 6455 Lusk Boulevard, San Diego, CA 92121.
**Dept. of Electrical & Computer Eng. University of California, Santa Barbara, CA 93106.
Tel/FAX: 805-893-2037/3262. email: gersho@ece.ucsb.edu

We present a multimode spectral coding algorithm which employs the enhanced MBE (EMBE)
spectral model and a new spectral quantization technique called transformed variable
dimension vector quantization (TVDVQ) offering good speech quality at low rate. The
EMBE model represents the short-term speech spectrum in a mode-specific way. TVDVQ
encodes the variable-dimension spectral components efficiently at low complexity. The
resulting 2.9 kb/s source coder offers good speech quality comparable to the 4.8 kb/s
CELP 1016 and the 4.15 kb/s IMBE coder. An additional 1.1 kb/s of channel coding preserves
the speech quality and intelligibility quite well with up to 2% random bit errors.
Paper

SC.3
CELP CODING BASED ON SIGNAL CLASSIFICATION USING THE

Joachim Stegmann, Gerhard Schroeder, Kyrill A. Fischer

Deutsche Telekom AG, Technologiezentrum,
Am Kavalleriesand 3, 64295 Darmstadt, Germany
e-mail: stegmann@fz.telekom.de

This paper describes a CELP speech-coding algorithm which
makes use of a specific signal classifier especially
designed for this purpose. The classification method is
based on the Dyadic Wavelet Transform (DyWT) and has
proved to be superior to common classifiers that use the
open-loop long-term prediction gain for mode selection.
The classifier's output is used for the control of several
coder parameters, such as the choice of the subframe length
and the selection of the synthesis model and the corresponding
codebooks. We designed a fully quantised coder operating at
a fixed bit rate of 4 kbit/s with a 20-ms frame. The proposed
coder improves the weighted segmental signal-to-noise ratio
(WSegSNR) by 2.3 dB on the average in comparison with a
conventional CELP coder, thereby achieving high speech quality.
Paper

SC.4
AN ALGORITHM FOR THE TRAINING OF CELP EXCITATION CODEBOOKS

Ulrich Balss, Herbert Reininger, Holger Schalk, Dietrich Wolf

Institut fuer Angewandte Physik der J.W. Goethe-Universitaet Frankfurt a.M.
Robert-Mayer-Strasse 2-4, D-60054 Frankfurt am Main, FRG
Tel: +49 69 798 28163; Fax: +49 69 798 28510
e-mail: balss@apx00.physik.uni-frankfurt.de

CELP schemes with trained excitation codebook are able to reproduce more
complex waveforms than stochastic CELP schemes. Here we present a new
algorithm for the design of trained CELP excitation codebooks which are well
adapted to the residual of speech even in transition regions. The vectors of
the excitation codebook are adapted to a training speech sequence by applying
an iterative algorithm. To obtain a high coding accuracy, the
analysis-by-synthesis error measure used during coding process is also used in
the codebook design procedure. Due to the simultaneous occurance of quantized
amplitude vector and quantized gain in the error measure, both codebooks are
optimized iteratively. The amplitude codebook vectors are designed as
subvectors of a so-called base excitation sequence by shifting their offset.
Comparative listening tests have shown that this method outperforms stochastic
CELP in objective SNR as well as in subjective quality.
Paper

SC.5
CRITICAL BAND QUANTISATION ANALYSIS FOR

Paul M. McCourt

Department of Electrical&Electronic Engineering
Queen's University of Belfast
Belfast BT9 5AH, UK
e-mail pm.mccourt@ee.qub.ac.uk

ABSTRACT
This paper presents new results on critical band masked distortion controlled
quantisation of a linear transform representation of speech. In particular,
fixed rate split vector quantisation of a critical band gain vector is
investigated. While shown to be objectively significant in meeting masked
distortion criteria, near-transparent quantisation of the critical band
gain spectrum is nonetheless achieved at 1.75 kbits/sec. The relevance
of this result is explained by a comparative interpretation of the parametric
spectral synthesis performed by current analysis-by-synthesis, multi-band
excitation and sinusoidal transform coders.
Paper

SC.6
PERCEPTUAL CODING OF SPEECH USING A FAST WAVELET PACKET TRANSFORM ALGORITHM

Benito Carnero and Andrzej Drygajlo

Signal Processing Laboratory
Swiss Federal Institute of Technology at Lausanne
CH-1015 Lausanne, SWITZERLAND
e-mail: carnero@lts.de.epfl.ch

This paper presents a new speech coding algorithm based on a fast
wavelet packet transform algorithm and psychoacoustic modeling. The
employed FFT-like overlapped block orthogonal transform allows us to
approximate the auditory critical band decomposition in an efficient
manner, which is a major advantage over previous approaches. Owing to
such a decomposition of the original signal, we make use of the human
ear masking properties to decrease the mean bit rate of the encoder.
Paper

SC.7
SUBJECTIVE PERFORMANCE OF SPECTRAL EXCITATION CODING OF SPEECH AT 2.4 KB/S

P. Lupini and V. Cuperman

School of Engineering Science, Simon Fraser University,

This paper presents a low rate speech codec (2.4 kb/s) based on a
sinusoidal model applied to the excitation signal. A frame classifier
in combination with a phase dispersion algorithm allows the same
model to be used for voiced as well as unvoiced and transitional
sounds. The phase dispersion algorithm significantly improves the
perceived quality for all frame classes resulting in more natural''
reconstructed speech. Informal MOS testing indicates that the 2.4
kb/s SEC system achieves MOS scores close to the existing 4 kb/s
standards (differences up to 0.2 on the MOS scale) and significantly
better than the existing 2.4 kb/s LPC-10 standard (difference of 1.5
on the MOS scale).
Paper

SC.8
ROBUST MULTIBAND EXCITATION CODING OF SPEECH
BASED ON VARIABLE ANALYSIS FRAME SIZES

Eric W.M. YU and Cheung-Fat CHAN

Department of Electronic Engineering, City University of Hong Kong,
Tat Chee Avenue, Kowloon, Hong Kong.
Phone: (852) 2788-7758   Fax: (852) 2788-7791
Email: eewmeyu@cityu.edu.hk   eecfchan@cityu.edu.hk

A robust technique for the coding of multiband excitation (MBE) model
parameters from a non-stationary speech segment is proposed in this
paper. The non-stationary speech segment which has an abrupt increase
in its signal energy with respect to the time is divided into 2 quasi-
stationary speech segments.  A variable analysis frame size technique
is proposed to analyze the lower energy portion and the higher energy
portion separately.  A high quality fixed 1.6 kbps variable frame size
MBE linear predictive (MBELP) speech coder was developed.
Paper

SC.9
Title
A PROTOTYPE WAVEFORM INTERPOLATION LOW BIT RATE SPEECH CODEC

Authors
Gloria Menegaz and Michele Mazzoleni

Affiliation
DE-LTS, Swiss Federal Institute of Technology
CH-1015 Lausanne, Switzerland
Tel: +39 2 66161267; fax: +39 2 66100448
e-mail: menegaz@mailer.cefriel.it
CEFRIEL, Via Emanueli 15, I-20126 Milano

Abstract
Voiced speech is characterized by a high level of periodicity.
In order to encode voiced speech with a good quality,
the correct degree of periodicity must be preserved.
The proposed coding algorithm attempts to respect such a
constraint even at low bit rates.
The method exploits the temporal redundancy
of voiced segments in order to achieve high compression rates.
Voiced speech is interpreted as a concatenation of slowly evolving
pitch-cycle waveforms. The signal is synthesized by waveform
interpolation from a downsampled sequence of pitch-cycles with
a rate of one prototype waveform per frame (20-30ms).
An original method of prototype representation, parametrization
and coding based on a proper mixed time-frequency representation
allows a high quality prototype reconstruction.
The effectiveness of such a parametrization renders it well
suited to low bit rate applications, yet maintaining a good
quality of the reconstructed signal. The method can be
combined with existing LP-based speech coders,
such as CELP, for unvoiced segments.
Paper

SC.10
QUANTIZATION OF THE LPC MODEL VV1TH THE
RECONSTRUCTION ERROR DISTORTION MEASURE

Jan 5. Erkelens and Piet M. T. Broersen
Delft University of Technology, Department of Applied Physics
P.O. Box 5046, 2600 GA Delft, The Netherlands
Tel +31 15 2781823 / +31 15 2786419
Fax +31 15 2784263
e-mail: erkelens@gtn.tudelft.nl / broersen@gtn.tudelft.nl

ABSTRACI

In Linear Predictive Coding algorithms, the codmg of the
speech signal consists of two separate stages: coding of the
LPC model and coding of the excitation. In CELP, the LPC
excitation is coded by Analysis-by-Synthesis in the
reconstruction domain, not by minimization of the error in the
LPC residual domain. Commonly used distortion measures for
quantization of the LPC spectral model are the Spectral
Distortion and the Likelihood Ratio. For small quantization
errors, they belong to a class of similar distortion measures
which express an error in the residual domain. A new spectral
distortion measure is proposed, the Reconstruction Error
Distortion measure, which expresses an error in the
reconstruction domain. Preliminary results indicate that about
five bits per frame can be gained with this new measure,
without a loss in subjective quality.
Paper

SE.1
PARAMETER IDENTIFICATION OF FREQUENCY-SELECTIVE NOISY FAST-FADING RAYLEIGH
DIGITAL CHANNELS VIA NONLINEAR YULE-WALKER-LIKE EQUATIONS

Roberto Cusani, Enzo Baccarelli
INFOCOM Dpt., University of Rome "La Sapienza", Rome, Italy
Tel. +39 6 4458589; fax: +39 6 4873300;
email: robby@infocom.ing.uniroma1.it

New procedure is proposed for the identification of data channels affected
by randomly time-variant fading. It is based on a set of nonlinear equations
employing a minimum number of lags of the observed autocorrelation function
(acf), and its solution gives the desired channel fading parameter estimates.
Better estimation accuracy is obtained in comparison with the use of classic
higher-order Yule-Walker procedure (although this latter employs a linear
equation system), in particular for small Doppler spreads and for signal-to-noise
ratios not very high.
Paper

SE.3
SUPER-RESOLUTION SPECTRUM ANALYSIS REGULARIZATION :
BURG, CAPON & AGO-ANTAGONISTIC ALGORITHMS

Frederic Barbaresco

THOMSON-CSF AIRSYS
Radar Development / Algorithms & New concepts Department (RD/RAN)
7-9, rue des Mathurins 92221 Bagneux, FRANCE
Tel : 33-(1) 40.84.20.04 ; Fax : 33-(1) 40.84.36.31
e-mail : barbareso@airsys.thomson.fr

ABSTRACT

We propose a regularized Burg algorithm, based on a frequency domain
smoothness prior constraint, which solves model order estimation problem
in case of short data records. A second algorithm deals with a recursive
eigendecomposition method from autoregressive parameters, that allows
Capon spectrum analysis regularization. Finally, we have developed a new
regularized detectors using log-likehood ratio from regularized reflection
coefficients.
Paper

SE.4
SPECTRAL ANALYSIS OF RANDOMLY SAMPLED SIGNALS

A. Ouahabi(1), C. Depollier(2), L. Simon(2), D. Kouame(1),
J.F. Roux(1) and F.Patat(1)

(1) LUSSI,GIP Ultrasons/EIT 7 Av M. Dassault BP 407 37004 TOURS Cedex France
Phone:(+33) 47 71 12 26
Fax: (+33) 47 28 95 33
e-mail: ouahabi@balzac.univ-tours.fr
(2) LAUM  URA CNRS 1101 Av. O. Messiaen, BP 535 72017 LE MANS Cedex France
Phone:(+33) 43 83 32 70
Fax: (+33) 43 83 35 20
e-mail: depol@laum.univ-lemans.fr

Abstract:
The power spectral density of randomly sampled signals is studied with
reference to fluid velocity measured by laser Doppler velocimetry.
In this paper, we propose a new method for spectral estimation of
Poisson-sampled stochastic processes. Our approach is based on polygonal
interpolation from the sampled process followed by resampling and usual
fast Fourier transform.
This study emphasizes the merit of the polygonal hold vs. the sample-and-hold.
Paper

SE.5
HIGH RESOLUTION SPECTRAL ANALYSIS USING A COMBINATION OF AN ORTHOGONAL
APPROACH AND A GENETIC ALGORITHM

Jean-Marc Vesin

Signal Processing Laboratory
Swiss Federal Institute of Technology
CH-1015 Lausanne, Switzerland
Tel: +41 21 693 3996; fax: +41 21 693 7600
e-mail: vesin@ltssg4.epfl.ch

We describe in this paper how a method for parsimonious sinusoidal
representation of signals based upon an orthogonalization technique
can be suitably modified by embedding it into a genetic algorithm.
We first describe the orthogonalizationformalism, then we present
the genetic algorithms in general and the specific form, based on a
floating-point parameter representation, that we have employed
in this work. Experiments are presented and possible extensions
are discussed.
Paper

SE.6

AN  ENHANCED METHOD FOR THE ESTIMATION OF A DOPPLER FREQUENCY

J. Crestel, M. Guitton, H. Chuberre

ENSSAT / LASTI, Universite de RENNES I
B.P. 447
22305  Lannion  (France)
Tel: (33) 96 46 56 43    Fax: (33) 96 37 01 99
e-mail:  crestel@merlin.enssat.fr

The enhanced method for the estimation of a Doppler frequency which is
dealt with aims at achieving a real time measure of the movements of a
vehicule, given an on-board configuration of microwave Radar sensors.
The prime idea is that the Doppler frequency can be assimilated to the
mean instantaneous frequency of the signal. Then this frequency is estimated
using the first moment of a quadratic time-frequency distribution. The
enhancing process of the method is involved both in a specific preprocessing
of the distribution so as to capture a reliable signal information, and
in a weighted rejection of the higher variance components, likely to be
meaningless. Simulations, as well as preliminary real tests, show probative
results.
Paper

SE.7
GABOR TRANSFORM AND ZAK TRANSFORM WITH RATIONAL OVERSAMPLING

Martin J. Bastiaans

Technische Universiteit Eindhoven, Faculteit Elektrotechniek, EH 5.33,
Postbus 513, 5600 MB Eindhoven, Netherlands,
tel: +31 40 2473319, fax: +31 40 2448375, e-mail: M.J.Bastiaans@ele.tue.nl

Gabor's expansion of a signal into a set of shifted and modulated versions
of an elementary signal is introduced, along with the inverse operation,
i.e. the Gabor transform, which uses a window function that is related to
the elementary signal and with the help of which Gabor's expansion
coefficients can be determined. The Zak transform - with its intimate
relationship to Gabor's signal expansion - is introduced. It is shown how
the Zak transform can be helpful in determining Gabor's expansion
coefficients and how it can be used in finding window functions that
correspond to a given elementary signal. In particular, a simple proof is
presented of the fact that the window function with minimum L2 norm is
identical to the window function whose difference from the elementary signal
has minimum L2 norm, and thus resembles best this elementary signal, and
that this window function yields the Gabor coefficients with minimum L2 norm.
Paper

SE.8
Title:
PARAMETER ESTIMATION OF EXPONENTIALLY DAMPED SINUSOIDS USING SECOND ORDER
STATISTICS

Authors:
K. Abed-Meraim*, A. Belouchrani**, A. Mansour***, and Y. Hua*

Affiliation:
* Department of Electrical and  Electronics Engineering, The University
of Melbourne, Parkville, Victoria 3052 Australia, a.karim@ee.mu.OZ.AU

** Department of Electrical Engineering and Computer Sciences,  The University
of California, Berkeley CA 94720, U.S.A, adel@robotics.eecs.berkeley.edu

*** LTIRF - INPG, 46 Av. Felix Viallet, 38031 Grenoble,  mansour@tirf.inpg.fr

Abstract:
In this contribution, we present a new approach for the estimation of
the parameters of exponentially damped sinusoids  based on the second
order statistics of the observations.  The method may be seen as an extension
of the minimum norm principal eigenvectors method  to cyclo-correlation
statistics domain.  The proposed method exploits   the nullity property
of the cyclo-correlation of  stationary  processes at non-zero cyclo-frequencies.
This property allows in a pre-processing step to get rid from  stationary
existing higher order statistics based approaches:  (i) First it deals
only with second order statistics which require generally few samples
in contrast to higher-order methods, (ii) it deals either with Gaussian
and  non-Gaussian additive noise, and (iii) also deals either with white
or temporally colored (with unknown autocorrelation sequence) additive
noise. The effectiveness of the proposed method is illustrated by some
numerical simulations.
Paper

SE.9
SUBSPACE-BASED PARAMETER ESTIMATION OF SYMMETRIC NON-CAUSAL AUTOREGRESSIVE
SIGNALS FROM NOISY MEASUREMENTS

Petre Stoica and Joakim Sorelius

Systems and Control Group, Uppsala University P.O. Box 27, S-751 03
Uppsala, Sweden; Tel: +46 18 183074; fax: +46 18 503611; e-mail:
js@syscon.uu.se

The notion of
Symmetric Non-causal Auto-Regressive Signals (SNARS) arises in several,
mostly spatial, signal processing applications. In this paper we
introduce a subspace fitting approach for parameter estimation of
SNARS from noise-corrupted measurements. We show that the subspaces
associated with a Hankel matrix built from the data covariances
contain enough information to determine the signal parameters in a
consistent manner. Based on this result we propose a MUSIC (MUltiple
SIgnal Classification)-like methodology for
parameter estimation of SNARS. Compared with the methods previously
proposed for SNARS parameter estimation, our SNARS-MUSIC
approach is expected to possess a better trade-off between
computational and statistical performances.
Paper

SE.10
AUTOREGRESSIVE MODELLING OF IRREGULARLY-SAMPLED DATA

R.J.Martin

GEC Hirst Research Centre, Elstree Way, Borehamwood, Herts WD6 1RX, UK
R.Martin@hirst.gmmt.gecm.com

We shall discuss how to reformulate AR modelling in terms of a stochastic differential
equation, and thence how to generalise the notion of prediction to irregular sampling.
This gives rise to spectral estimation and FIR filtering methods for irregularly-sampled
data. We also present an extension of Shannon's theorem for the missing data problem.
Paper

SP.1
Title:
NONLINEAR PREDICTION OF SPEECH SIGNALS

Author:
Martin Birgmeier

Affiliation:
Department of Communication and Radio Frequency Engineering
Vienna University of Technology
Gusshausstrasse 25/E389
A-1040 Vienna
Austria

Phone:
(+43 1) 58801 x 3661

Fax:
(+43 1) 5870583

e-mail:
Martin.Birgmeier@nt.tuwien.ac.at

Abstract:
In this paper, we compare the capabilities of various forms of
radial basis function networks as nonlinear short-term predictors
for speech signals representing sustained utterances of German
vowels. We use RBF and RBF-AR network architectures, trained
using a standard algorithm or alternatively the extended Kalman
filter (EKF) algorithm, and linear least squares predictors.
We also look at cascaded forms of linear/nonlinear predictors.
We evaluate both prediction gain and spectral flatness measure
of the residual. The results indicate: The RBF-AR structure is
the most powerful, EKF training yields better results than
standard training for RBF networks, and a non-cascaded RBF-AR
predictor produces results superior to cascaded predictors.
Paper

SP.2
NONLINEAR FORMANT-PITCH PREDICTION USING RECURRENT NEURAL NETWORKS

Department of Electrical and Electronics Engineering
Eastern Mediterranean University, Gazi Magosa, Mersin-10, Turkey
Tel: +90 (392) 366 65 88; Fax: +90 (392) 366 44 79;
e-mail: evaroglu@salamis.emu.edu.tr

ABSTRACT
In this study, a parallel structure is proposed for  the nonlinear formant
and pitch prediction of speech signals using Recurrent Neural Networks (RNN)
The well known Real Time Recurrent Learning (RTRL) algorithm is used as the
learning algorithm.  Its performance is evaluated in terms of the mean-square
error and sensitivity to pitch errors through extensive computer simulations
and compared to the combined formant-pitch RNN predictor and to the linear
predictor.
Paper

SP.3
SPEECH ENHANCEMENT FOR HEARING AIDS

Douglas R. Campbell

Department of Electrical and Electronic Engineering, University of Paisley,
High Street, Paisley, Scotland, UK, PA1 2BE
Tel/Fax: +44 (0)141 848 3400/3404, email: d.r.campbell@paisley.ac.uk

ABSTRACT
The performance of hearing aids in noisy reverberant surroundings remains a
major source of complaint and discomfort to wearers. Given the current
capabilities and pace of development in microelectronics, the major problem is
to find successful speech enhancement schemes. Binaural unmasking experiments
demonstrate an enhancement advantage, due to binaural correlation properties,
which can lower the hearing threshold in noise and there is evidence that this
may operate in frequency sub-bands. The performance is presented of an adaptive
sub-band noise cancellation scheme which supports the possibility of performing
"binaural unmasking" outwith the body, and is shown to be capable of
out-performing a standard noise-cancellation scheme in the presence of
reverberation.
Paper

SP.4
ON SPEECH ENHANCEMENT ALGORITHMS BASED ON THE MMSE ESTIMATION

Pascal SCALART1, Jozue VIEIRA FILHO2,3, Jos GERALDO CHIQUITO3

1FRANCE TELECOM - CNET LAA/TSS/CMC
Technopole ANTICIPA, 2 Avenue Pierre Marzin, 22307 Lannion Cedex, FRANCE
Ilha solteira- SP, BRAZIL

E-mail : scalart @lannion.cnet.fr

This paper addresses the problem of single microphone frequency domain
MMSE noise reduction technique for speech enhancement in noisy environments.
We first analysed asymptotic performance of the MMSE estimate and compared
these results with the Wiener filter. Practical implementation of the
MMSE filter is then presented. Comparisons between optimal and practical
behaviour of the MMSE filter demonstrate that an effective improvement
in the noise reduction process can be gained if greater attention is given
to the these estimators.
Paper

SP.5
EVALUATION OF DIGITAL HEARING AID ALGORITHMS ON WEARABLE SIGNAL PROCESSOR SYSTEMS

Uwe Rass, Gerhard H. Steeger

Georg-Simon-Ohm-Fachhochschule, FB NF
PO-Box 210320, D-90121 Nuernberg, Germany
Tel: +49-911-5880-147, Fax: +49-911-5880-109
e-mail: rass@nf.fh-nuernberg.de, steeger@nf.fh-nuernberg.de

ABSTRACT
The benefit of hearing aid algorithms in everyday life can hardly be estimated
from results obtained in the laboratory. Extensive field tests with many hearing
impaired subjects are necessary to evaluate these processing schemes. A wearable
digital hearing aid prototype is described which was developed specifically for
that purpose. It is based on a fixed-point digital signal processor. This unit
enables the testing of even highly sophisticated algorithms, with a changing
interval of the accumulator pack of 10 hours. As application examples, a very
flexible three channel dynamic compression algorithm and a binaural processing
scheme for enhancing speech signals in noisy and reverberant environments are
described. Application of 20 units in 3 European clinics has been started recently.
Paper

SP.6
REDUCED-RANK NOISE REDUCTION: A FILTER-BANK INTERPRETATION

Soeren Holdt Jensen (1) and Per Christian Hansen (2)

(1) CPK, Aalborg University, Fredrik Bajers Vej 7, DK-9220 Aalborg OEst, Denmark.
E-mail: shj@cpk.auc.dk
(2) UNI-C, Building 304, Technical University of Denmark, DK-2800 Lyngby, Denmark.
E-mail: Per.Christian.Hansen@uni-c.dk

The key step in reduced-rank noise reduction algorithms is to approximate a
matrix by another one with lower rank, typically by truncating a singular
value decomposition (SVD). We give an explicit and closed-form derivation
of the filter properties of the rank reduction operation and interpret this
operation in the frequency domain by showing that the reduced-rank output
signal is identical to that from a filter-bank whose analysis and synthesis
filters are determined by the SVD. Our analysis includes the important
general case in which pre- and dewhitening is used.
Paper

SP.7
SPEAKER LOCALIZATION
AND ITS APPLICATION TO TIME DELAY ESTIMATORS
FOR MULTI-MICROPHONE SPEECH ENHANCEMENT SYSTEMS

Martin Drews

Institut fuer Fernmeldetechnik, Technische Universitaet Berlin
Einsteinufer 25, D-10587 Berlin, Germany
phone: +49 30 31424573, fax: +49 30 31425799
e-mail: drews@ftsu00.ee.tu-berlin.de

ABSTRACT
A time delay estimator for a multi-microphone speech enhancement system with
16 microphones is presented. It is based on a generalized cross-correlator and
an improved peak detector. The problems associated with delay estimation in
noisy speech signals are solved by performing a speaker localization and
a plausibility check of the time delays derived from the speaker position. By
applying these techniques to the time delay estimator, a significant reduction
of the computational load is achieved, and the TDOA estimation errors are
reduced.


Paper

SP.8
A WIDE-BAND SPEECH-MODEL PROCESS AS A TEST SIGNAL

M.R. Serafat and U. Heute

Institute for Network & System Theory, University Kiel, Germany
Tel: +49 431 77572 401, Fax: +49 431 77572 403,
E-Mail: res@techfak.uni-kiel.d400.de

some of the major problems in objective quality assessment of speech coding
systems or in testing other adaptive speech transmission systems are the
speaker dependence, reproducibility, and the comparability of the measurement
results, if natural speech is used as the test signal. This problem can be
avoided by using suitable speech-model processes.
In this paper, we present a wide-band speech--model process, which includes the
same long- and short-time characteristics as natural speech. The controlling
part of the generator of this process involves several trained Markov chains
(mc) to adapt the time-varying properties of the process to those of natural
speech. Furthermore, special care is taken of the necessary probabilty density
function (PDF) asymmetries, because the natural wide-band speech has an
asymmetric PDF.
Paper

SP.9
QUADRATIC CLASSIFIER WITH SLIDING TRAINING DATA SET IN ROBUST RECURSIVE
IDENTIFICATION OF NON-STATIONARY AR MODEL OF SPEECH

Milan Markovic

Institute of Applied Mathematics and Electronics
Kneza Milosa 37
Yugoslavia
fax: 381-11 324-8681
e-mail: emarkovm@ubbg.etf.bg.ac.yu

ABSTRACT

In this work, a robust recursive procedure based on WRLS algorithm with VFF
and a quadratic classifier with sliding training data set for identification
of non-stationary AR model of speech production system is proposed.
Experimental analysis is done according to the results obtained in analyzing
speech signal with voiced and mixed excitation segments. Presented experimental
results justify that two main problems of LPC speech analysis, non-stationarity
of LPC parameters and non-appropriateness of AR modeling of speech
(particularly on the voiced frames), can be solved by using the proposed robust
procedure.
Paper

SR.1
A New Training Algorithm For Hybrid HMM/ANN Speech Recognition Systems

Herve Bourlard, Yochai Konig, Nelson Morgan, and Christophe Ris

Faculte Polytechnique de Mons - TCTS, 31 Bld. Dolez, B-7000 Mons, Belgium.
International Computer Science Institute, 1947 Center Street, Suite 600, Berkeley,
CA 94704, USA.
Email: bourlard@tcts.fpms.ac.be

In this paper, we briefly describe REMAP, an approach for the training and estimation
of posterior probabilities, and report its application to speech recognition. REMAP
is a recursive algorithm that is reminiscent of the Expectation Maximization (EM)
algorithm for the estimation of data likelihoods. Although very general, the method
is developed in the context of a statistical model for transition-based speech recognition
using Artificial Neural Networks (ANN) to generate probabilities for Hidden Markov
Models (HMMs).
In the new approach, we use local conditional posterior probabilities of transitions
to estimate global posterior probabilities of word sequences. As with earlier hybrid
HMM/ANN systems we have developed, ANNs are used to estimate posterior probabilities.
In the new approach, however, the network is trained with targets that are themselves
estimates of local posterior probabilities.
Initial experimental results support the theory by showing an increase in the estimates
of posterior probabilities of the correct sentences after REMAP iterations, and
a decrease in error rate for an independent test set.
Paper

SR.2
AUTOMATIC DISCOVERY OF WORD CLASSES THROUGH LATENT SEMANTIC ANALYSIS

Jerome R. Bellegarda, John W. Butzberger, Yen-Lu Chow, Noah B.
Coccaro, Devang Naik

Interactive Media Group, Apple Computer, Cupertino, California 95014, USA
(jerome @ apple.com)

A new approach is proposed for the automatic discovery of word classes
in a given vocabulary.  The method is based on a paradigm first
formulated in the context of information retrieval, called latent
representation of each word in a suitable vector space, where familiar
clustering techniques can be applied.  The resulting word classes are
intuitively satisfactory, and lead to a language model whose
predictive power, as measured by perplexity, compares favorably with a
conventional bigram's.  Because its semantic nature, this approach may
prove useful as a complement to syntactically-oriented class-based
n-gram techniques.

Paper

SR.3
CONTINUOUS SPEECH RECOGNITION USING A NEW NEURAL NETWORK WITH TWO DIFFERENT STRUCTURES

Noriyuki Ohtsuki+, Yoshikazu Miyanaga++, and Koji Tochinai++

+Department of Information Engineering, Kushiro National College of Technology
Kushiro-shi 084, Japan.
E-mail ohtsuki@kushiro-ct.ac.jp
++Division of Information Media Engineering, Faculty of Engineering Hokkaido University,
Sapporo-shi 060, Japan.
Tel. +81-11-706-6534, FAX. +81-11-709-6277
E-mail miyanaga@hudk.hokudai.ac.jp

Abstract
This report proposes a continuous speech recognition method using a new neural
network which has two different structures.  This method is able to recognize
time-varying speech phonemes.  The new neural network in this method consists
of a self-organized clustering network and a multi-layered neural network.
The self-organized clustering network extracts some characteristics of speech
in spectrum domain.  The multi-layered neural network finds the time-varying
characteristics of speech.  From some experimental results,  this report shows
that the system is quit suitable for speech recognition.
Paper

SR.4
SPEECH RECOGNITION WITH A NEURAL NETWORK TRACE-SEGMENTATION

Euvaldo F. Cabral Jr.

So Paulo University, Polytechnic School, Department of Electronic Engineering
So Paulo - SP - Brazil
Tel: +55 11 818-5267; Fax: +55 11 818-5718
email: euvaldo@lcs.poli.usp.br

ABSTRACT
Trace-segmentation (TS) is a method for non-linear time-normalization of
a sequence of speech representation frames prior to recognition of the
sequence. It has been shown in a recent work [1] that an Individual Trace-
Segmentation (ITS), i.e. a  separate segmentation of the trajectory
described by each individual coefficient in the speech frame leads to a
much improved recognition which exceeds the performance provided by
DTW recognition on the same database.
This paper describes a follow on work on the ITS technique where a Multi-
layer Perceptron has been used to perform an internal mapping in the
original ITS input space in order to provide a tighter set of clusters of the
speech sequences. This novel technique is called Neural Network Trace-
Segmentation (NNTS) and has produced a significant improvement on the
ITS original performance.
Paper

SR.5
RECOGNITION OF VOICED SPEECH FROM THE BISPECTRUM.

Delopoulos, Anastasios
Rangoussi, Maria
Andersen, Janne.

National Technical University of Athens,
Dept. Of Electrical and Computer Engineering,
Division of Computer Science,
9 Iroon Polytechneioy str.
ATHENS, GR-15780, GREECE

Recognition of voiced speech phonemes is addressed in this paper using features
extracted from the bispectrum of the speech signal. Voiced speech is modeled
as a superposition of coupled harmonics, located at frequencies that are
multiples of the pitch and modulated by the vocal tract.
For this type of signal, nonzero bispectral values
are shown to be guaranteed by the estimation procedure employed.
The vocal tract frequency response is reconstructed from the bispectrum on
a set of frequency points that are multiples of the pitch.
An AR model is next fitted on this transfer function. The AR coefficients are
used as the feature vector for the subsequent classification step.
Any finite dimension vector classifier
can be employed at this point. Experiments using the LVQ neural classifier
give satisfactory classification scores
on real speech data, extracted from the DARPA/TIMIT speech corpus.
Paper

SR.6
Extraction of LP-Based Features from  One-Bit Quantized Speech
Signals for Recognition Purposes

M.Felici, A.Ferrari, M.Borgatti, R.Guerrieri

D.E.I.S - University of Bologna
Viale Risorgimento, 2
40136 Bologna - ITALY
{mfelici, aferrari, mborgatti, rguerrieri}@deis.unibo.it

A simplified fixed-point computation of cepstral coefficients, based on
linear predictive analysis and infinite clipping of speech signals, is
described.
The autocorrelation function of the clipped signal is directly
used to compute the linear predictor coefficients.
The performance of an isolated word recognition system based on these
coefficients is presented and compared with a system which uses standard
linear predictive cepstral features.
The results show that these coefficients can be efficiently used for
small dictionary speech recognition systems and, since the analog-to-digital
conversion can be avoided, they are suitable for a low-voltage and low-power
hardware implementation.
Paper

SR.7
BLIND EQUALIZATION FOR ROBUST TELEPHONE BASED SPEECH RECOGNITION

Laurent MAUUARY

e-mail: mauuary@lannion.cnet.fr
France Telecom, Centre National d'etudes des telecommunications,
CNET/LAA/TSS/RCP,
Technopole Anticipa, 2, avenue Pierre Marzin
22307 LANNION, FRANCE

ABSTRACT
An adaptive filter in a blind equalization scheme has recently been proposed
in order to reduce telephone line effects for speech recognizers. This paper
presents the principles of this filter and describes the implementation of a
circular-convolution frequency domain adaptive filter in the blind equalization
scheme. The property of a constant long-term speech spectrum helps to compute
the gradient used for the adaptation of the weights. However, using this
property in a straightforward manner results in a crude implementation of this
filter. Alternative computations of the standard stochastic gradient algorithm
are therefore evaluated. On the basis of the speech recognition results
obtained from different speaker independent telephone databases, this filter
proves to be efficient for the channel equalization task.
Paper

SR.8
Connected Word Recognition in Extreme Noisy Environment using Weighted
State Probabilities (WSP).

T. Vaich and A. Cohen

Recognition of continuous speech in extreme noisy environments is a difficult
task. A novel algorithm is suggested to enhance the performance of  recognition
in very low SNRs. The left to right HMM Weighted State Probabilities (WSP)
method considers not only the probability of getting the given observation
sequence, but also the pattern of states probabilities. On a ten digits
(Hebrew) recognition task, with SNR of 10 db, the WSP has improved recognition
results from 0% to 50%. It is suggested to apply the method, in conjunction
with PMC enhancement algorithm, to very low SNR word spotting systems.
Paper

SR.9
HANDLING DISYNCHRONIZATION PHENOMENA WITH HMM IN CONNECTED SPEECH

Pierre Jourlin

Laboratoire d'Informatique
C.E.R.I
339, Chemin des Meinajari\es
BP 1228
84911 Avignon Cedex 9
France
Tel: +33 90 84 35 35
fax: +33 90 84 35 01
e-mail: jourlin@univ-avignon.fr

Anticipation and retention phenomena between the different phonatory organs
have been widely studied in the speech perception and production domain.
However, few automatic speech recognition systems are able to handle them.
In this paper, we define a product of valuated transitions automata handling
these difficulties.
Then, we use such automata in a recognition system based on HMM.
This method is evaluated in two different contexts : bimodal and unimodal
speech recognition.
The results show an improvement for the the product model against a synchronous
one of 1.9% in the bimodal field and of 1.2% in the unimodal one.
Paper

SR.10
STATISTICAL LIP MODELLING FOR VISUAL SPEECH RECOGNITION

Juergen Luettin (1,2), Neil A. Thacker (1) and Steve W. Beet (1)

(1) Dept. of Electronic and Electrical Engineering
University of Sheffield
Sheffield S1 3JD, UK
(2) IDIAP
CP 592, 1920, Martigny, Switzerland
Luettin@idiap.ch, N.Thacker@shef.ac.uk, S.Beet@shef.ac.uk

ABSTRACT
features extracted from grey level image sequences of the speaker's lips.
Active shape models are used to track the lip contours while visual speech
information is extracted from the shape of the contours. The distribution
and temporal dependencies of the shape features are modelled by continuous
density Hidden Markov Models. Experiments are reported for speaker independent
recognition tests of isolated digits. The analysis of individual feature
components suggests that speech relevant information is embedded in a
low dimensional space and fairly robust to inter- and intra-speaker variability.
Paper

SSP.1
ANALYTICAL LINKS BETWEEN STEERING VECTORS AND EIGENVECTORS

Nadge THIRION, Jrme MARS, Jean-Louis LACOUME
CEPHAG-ENSIEG, BP 46, rue de la Houille Blanche,
38402 ST MARTIN D'HERES Cedex France
Tl/Fax: (33) 76.82.64.21 / (33) 76.82.63.84
e-mail: thirion@cephag.observ-gr.fr

We consider the problem of separation of convolutive mixtures of wideband
signals impinging on an antenna of sensors focusing on the case of interfering
seismic waves. We are looking at the spectral matrix filtering method.
The analytical study of its resolving power, makes it possible for us
to theoretically justify its use but even to explain its deficiencies
in difficult context (waves of very close energies or/and too near slowness
for instance). But first, this question induces us to discuss on the links
between two basis: the eigenvectors one and the steering vectors one.
Paper

SSP.2
SEPARATION OF SEISMIC SIGNALS: A NEW CONCEPT BASED ON A BLIND
ALGORITHM

Nadge THIRION *, Jrme MARS *, Jean-Luc BOELLE **
*	CEPHAG-ENSIEG, BP 46, rue de la Houille Blanche,
38402 ST MARTIN D'HERES Cedex France
Tl/Fax: (33) 76.82.64.21 / (33) 76.82.63.84
e-mail: thirion@cephag.observ-gr.fr
**	Socit Elf-Aquitaine CSTJF Avenue Larribeau, 64018 PAU Cdex, France

In geophysical operations, the aims of signal processing are the separation
and the identification of waves to get a better understanding of the onshore.
The limits of the usually used techniques may appear when waves are too close
in terms of energies or/and slowness. We propose an alternative via a blind
algorithm that exploits some of the concepts of blind separation of sources.
The performances of such an approach are illustrated on field data.
Paper

SSP.3
MULTICHANNEL DISTANCE FILTERING OF SEISMIC ELECTRIC SIGNALS

G. Economou, A. Ifantis*, D. Sindoukas

University of Patras, Physics Department, Electronics Laboratory, GR-26110 Patras,
GREECE.
Tel.: +30 61 997463, FAX: +30 61 997456, email: economou@physics.upatras.gr
*- Technological Educational Institute of Patras, Dept. of Electrical Engng., Patras 26334.

ABSTRACT
A novel type of distance weighted multichannel filter is used to filter correlated multichannel
1-D seismic electric signals. These signals are weak, short time variations of the geoelectric
field occurring prior to an earthquake. The new filters use intersample distances to compute
coefficients. Both vector and componentwise correlation is utilised in the computation. The
new composite distance filters preserve better, sharp edges and correlated signal features
while at the same time possess very good noise suppression properties.
Paper

SSP.4
HIGHER ORDER STATISTICS APPLIED TO WAVELET IDENTIFICATION OF MARINE SEISMIC
SIGNALS

Mohammed Boujida & Jean-Marc Boucher

Tlcom Bretagne, Dpartement Signal et Communications
BP 832, 29285 BREST Cedex, FRANCE
Tel : 98 00 13 57, Fax: 98 00 10 12, E-mail : JM.Boucher @enst-bretagne.fr

ABSTRACT

The purpose of this paper is to present the use of higher order statistics
to solve the blind identification problem of reflection seismic data.
We develop and compare some non-parametric and parametric methods based
on higher order statistics. To compare these methods, non-minimum phase
wavelet and non-gaussian reflectivity function are simulated. They are
then applied to real data of high resolution marine seismic reflection.
Paper

SSP.5
FRESNEL RAYS AND RESOLUTION OF TOMOGRAPHIC IMAGING

Claudio Chiaruttini
D.I.N.M.A., University of Trieste,
via Valerio, 10, I-34127 Trieste, Italy
tel: +39 40 676 7157; fax: +39 40 676 3497
e-mail: chiaruttini@univ.trieste.it
Alessandro Pregarz and Enrico Priolo
Osservatorio Geofisico Sperimentale (OGS), Trieste, Italy

Ray-theoretic travel-time tomography assumes an infinite signal bandwidth.
When this condition is not met, energy propagates from source to receiver
along Fresnel rays of finite cross-section, instead of infinitely thin
mathematical rays.  We use approximate analytical solutions of the weak
scattering problem and numerical modelling of the full wave equation to
discuss the resolution of bandlimited records.  The setting of the
numerical simulations is illustrative of a cross-well seismic experiment.
We show that bandlimited travel-time data suffer an unexpected loss of
resolution just along the mathematical ray.  Nevertheless, this loss can
be fully recovered including signal amplitude in an inversion procedure.
We also discuss the problem of time picking, and show that, to be
consistent with the weak scattering assumption, arrival time must be
estimated at the signal peak.
Paper

VCI.2
A TESTBED FOR THE EVALUATION OF MPEG VIDEO TRANSMISSION ON ATM NETWORKS

Christian J. van den Branden Lambrecht* and Andrea Basso+

*Signal Processing Laboratory, Swiss Federal Institute of Technology, CH-1015
Lausanne, Switzerland, vdb@lts.de.epfl.ch, http://ltswww.epfl.ch/~vdb/
+Telecommunications Laboratory, Swiss Federal Institute of Technology, CH-1015
Lausanne, Switzerland, basso@tcom.epfl.ch, http:/tcomwww.epfl.ch/

Most of the new broadcasting and multimedia applications intensively
rely on networked video. The current trend for distributing digital
streams on ATM networks. End-to-end testing of such communication
systems is very important and requires robust testing methodologies
that are capable of evaluating both coding and transmission errors.
This paper proposes a complete architecture for doing so. The system
is entirely automatic, relies on synthetic test patterns and
estimates the subjective quality of video coding and network
transmission.
Paper

VCI.3
A PERFORMANCE MODEL FOR THE MPEG CODER

G. Calvagno, G.A. Mian, A. Moro, R. Rinaldo

Dipartimento di Elettronica e Informatica
Tel: +39-49-827 7731, Fax: +39-49-827 7699, E-mail: calvagno@dei.unipd.it

Abstract
The MPEG video coding standard provides the syntax and semantics of
bit streams representing compressed video. The underlying algorithm
uses block matching motion compensation and block based DCT, with
run-length coding of the quantized coefficients. It is important to
derive models that allow to predict, for a given input sequence, the
algorithm performance in terms of quality versus bit rate. In this
work, we show that a simple model can be used to this purpose, despite
the complexity of the overall MPEG algorithm. The model can be
conveniently used to determine the quantizer parameters that give a
desired quality or bit rate. For instance, in buffer control, it is
necessary to precisely adapt the input rate to the buffer content in
order to prevent overflow and underflow.
Paper

VCI.4
FEED-FORWARD BUFFERING AND RATE CONTROL BASED ON
SCENE CHANGE FEATURES FOR MPEG VIDEO CODER

Yoo-Sok Saw, Peter M. Grant, and John M. Hannah

Dept of Electrical Engineering, University
of Edinburgh, Edinburgh, EH9 3JL, UK.
Tel: +44 131 6505655; fax: +44 131 650 6554
e-mail: ys@ee.ed.ac.uk

Video traffic management has been a challenging task in the
fields of network management and multi-media communication. Transmission
buffering is widely used to smooth bursty traffic and maintain a steady
traffic level by adapting the incoming source traffic to the buffer.
This paper describes an efficient adaptive buffering scheme which is
based on feed-forward control to adaptively handle the non-stationary
nature of bursty video traffic. The performance of a series of
quantisation scale mapping curves is presented in terms of occupancy
and video quality.
Paper

VCI.5
TREE-STRUCTURED LATTICE VECTOR QUANTIZATION

Vincent Ricordel and Claude Labit

IRISA/INRIA Rennes, Campus de Beaulieu, 35042 Rennes Cedex, France
e-mail: ricordel@irisa.fr, labit@irisa.fr

We have already introduced a new vector quantizer (VQ) for the compression of
digital image sequences. Our approach unifies both efficient coding methods :
a fast lattice encoding and an unbalanced tree-structured codebook design
according to a distortion vs. rate tradeoff. This tree-structured lattice VQ
is based on the hierarchical packing of embedded truncated lattices.
Now we investigate the determination of the most efficient lattice
respectively to this method. We also describe a fast test which permits to
detect the input vectors whose norm is above than the maximum allowed by the
TSLVQ. Finally we analyse experimental results applied to image sequence with
our VQ taking place in a region-based coding scheme for a videophone
application.
Paper

VCI.6
Improving bit-rate and quality control for MPEG-2 video sources

Giancarlo Cicalini*, Lorenzo Favalli*, Alessandro Mecocci**

*Universit di Pavia,- Dipartimento di Elettronica via Ferrata, 1, I-27100
Pavia (PV) Italy;
Tel: +39-382-505923; fax: +39-382-422583; e-mail: lorenzo@comel1.unipv.it

**Universit di  Siena,- Facolt di Ingegneria; via Roma, 77, I-53100
Siena (SI), Italy
tel: +39-577-2636041 fax: +39-577-263602; e-mail: mecocci@comel1.unipv.it

Abstract.

In video compression techniques, it is very important to implement the
most efficient bit allocation strategy in order to achieve the best quality
with the minimum number of bits. This paper presents a new feedback/feedforward
controller, for MPEG-2 coding, that dynamically tunes the quantization
parameters by analysing the image sequence from a psycovisual point of
view. The analysis is carried out on an 8x8 pixels block basis to determine
the visual characteristic of each macroblock. This pre-analysis classifies
macroblocks and assigns quantization parameters to them according to a
proposed scale measuring their visual relevance. A post-analysis procedure
provides the final tuning. The system generates images with higher quality
with respect to the standard Test Model 5.
Paper

VCI.7
CELL DELAY VARIATION PERFORMANCE OF CBR AND VBR MPEG-2 SOURCES
IN AN ATM MULTIPLEXER

Javier Zamora, Dimitris Anastassiou and Kand Ly
Department of Electrical Engineering
and Image Technology for New Media Center
Columbia University, New York, NY 10027, USA
e-mail: javier@ee.columbia.edu

Video services require specific constraints regarding the delay variation or
jitter experienced when they are transmitted in packet networks such as ATM.
This delay component is mainly generated in multiplexing processes and it has a
direct impact on the final QoS. In this paper the jitter issue is addressed in
the environment of a video server connected to an ATM Network. Both CBR and VBR
MPEG-2 streams are considered as traffic sources. For each video source its
delay variation is studied using first order and second order statistics such
as jitter variance and GCRA, respectively. We study several traffic scenarios,
where correlation between video sources is considered . Finally the obtained
results are compared with the M+D/D/1 model.
Paper

VCI.8
A TEMPORAL MODE SELECTION IN THE MPEG-2 ENCODER SCHEME

Laurent Piron

Signal Processing Laboratory
Swiss Federal Institute of Technology
CH-1015 Lausanne, Switzerland
Tel: +41 21 693 2605; fax +41 21 693 7600
e-mail: piron@ltssg2.epfl

This paper deals with the mode decision in an MPEG-2 framework. An
algorithm for mode decision is introduced. This algorithm is based on
a rate-distortion criterion and takes into account the temporal
dependency of the frames. This approach can allow a quality gain of
more than one dB compared to the Test Model 5 (TM5) mode decision
algorithm.
Paper

VCI.9
REGION BASED CODING SCHEME WITH
SCALABILITY FEATURES

Olivier Egger, Frank Bossen, and Touradj Ebrahimi
Signal Processing Laboratory
Swiss Federal Institute of Technology at Lausanne
CH-1015 Lausanne, Switzerland
Email: egger@lts.de.epfl.ch

ABSTRACT

In order to satisfy the needs of new applications in a
multimedia environment the problem of object-oriented
coding has to be addressed. In this paper two main ap-
proaches are presented to tackle this problem. First,
an algorithm for shape coding is presented. It is based
on a chain coding algorithm where powerful modeling
techniques are used to increase the compression ratio.
Second, an algorithm for interior coding is described.
It is based on an arbitrarily-shaped subband transform
followed by a generalized embedded zerotree wavelet al-
gorithm. It is shown in the paper that it achieves good
compression results and has additional properties such
as supporting arbitrarily-shaped regions, being compu-
tationally efficient, keeping the same dimensionality in
the transformed domain, allowing perfect reconstruction
and an intrinsic rate control mechanism. The presented
results show that the two algorithms build an efficient
basis to design object-oriented video coding schemes.
Paper

VCI.10
A MODIFIED MPEG-1 SYSTEM BASED ON GENLOT

S. H. Oguz, T. Q. Nguyen and Y. H. Hu

1415 Johnson Drive, Madison, WI 53706 U.S.A.
Tel: 1 608 2655739; Fax: 1 608 2654623
e-mail: oguz@cae.wisc.edu, nguyen@ece.wisc.edu, hu@engr.wisc.edu

In  this study,  a modification to  ISO MPEG-1 and MPEG-2 digital
video  coding  standards is proposed  and  preliminary results on
its  performance  are  reported. The  proposed  modification aims
to  improve  the visual quality  of  MPEG-1 and MPEG-2  coding at
medium-to-low   bit-rate  regimes  by  eliminating  the  blocking
effect  caused  by  the  Discrete  Cosine Transform. This goal is
achieved  without  introducing a  significant change  in the MPEG
hierarchy  and   algorithm.  The  theory  of   Lapped  Orthogonal
Transforms  which  constitutes a rather  recently introduced tool
for  block  transform coding  suggests  that they  can reduce the
blocking  effect  to very low  levels.  Hence,  in  the  modified
MPEG-like   system,  instead  of  the  original  two  dimensional
Discrete  Cosine Transform,  a  Lapped  Orthogonal Transformation
is  used  as  the  basic  spatial correlation reduction operation
and  also  customized  quantization and variable  length codeword
tables  are provided  to  ensure  efficiency. The modified coding
algorithm  is  implemented  in  software. Simulations are made to
compare  its  performance  to  the original  MPEG-1 algorithm. As
performance criteria, PSNR versus compression ratio (equivalently
bit-rate) plots and also subjective ratings of visual quality are
used.
Paper

VCII.1
PARTITION PREDICTION FOR SEGMENTATION-BASED CODING TECHNIQUES

Ferran Marques, Bernat Llorens and Antoni Gasull

Universitat Politecnica de Catalunya
Campus Nord - Modul D5
C/ Gran Capita, 08034 Barcelona, Spain
E-mail: ferran@gps.tsc.upc.es

This paper presents a general partition prediction scheme. It consists of four
steps: region parametrization, region prediction, region ordering and partition
creation. The evolution of each region is separated into two types: regular
motion and shape deformation. Fourier Descriptors are used to parametrized both
types of evolution and they are separately predicted in the Fourier domain.
The predicted partition is built from the ordered combination of the predicted
regions, using morphological tools. This technique is applied in the framework
of segmentation-based video coding techniques for coding sequences of complete
partitions as well as sequences of binary images (shape information in Video
Object Planes -VOP-).
Paper

VCII.2
TITLE:
BIORTHOGONAL B-SPLINE FILTER BANKS FOR LOW BIT RATE VIDEO CODING

AUTHORS:
Sergio M. M. de Faria
Mohammed Ghanbari

AFFILIATION:
Dep. of ESE, University of Essex
Wivenhoe Park -- Colchester CO4 3SQ -- England
Tel: +44 1206 872448; fax: +44 1206 872900
e-mail: defasa@essex.ac.uk, ghan@essex.ac.uk

ABSTRACT:
In this paper we investigate the performance of B-Spline filter banks for
low bit rate image coding. The influence of certain characteristics of
the analysis and synthesis of  FIR filters are studied. These include
the B-Spline polynomial order, the effects of coefficient truncation, coding
quantisation and the distortion introduced by the filters themselves. Due to
the high concentration of energy in the low frequency band, these biorthogonal
filter banks have better capabilities to reconstruct signals from the lower
frequency band than their counterparts. As a result a very low bit
rate video codec can be designed by coarse quantisation of the higher bands.
Paper

VCII.3

SCALABLE VIDEO CODING AT VERY LOW BIT RATES EMPLOYING RESOLUTION PYRAMIDS

Klaus Illgner and Frank Mueller

Institut fr Elektrische Nachrichtentechnik
RWTH Aachen, 52056 Aachen, Germany
Tel: +49-241-80-7681; Fax: +49-241-8888-196
{illgner,mueller}@ient.rwth-aachen.de

In this paper an approach for scalable video coding is described,
based on the hybrid coding scheme. The scalability is achieved by
decomposing	the frames to be coded into a resolution pyramid. Motion
estimation and compensation is performed at each level. The focus of
the paper is to design motion estimation and compensation such, that
the resulting pyramid of vector fields as well as the pyramid of
prediction errors can be coded in an efficient fashion.
Paper

VCII.4
ADAPTIVE SUBBAND VQ FOR VERY LOW BIT RATE VIDEO CODING

Stathis P. Voukelatos and John J. Soraghan

Signal Processing Division, Dept. of Electronic and Electrical Eng.,
University of Strathclyde, Glasgow G1 1XW, Scotland, U.K.,
E-Mail: stathis@spd.eee.strath.ac.uk

ABSTRACT
A novel adaptive VQ based subband coding scheme for very low bit rate
coding of video sequences is presented. Overlapped block motion
estimation/compensation is employed to exploit interframe redundancy.
A 2D wavelet transform (WT) is applied to the resulting displaced frame
difference (DFD) signal. The WT coefficients are encoded using an adaptive
vector quantization scheme in combination with a dynamic bit allocation
strategy based on marginal analysis. Simulation results on videophone-type
test sequences are given to evaluate the performance of the codec at
very low bit rates. A comparative performance with the H.261 and H.263
video coding standards is also shown.
Paper

VCII.5
VECTOR REPRESENTATION OF CHROMINANCE
FOR VERY LOW BIT RATE CODING OF VIDEO

Maciej Bartkowiak (1)
Marek Domanski (1)
Peter Gerken (2)

(1) Politechnika Poznanska
Instytut Elektroniki i Telekomunikacji
ul. Piotrowo 3a
60-965 Poznan, Poland
E-mail: mbartkow@et.put.poznan.pl domanski@et.put.poznan.pl
(2) Institut fuer Theroretishe Nachrichtentechnik
und Informationsverarbeitung, Universitaet Hannover
Appelstrasse 9A
30167 Hannover, Germany
E-mail: gerken@tnt.uni-hannover.de

A chrominance vector quantization  technique  is  proposed  as  a
preprocessing step prior to any kind (e.g. DCT-based or OBASC) of
video coding. The operation converts the stream of  two-component
vectors into a scalar stream of chrominance labels. Therefore the
coder  processes  two  channels  only:  one  luminance  and   one
chrominance.  After  decoding  the  two  chrominance channels are
reconstructed from the stream of labels of  chrominance  codebook
entries. Experimental results with still images show recognizable
improvement of the subjective quality by a  constant  compression
ratio.
Paper

VCII.6
A LOW BIT RATE HIERARCHICAL VIDEO CODEC

Kui Zhang, Miroslaw Bober and Josef Kittler

Department of Electronic and Electrical Engineering,
University of Surrey, Guildford GU2 5XH, United Kingdom
e-mail:K.Zhang@surrey.ac.uk, M.Bober@surrey.ac.uk, J.Kittler@surrey.ac.uk

The performance of a very low bit rate video codec largely depends on
the efficient use of motion compensated prediction technique and on a
good coding control strategy. In our previous approach, we proposed a
multiple layer  video codec using affine motion compensation. In this
paper, we further extend our affine compensated multi-layer codec by
incorporating a new block level and designing a coding control strategy.
A measure of coherent motion is used in the decision process which
makes the codec perform efficiently at very low bit rate and for small size
image sequences (QCIF and sub-QCIF format). The experimental results
conduced on 15 MPEG test sequences in QCIF format show improvement in
PSNR of 0.2 dB and reduction in bit rate of 0.9 kbits/second.
Paper

VCII.7
3-D SUBBAND CODING OF VIDEO
USING RECURSIVE FILTER BANKS

Marek Domanski and Roger Swierczynski
Politechnika Poznanska, Instytut Elektroniki i Telekomunikacji, ul. Piotrowo
3a, 60-965 Pozna, Poland
Phone: +48 61 782 762,  Fax: +48 61 782 572,  E-mail: domanski@et.put.poznan.pl
, roger@et.put.poznan.pl

Abstract
A video coding technique based on a three-dimensional subband analysis
with recursive spatial filter banks is proposed. Moreover a simple technique
to compress digital data in the subbands is described. In order to avoid
annoying artifacts at edges and thin lines the filter banks are switched
adaptively. Flat areas are processed with recursive filters exhibiting
long impulse responses and good selectivity, while object edges and other
detailed regions are processed with recursive filters with highly attenuated
impulse responses and poorer selectivity. For very simple encoding scheme
good visual quality has been obtained for real test video sequences in
the CIF format encode at the bitrates about 150 kbps. Obviously further
bit rate reduction could be obtained using a more sophisticated coder.
The very important advantage of the technique proposed is its simplicity.
Paper

VCII.8

MOVING PICTURE FRACTAL CODING USING A MIXED APPROACH IFS AND MOTION

Institut EURECOM
Multimedia Communications dept.,
2229, route des Cretes, B.P. 193, 06904 Sophia Antipolis Cedex.
Tel: +33 93 00 26 41; Fax: + 33 93 00 26 27
e-mail. dugelay@eurecom.fr
url. http://www.eurecom.fr/~image

This paper deals with a possible extension of the
fractal compression algorithm defined for still
image to moving picture. The addressed approach
is a mixed approach based on a combinaison
between inter-frame coding using block-matching,
and an intra-frame coding using IFS.
Paper

VCII.9
A NOVEL METHOD IN REDUCING THE COMPLEXITY OF
FRACTAL ENCODING
L.K. Ma, O.C. Au*, and M.L. Liou**
Department of Electrical and Electronic Engineering
The Hong Kong University of Science and Technology
Clear Water Bay, Kowloon, Hong Kong.
Tel: +852 2358-7053*, +852 2358-7055**
Email: eeau@ee.ust.hk*, eeliou@ee.ust.hk**

ABSTRACT

Fractal coding is a promising technique for image compression.
However, one of the challenges for cost effective implementation is
to reduce the huge computational complexity of the encoder. In this
paper, we propose a novel algorithm to address this issue. Firstly,
we replace mean square error with mean absolute error as
distortion measure to reduce multiplication. Secondly, we use
statistical normalisation to eliminate the need to compute the
scaling factor and offset during the search. Thirdly, we change the
domain block search to range block search to reduce memory
requirement. Simulation results suggest that our algorithm can
reduce computation by three order of magnitude for a QCIF
Paper

VCII.10
AUTOMATIC FRAME FITTING FOR SEMANTIC-BASED
MOVING IMAGE CODING USING A FACIAL CODE-BOOK

Paul M. Antoszczyszyn, John M. Hannah and Peter M. Grant

Department of Electrical Engineering, The University of Edinburgh
Edinburgh, EH9 3JL, UK
Tel: +44 131 6505655; fax: +44 131 650 6554
e-mail: plma@ee.ed.ac.uk

An entirely new method of automatic wire-frame fitting for
semantic-based moving image coding is proposed. The algorithm
utilises a code-book of facial images. All elements of the
facial data-base are pre-processed and manually fitted with the
wire frame model. Both pre-processing and manual fitting are a part
of the facial images data-base preparation. As such, they are
not a part of on-line processing of an unknown image.
Only the pre-processed images (monochrome bitmaps) are used in
automatic frame fitting. This allows a reduced space requirement
for storage of the reference data-base.
Paper

PL.2
MIXED ANALOG-DIGITAL MULTIRATE SIGNAL PROCESSING

Sanjit K Mitra
Department of Electrical and Computer Engineering
University of California
Santa Barbara, CA 93106, U.S.A.

Jose E. Franca
Department of Electrical and Computer Engineering
Instituto Superior Tecnico
Av. Rovisco Pais, 1, 1096 Lisboa Codex, Portugal

ABSTRACT

To achieve higher levels of integration there has been a
growing interest in recent years in designing systems
containing both analog and digital functions on a single
integrated circuit. In most cases, these are inherently
multirate systems because of the different sampling rates
employed at various stages of the system. This paper
reviews some recent developments in this area of integrated
multirate analog-digital systems, with a special emphasis on
their applications to communication systems.
Paper

PL.3
EXPECTATION-BASED MULTI-FOCAL VISION FOR VEHICLE GUIDANCE

Ernst D. Dickmanns

Universitaet der Bundeswehr Muenchen
D-85577 Neubiberg, Germany
Tel: +89 6004 2077/3583; Fax: +89 6004 2082;
e-mail: Ernst.Dickmanns@unibw-muenchen.de

ABSTRACT
Based on experience with several generations of vision
systems for road vehicle guidance a new complex vehicle
eye and corresponding control schemes for viewing
direction control and feature extraction are proposed
allowing new levels of performance with state of the art
general purpose processors. Modeling along the time axis
is the key to an efficient use of the degrees of freedom
Paper

PL.4
OF ARTIFICIAL LEARNING

A. R. Figueiras-Vidal, A. Arts-Rodrguez, J. Cid-Sueiro(*),M. Martnez-Ramn
Ph: +34 1 336 72 26; Fax: +34 1 336 73 50; E-Mail: anibal@gtts.ssr.upm.es

ABSTRACT

Since many signal processing problems can be posed as sample-based decision
and estimation  tasks, we discuss how studies from other fields such as
neural networks might suggest improved architectures (models) and algorithms
for these types of problems. We then concentrate on PAM equalization,
showing that a reordering of the equivalent classification problem generates
a 'staircase' which, while retaining the simplicity of the classical equalizer,
allows modifications to made in the outputs and in the training objectives
which provide advantages even in the least complex cases.We go on to demonstrate
that these advantages increase when one considers, for example,  nonlinear
channels with memory.

From our simulations we draw conclusions  and suggest futher related research.
We also present two new avenues of inquiry, offering significant practical
advantages, which are motivated by the discussions.
Paper

SS.1.1
A MRF BASED APPROACH TO COLOR IMAGE RESTORATION

C.S. Regazzoni, E. Stringa, A.N. Venetsanopoulos*
Department of Biophysical and Electronic Engineering (DIBE), University
of Genoa
Via all'Opera Pia 11A, 16145 Genova, ITALY
Tel: +39 10 3532792; fax: +39 10 3532134
e-mail: carlo@dibe.unige.it
*Department of Electrical and Computer Engineering, University of Toronto

ABSTRACT
In this paper, a Markov Random Field (MRF)-based method is presented.
MRF methods are based on a probabilistic representation of a image processing
problem; the problem is represented as the maximization of a probability
measure computed starting from input data for all possible solutions.
The optimization process is often computationally expensive. The coupled
problem of restoring and extracting edges from an image is here considered.
An extension to the color case of the deterministic mean-field annealing
method presented in [1] is presented. The main advantage of this approach
is its capability of obtaining a sub-optimum solution in a faster way
with respect to optimal stochastic methods (e.g., Simulated Annealing).
Paper

SS.1.2
Resolution Enhancement of Color Video

Brian C. Tom and Aggelos K. Katsaggelos}
Northwestern University
Department of Electrical Engineering and Computer Science
McCormick School of Engineering and Applied Science
Evanston, IL  60208-3118   USA
Tel: (847) 491-7164
Fax: (847) 491-4455
Email: briant@eecs.nwu.edu, aggk@eecs.nwu.edu

In this paper, an approach to improve the spatial resolution of color video
is presented.  Such high resolution images are desired, for example, in
video printing.  Previous work has shown that the most important step in
achieving high quality results is the accuracy of the motion field.  It is well
known that motion estimation is an ill-posed problem.
However, in processing color video, additional information contained in the
color channels may be used to improve the accuracy of the motion field over the
motion field obtained with the use of only one channel.  In turn, this
improvement in the motion field will be shown through several experimental
results to significantly improve the estimation of a high resolution image
sequence from a corresponding observed low resolution sequence.
Paper

SS.1.3
Noise modeling for smoothing the colour histogram

L.Shafarenko, M.Petrou and J.Kittler
Dept. of Electronic and Electrical Engineering,
University of Surrey,
Guildford GU2 5XH, United Kingdom.
e-mail: l.shafarenko,m.petrou @ee.surrey.ac.uk

In this paper we present a segmentation algorithm for colour images
that uses the watershed algorithm to segment either the 2D or the 3D
colour histogram of an image. For compliance with the way humans
perceive colour, this segmentation has to take place in a perceptually
uniform colour space like the   space. To avoid oversegmentation,
the watershed algorithm has to be applied to a smoothed out
histogram. The noise, however, is inhomogeneous in the   space and
we present here the noise analysis for this space based on assumptions
that are experimentally justified.
Paper

SS.1.4
ADAPTIVE MULTICHANNEL L FILTERS BASED ON REDUCED ORDERING

N. Nikolaidis    I. Pitas

Dept. of Electrical and Computer Engineering, University of Thessaloniki,
Thessaloniki, GREECE, nikolaid@zeus.csd.auth.gr
Dept. of Informatics, University of Thessaloniki,
Thessaloniki, GREECE,  pitas@zeus.csd.auth.gr

Multichannel L filters that are based on the reduced ordering principle
have been proposed lately as an effective nonlinear filtering structure
for multivariate data.  The evaluation of the optimal  coefficients for
these filters requires a priori  information on  the signal statistics
which might not be always available. To overcome this,  we propose
adaptive multichannel L filters that are based on the LMS algorithm.
Convergence issues for the new adaptive filter structures are  studied.
Experiments involving color images prove the superior performance of the
proposed filters in noise removal.
Paper

SS.1.5
NEAREST NEIGHBOUR MULTICHANNEL FILTERS FOR IMAGE PROCESSING

K.N. Plataniotis, D. Androutsos, A.N. Venetsanopoulos}
Department of Electrical and Computer Engineering
University of Toronto
http://www.comm.toronto.edu/~dsp/dsp.html
e-mail: kostas@dsp.toronto.edu

This paper addresses the problem of noise attenuation for multichannel
data. Two multichannel filters which utilize adaptively determined data
dependent coefficients are introduced. The special case of colour image
processing is studied as an important example of multichannel signal processing.
Simulation results indicate that the new filters are computationally attractive
and have excellent performance.
Paper

SS.1.6
COLOR IMAGE FILTERING USING GENERALIZED COST FUNCTIONS

D. Sindoukas, S. Fotopoulos, G. Economou

University of Patras, Physics Department, Electronics Laboratory, GR-26110
Patras,GREECE.
Tel.: +30 61 997465, FAX: +30 61 997456
email: spiros@physics.upatras.gr

ABSTRACT
The concept of cost function (CF) in the context of image filtering is put under investigation
in this work. Optimal behaviour of the resulting filters in respect with noise attenuation and
edge preservation is sought through the minimization of these functions. This behaviour can
be controlled by proper adjustment of certain parameters in some cases. Function
combinations are also considered. Finally, the proposed schemes are tested on real images and
objective as well as subjective results are reported.
Paper

SS.1.7
MORPHOLOGICAL LIKE OPERATORS FOR COLOR IMAGES

Constantin Vertan, Viorel Popescu, Vasile Buzuloiu

Department of Applied Electronics, Bucuresti "Politehnica" University

fax: + 40 1 312. 31. 93

email: vertan@alpha.imag.pub.ro, vpopescu@edil.edil.pub.ro, buzuloiu@alpha.imag.pub.ro

Primarily based on Serra's framework, mathematical morphology has become
one of the most used nonlinear processing and analysis techniques. Later
work extended the initially set operators to functions, in a general algebraic
definition for multidimensional scalar signals. The case of vector valued
images (or signals) is not included in this theory. The extension of
mathematical morphology to color images is equivalent to the definition
of an ordering relation in a vector space. In this paper we will investigate
several ordering relations in the color space, each of them yielding to
the definition of morphological operations. The performance of the filtering
based on these operations is evaluated in terms of Normalized Mean Square
Error (NMSE), Mean Chromaticity Error (MCRE), space topology preservation
and visual subjective perception of image quality.
Paper

SS.1.8
IMAGE SEGMENTATION BY AREA DECOMPOSITION OF HSV COMPONENTS

Stephen J. Impey and J. Andrew Bangham

School of Information Systems, University of East Anglia, Norwich NR4 7TJ, UK

Email: sji@sys.uea.ac.uk

Coloured images may be simplified with an area based sieve whilst preserving
edges and, usually, colour up to the edges using either the hue, saturation
and value (HSV) or red, blue, green (RGB) components. Furthermore, an image
may be segmented by area. Applying the sieve to HSV components from a colour
image appears to significantly improve the chances of finding objects in a
scene, particularly when the objects have different colours. An example of
finding cars in a car park scene is presented.
Paper

SS.1.9

CLASSIFICATION OF MULTISPECTRAL REMOTE-SENSING IMAGES BY NEURAL
NETWORKS

F. Roli(1), S.B. Serpico(2), L. Bruzzone(2), and G. Vernazza(1)

(1) Dept. of Electrical and Electronic Eng., University of Cagliari
Piazza dArmi, I-09123, Cagliari, Italy
tel: +39 70 6755897; fax: +39 70 6755900
e-mail: vernazza@elettro1.unica.it
(2) Dept. of Biophysical and Electronic Eng., University of Genoa
Via All Opera Pia, 11A, 16145, Genova, Italy
tel: +39 10 3532752; fax: +39 10 3532134
e-mail: vulcano@dibe.unige.it

ABSTRACT

This paper addresses the classification of multispectral remote-sensing
images by the neural-network approach. In particular, an experimental comparison
on the performances provided by different neural models for classifying
multisensor remote-sensing data is reported. Four neural classifiers are
considered in the comparison: the Multilayer Perceptron, Probabilistic Neural
Networks, Radial Basis Function networks and a kind of Structured Neural Networks.
Paper

SS.1.10
NEURAL PROCESSING OF MULTISPECTRAL
AND MULTITEMPORAL AVHRR DATA

Vito Cappellini(*),  Marco Benvenuti (), Carlo Di Chiara (), Stefano
Fini ()

(*) University of Florence, Department of Electronic Engineering
Via di S. Marta, 3 - 50139 Florence - Italy

() Fondazione per la Meteorologia Applicata
Via Caproni, 8 - 50145 Florence - Italy

() Centro di Studi per l'Informatica applicata in Agricoltura
Via Caproni, 8 - 50145 Florence - Italy

ABSTRACT

In the last years a large amount of multisensor data has been generated
in consequence of  the development of remote sensing techniques for the
analysis of the Earth's surface.
The study of the evolution of the vegetation status is particularly useful
in planning agro-ecological operations and in the estimation of the vegetation
development.
In this paper, vegetation index data (NDVI) collected by the AVHRR sensor
on the NOAA satellite are processed. These multitemporal data belong to
a historical archive composed of ten years of ten-day images of the whole
African continent. This archive has been implemented in the framework
of a co-operation between NASA-GSF and the FAO Remote Sensing Centre (ARTEMIS
project). The archive starts from August 1981 to June 1991 and is composed
of 356 georeferenced images having a spatial resolution of  7.6 km x 7.6
km.
This set of NDVI data collected over a so long period of time is extremely
useful when the annual and seasonal variations of the reflectance of the
Earth surface have to be investigated. In this work a new approach to
NDVI data processing is presented: it is composed of both statistical
analysis techniques and neural algorithms. The large number of images
in the archive makes extremely difficult to analyse the whole data set
and this is particularly true when personal computer are used for processing.

The method can be summarized in two fundamental steps: i) reduction of
the number of images to be processed controlling the loss of information
by means of statistical techniques; ii) the use of a neural network for
clustering the scene in order to put in evidence areas showing similar
vegetation index.variability.
In the first processing step, the Principal Component transformation is
applied to images of each year thus eliminating redundant information.
In this way the number of images to be processed by the unsupervised classifier
is dramatically reduced. The optimal number of classes is chosen by the
chi-squared statistical test, suitably modified and applied to different
classifications with variable number of clusters. A three-layered neural
network is used for clustering. This newtork is obtained with the combination
of two well known architectures: the first one is unsupervised (Kohonen
map) whilst the second one is supervised (Grossberg layer). At the end,
means and standard deviations of the vegetation index for each cluster
as well as for each decade are computed.
Paper

SS.2.2
VIDEO CODING USING ADAPTIVE GLOBAL MC AND LOCAL AFFINE MC

Hirohisa Jozawa, Kazuto Kamikura, Kazuhisa Yanaka, and Hiroshi Watanabe

NTT Human Interface Laboratories (jozawa@nttvdt.hil.ntt.jp)

This paper describes an efficient video coding method using two-stage
motion compensation (MC). The proposed MC method employs global MC (GMC)
and overlapped block affine MC. GMC is adaptively turned on/off for each
macroblock since GMC cannot predict all regions in an image. Simulation
results show that the proposed coding method using two-stage MC significantly
outperforms H. 263 for sequences with fast motion. Performance improvements
in PSNR are about 3-4 dB over H. 263.
Paper

SS.2.3
STANDARDS BASED VIDEO COMMUNICATIONS AT VERY LOW BIT-RATES

Bernd Girod, Niko Faerber, and Eckehard Steinbach

Lehrstuhl fuer Nachrichtentechnik
University of Erlangen-Nuremberg
Cauerstrasse 7, D-91058 Erlangen, Germany
Tel: +49 9131 857100; fax: +49 9131 303840
E-mail: girod@nt.e-technik.uni-erlangen.de

Video communication at very low bit-rates has
made significant progress recently through the new ITU-T
standard H.263. In this paper, we are reviewing the performance
advances over the 1990 ITU-T standard H.261, and present
a novel extension that allows robust transmission of moving
video over highly unreliable channels, such as the mobile channel.
Paper

SS.2.4
SELECTIVE CODING BY FOCUS OF ATTENTION: A NEW TOOL TO ACHIEVE VLBR VIDEO CODING

Eric Nguyen, Claude Labit

IRISA, Campus Universitaire de Beaulieu
35042 Rennes Cedex, France
Tel: +33 99 84 72 60; fax: +33 99 84 71 71
{nguyen,labit}@irisa.fr

Selective source coding is an essential part of very low bit rate
(VLBR) image/video compression where a significant irrelevancy
reduction has to be performed.  In this paper, this reduction is
described in the context of visual attention: the selection of
relevant spatial information at the expense of other (non-relevant)
information in order to maximize the efficiency of a particular visual
communication task. We first give a general framework of selective
coding. We then illustrate it with some examples of implementation
using the generic wavelet representation as a stand-alone technique or
for spatial encoding of the MC residuals in a MC-DPCM hybrid video
coding scheme.
Paper

SS.2.5
LOW BIT RATE VIDEO CODING FOR MOBILE MULTIMEDIA COMMUNICATIONS

Reginald L. Lagendijk, Jan Biemond and Cor P. Quist

Delft University of Technology, Department of Electrical
Engineering, Information Theory Group
P.O. Box 5031, 2600 GA Delft, The Netherlands
Tel: +31 15 278 3731; Fax: +31 15 278 1843
e-mail: {lagendijk,biemond}@et.tudelft.nl; WWW: http://www-
it.et.tudelft.nl

In this paper we first describe the objectives of the Delft
Mobile Multimedia Communications project. Next, the subject of
lossy contour compression is considered in more detail as it is
an essential component of most object or region-based
compression techniques for low bit rate video coding. We
propose an optimized B-splines approximation approach, which
results in a 40 percent higher compression than the lossless
conditional chain code method. Achieved rates are, depending on
the tolerable deviation between original and coded contour, in
the order of 0.70 to 0.90 bit per contour pixel.
Paper

SS.2.6
A VERY LOW BIT-RATE VIDEO CODEC WITH OPTIMAL
TRADE-OFF AMONG DVF, DFD AND SEGMENTATION

Guido M. Schuster and Aggelos K. Katsaggelos

Northwestern University
Department of Electrical and Computer Engineering
2145 Sheridan Road, Evanston, Illinois  60208-3118, USA
E-mail: gschuster@nwu.edu, aggk@eecs.nwu.edu

In this paper we present a theory for the optimal bit allocation
among quad-tree (QT) segmentation, displacement vector field (DVF)
and displaced frame difference (DFD). The theory is applicable to
variable block size motion compensated video coders (VBSMCVC),
where the variable block sizes are encoded using the QT structure,
the DVF is encoded by first order differential pulse code modulation
(DPCM), the DFD is encoded by a block based scheme and an additive
distortion measure is employed. We consider the case of a lossless
VBSMCVC first, for which we develop the optimal bit allocation
algorithm using Dynamic Programming (DP). We then consider a lossy
VBSMCVC, for which we use Lagrangian relaxation and show how an
iterative scheme, which employees the DP-based solution, can be used
to find the optimal solution. We finally present a VBSMCVC, which is
based on the proposed theory,  which employees a DCT-based DFD
encoding scheme. We compare the proposed coder with H.263.
The results show that it outperforms H.263 by about 25%
in terms of bit rate for the same quality reconstructed image.
Paper

SS.2.7
SELECTIVE USE OF MODEL-BASED CODING FOR LARGE
MOVING OBJECTS

Don Pearson
Departement of Electronic Systems Engineering
University of Essex, Colchester CO4 3SQ, UK
Tel: +44 1206 872865; Fax: +44 1206 872900
Email: dep@essex.ac.uk

Measurements using a continuous quality recording method have
revealed the extent of quality variations that occur in MPEG2
pictures at low bit rates. large moving objects in particular
can give rise to particularly  severe troughs in quality. The
complementary characteristics of model-based coding are examined
with a view to a synthesis of the two methods in a switched
coder, with possible increased overall coding efficiency.
Paper

SS.2.8
VERY LOW BITRATE VIDEO CODING AND MPEG-4:
STILL A GOOD RELATION

Fernando Pereira

Instituto Superior Tcnico - Instituto de Telecomunicaes
Av. Rovisco Pais, 1096 Lisboa Codex, PORTUGAL
Telephone: + 351 1 8418460; Fax: + 351 1 8418472
E-mail: eferbper@beta.ist.utl.pt

ABSTRACT
MPEG-4 emerged recently as an important development in the field of audio-visual coding aiming at
establishing the first content-based audio-visual coding standard. This paper intends to analyse the current
relation between MPEG-4 and very low bitrate video coding and corresponding applications, notably by
considering the MPEG-4 objectives, functionalities and recent technical developments related to video
coding.
Paper

SS.2.9
DYNAMIC CODING FOR VISUAL COMMUNICATIONS

Emmanuel REUSENS, Touradj EBRAHIMI, Roberto CASTAGNO, Corinne LE BUHAN and Murat KUNT

Signal Processing Laboratory
Swiss Federal Institute of Technology
CH-1015 Lausanne,
SWITZERLAND
E-mail: reusens@lts.de.epfl.ch

In this paper,  a new approach to the problem of visual data representation in
the framework of multimedia is introduced. The approach, named 'dynamic coding', consists
in a dynamic combination of multiple representation models and segmentation strategies.
Given an application, these two degrees of freedom are assembled so as to yield a
specific profile which meets the specifications dictated by the application. The data is
represented as the union of data segments,
each described with a locally appropriate representation model. In order to illustrate
this approach, a video compression system, based on the principles of dynamic coding,
is proposed in the context of video-telephone/conference applications.
Paper

SS.2.10
SEGMENTATION-BASED VIDEO CODING: TEMPORAL LINKING AND RATE CONTROL

Philippe Salembier, Ferran Marques and Montse Pardas

Universitat Politecnica de Catalunya
Campus Nord - Modul D5
C/ Gran Capita, 08034 Barcelona, Spain
E-mail: {philippe,ferran,montse}@gps.tsc.upc.es}

This paper analyzes the main elements that a segmentation-based video
coding approach should be based on so that it can address coding
efficiency and content-based functionalities. Such elements can be
defined as temporal linking and rate control. The basic features of
such elements are discussed and, in both cases, a specific
implementation is proposed.
Paper

SS.3.1
Chinese Remainder Theorem: Recent Trends and New Results in Filter Banks Design

C.W.Kok and T.Q.Nguyen
ECE Dept., University of Wisconsin Madison,
1415 Engineering Drive, Madison, Wl 53706
Tel: (608)-265-4885	Fax: (608)-262-4623
email:  ckok@cae.wisc.edu and nguyen@ece.wisc.edu

Recent advances in the time domain methods have led to many new
approaches in filter bank designs. The objective of this paper is to
derive a unified theory for these time domain methods, based on the
Chinese Remainder Theorem. Topics discussed in this paper include
two-channel filter banks, M-channel filter banks and 2-D filter
banks. Design examples are presented to demonstrate the theory.
Paper

SS.3.2
ON PERFECT-RECONSTRUCTION FIR FILTER BANKS

Eleftherios Kofidis{1} S. Theodoridis{2} N. Kalouptsidis{2}

1: Department of Computer Engineering and Informatics, University of Patras,
Patras 265 00, Greece. E-mail: kofidis@cti.gr
2: Department of Informatics, Division of Communications and Signal Processing,
University of Athens, Athens 157 71, Greece.
E-mail: {stheodor,kalou}@di.uoa.gr

This paper deals with the problem of designing an N-band maximally-decimated
analysis filter bank given K of its filters, so that perfect reconstruction
with FIR synthesis filters is possible. An algorithm for computing the
N-K unknown analysis filters and the synthesis filters is given and the
solution set is completely parametrized. The parametrization is exploited
in optimizing the frequency responses of the resulting filters and to derive
also a simple parametrization for the paraunitary case. The linear-phase case
is also discussed with emphasis on the 2-band filter banks.
An example is provided to illustrate the theory.
Paper

SS.3.3
LATTICE STRUCTURE FOR TWO-CHANNEL FILTER BANKS WITH COMPLEX
COEFFICIENTS, WHICH YIELD SYMMETRIC WAVELET BASES

Todor Cooklev* , Akinori Nishihara^ , and Masaki Kato^

* Dept. Electr.   Comp. Eng.      ^Dept. Physical Electronics
University of Toronto 	          Tokyo Inst. Technology
10 King's College Rd. 	          2-12-1 Ookayama, Meguro-ku
Toronto, ON M5S 1A4, Canada       Tokyo, 152 Japan  minipage
cooklev@dsp.toronto.edu           aki@ss.titech.ac.jp

ABSTRACT

A new lattice structure is described. It is capable of implementing all
paraunitary two-channel filter banks where the filters have complex
coefficients and yield symmetric wavelet bases.
This lattice structure, while being a general design method, can also be
used to actually design the filter bank.
These filter banks are, in fact, a special case of multifilter banks and can
also be related to Golay-Rudin-Shapiro complementary polynomial pairs.
The applications of such filter banks are to be found in subband coding and
communications systems.
Paper

SS.3.4
FIR OVERSAMPLED  FILTER  BANKS AND FRAMES IN l2(Z)

Zoran Cvetkovic and Martin Vetterli

Department of Electrical Engineering and Computer Sciences
University of California, Berkeley, CA 94720, USA

zoran@eecs.berkeley.edu, martin@eecs.berkeley.edu

Perfect reconstruction FIR filter banks implement a particular
class of signal expansions in l2(Z). These expansions
are studied in this paper. Necessary and sufficient conditions on an FIR
filter bank to implement a frame or a tight frame decomposition are given, as
well as the necessary and sufficient condition for a feasibility of perfect
reconstruction using FIR filters. Complete parameterizations of FIR filter
banks satisfying these conditions are given. Further, we study the condition
under which the minimal dual frame to the frame associated to an FIR filter
bank is also FIR, and give a parameterization of a class of filter banks
having this property. We then concentrate on the least constrained class,
namely nonsubsampled filter banks, for which these frame conditions have
particular forms.
Paper

SS.3.5
AN ADAPTIVE PROJECTION ALGORITHM FOR MULTIRATE FILTER BANK OPTIMIZATION

Dong-Yan Huang and Phillip A. Regalia

Departement Signal & Image
Institut National des Telecommunications
9, rue Charles Fourier
F-91011 Evry cedex France
huang@int-evry.fr, regalia@galaxie.int-evry.fr

Abstract:
We develop a new algorithm for multirate
filter bank optimization, which finds
application in subband coding or wavelet signal analysis.
Although some impressive off-line algorithms have recently been
developed for this purpose, the computation demand of such
algorithms often renders them prohibitive for real-time applications.
In this vein, adaptive filtering solutions remain of interest.
A simple gradient descent algorithm may be ill suited
due to the nonquadratic nature of the cost function to be minimized,
and accordingly non gradient algorithms may offer
some attractive alternatives.
The present paper describes a projection type algorithm,
which aims to construct a lossless filter bank such that
one of its impulse responses lies close to
an extremal eigenvector of the input signal autocorrelation
matrix.
Though a formal convergence proof of the algorithm is not offered,
simulations show that the algorithm converges to an acceptable
vicinity of the global minimum point of the cost function.
Paper

SS.3.6
CONSIDERATIONS IN THe DESIGN OF OPTIMUM
COMPACTION FILTERS  FOR SUBBAND CODERS

Yuan-Pei Lin and P. P. Vaidyanathan

yplin@systems.caltech.edu ppvnath@systems.caltech.edu
Dept. of Electrical Engineering, 136-93, Caltech, Pasadena, CA
91125, U.S.A.

Abstract

Recently there has been considerable interest in the
design of optimal paraunitary filter banks for a given class of inputs.  In
this paper we address a number of practical considerations associated with
the design and implementation of optimal paraunitary filter banks.
Paper

SS.3.7
ORTHOGONAL TRANSMULTIPLEXER: A MULTIUSER COMMUNICATIONS
PLATFORM FROM FDMA TO CDMA

Ali N. Akansu and Mehmet V. Tazebay

New Jersey Institute of Technology
Department of Electrical and Computer Engineering
Center for Communications and Signal Processing Research
University Heights, Newark, NJ 07102

ABSTRACT

Orthogonal transmultiplexers have been successfully utilised for
multi-user communications. They are of the FDMA type in their
most common version. Mostly, frequency-selective PR-QMFs
were used in transmultiplexers as orthogonal user codes for
CDMA communications reported in the literature. This conflicts
with the fundamentals of CDMA theory. We introduce novel
M-valued spread spectrum PRQMF codes in this paper. It is
shown that the proposed M-valued spread spectrum PR-QMF
codes with minimised auto- and cross-correlation properties
outperform the conventional Gold codes in CDMA
communication scenarios considered in the paper.
Paper

SS.3.8
ON EFFICIENT IMPLEMENTATION OF MULTIDIMENSIONAL MULTIRATE FILTERS
DERIVED FROM ONE-DIMENSIONAL FILTERS

Tsuhan Chen

AT&T Research
Room 4C528, 101 Crawfords Corner Road, Holmdel, NJ 07733, USA
Tel: +1 908 949-2708
Fax: +1 908 957-8388
e-mail: tsuhan@research.att.com

We study the efficient implementation of multidimensional (MD) filters
used in multirate systems.  These filters, typically having
parallelepiped-shaped passband supports, can be derived from
one-dimensional (1D) prototype filters.  The resulting nonseparable MD
filters have separable polyphase components that are combinations of
the polyphase components of the 1D prototypes, so efficient
implementation exists.  We show that, for the two-dimensional case, all
the polyphase components of the 1D prototypes are utilized.  Therefore,
there is no design overhead in this scheme.
Paper

SS.3.10
MEASUREMENT AND SYMBOLIC ANALYSIS OF IMPLEMENTED MULTIRATE SYSTEMS

Hans W. Schuessler and Frank Heinle

Lehrstuhl fuer Nachrichtentechnik,
Universitaet Erlangen-Nuernberg,
Cauerstrasse 7, D-91058 Erlangen, Germany
Phone : +49-9131-857101
Fax   : +49-9131-303840
E-mail: heinle@nt.e-technik.uni-erlangen.de

Multirate systems (MRS) play a major role in modern telecommunication.
Important examples are filter banks for image or speech coding,
transmultiplexers, and sampling rate converters. In general, these
systems are designed without consideration of implementation aspects
such as wordlength limitations. The performance of realized systems
will therefore differ from the desired one depending on the system
structure. Not all deviations can be calculated in closed form and even
practicable calculations are often extensive and error-prone. Therefore,
we present a method for measuring quantization effects in realized MRS.
Furthermore, we introduce a new program for the symbolic analysis of MRS
using the computer algebra program MAPLE.
Paper

SS.3.11
EFFICIENT IIR SWITCHED-CAPACITOR DECIMATORS AND INTERPOLATORS

F. A. P. Baruqui (1), A. Petraglia (2), S. K. Mitra (3) and J. E. Franca (4)

(1) Programa de Engenharia Eletrica COPPE, EE/UFRJ - 21945-970
Rio de Janeiro, RJ, Brasil. baruqui@coe.ufrj.br
(2) Programa de Engenharia Eletrica COPPE, EE/UFRJ - 21945-970
Rio de Janeiro, RJ, Brasil. antonio@coe.ufrj.br
(3) Depto. of Elec. & Comp. Engineering - Univ. of California, Santa Barbara,
CA 93106-9560. mitra@ece.ucsb.edu
(4) Grupo de Circ. e Sist. Integrados, Inst. Superior Tecnico -
Av. Rovisco Pais 1, 1096 Lisboa Codex Portugal. franca@ecsm4.ist.utl.pt

ABSTRACT
The IIR switched-capacitor decimators and interpolators proposed in this paper
are based on the polyphase decomposition of an M-th band IIR lowpass filter,
and uses first- and second-order allpass switched-capacitor filters as basic
building blocks, which operate at the lower sampling rate, reducing power
consumption, capacitance spread and total capacitance area. The resulting
switched-capacitor network has low sensitivity with respect to capacitance
ratio errors, specially in the passband, where very low sensitivity is
guaranteed by using structurally allpass filters. These properties have been
verified by computer based sensitivity analysis, and an illustrative design
example, considering realistic specification for video communication
applications, included in the paper, along with comparisons with other
approaches reported in the literature. Laboratory results obtained with a
prototype filter are shown as well.
Paper

SS.4.1
COMBINED ACOUSTIC ECHO CONTROL AND NOISE REDUCTION FOR HANDS-FREE TELEPHONY -
STATE OF THE ART AND PERSPECTIVES

Rainer Martin and Peter Vary

IND, Aachen University of Technology
52056 Aachen, Germany
Tel: +49 241 806984; fax: +49 241 8888186
e-mail: martin@@ind.rwth-aachen.de

In this paper we summarize and discuss recent results in acoustic
echo cancellation and noise reduction with emphasis on methods which
combine both aspects. It is shown that echo control and noise reduction
can support each other in a true synergy. The paper discusses
fundamental issues of algorithm design and suggests that a frequency
domain multi-microphone solution might be best suited to achieve the
desired performance.
Paper

SS.4.2
BINAURAL ANALYSIS METHODS AND THEIR RELATIONSHIP TO QUALITY EVALUATION
OF HANDS-FREE TELECOMMUNICATION EQUIPMENT

H. W. Gierlich

HEAD acoustics GmbH, Ebertstr. 30a 52134 Herzogenrath, Germany,Tel.: +49
2407 57722; Fax: +49 2407 57799, e-mail:head-gr@infoac.rmi.de

Since modern telecommunication equipment, especially hands-free telephones,
incorporates sophisticated signal processing, the analysis methods must
take into account the properties of the human hearing. The basis for the
correct aquisition of test data -used for auditory but instrumental measurements
as well- is the binaural rcording and binaural analysis of the test stimuli.
The paper gives an overview, in what ways binaural methods can be applied
for Quality evaluation. The paper focusses on methods for aquiring test
data in the listening situation, in the converational situation and for
instrumental measurements using defined, artificial test stimuli. Various
methods for playback of binaurally recorded sounds in different situations
are shown.
Paper

SS.4.3

IMPLEMENTATION AND EVALUATION OF AN ACOUSTIC ECHO CANCELLER USING DUO-FILTER
CONTROL SYSTEM

Yoichi Haneda, Shoji Makino, Junji Kojima, and Suehiro Shimauchi

NTT Human Interface Laboratories (E-mail: haneda@splab.hil.ntt.jp)

The developed acoustic echo canceller uses an exponentially weighted step-size
projection algorithm and a duo-filter control system to achieve fast convergence
and high speech quality. The duo-filter control system has an adaptive
filter and a fixed filter, and uses variable-loss insertion. Evaluation
of this system with multi-channel A/D and D/A converters showed that (1)
the convergence speed is under 1.5 seconds for speech input when the adaptive
filter length is 125 ms, (2) the residual echo level is nearly as low
as the ambient noise level (average: under -20 dB; maximum: under -35
dB), and (3) near-end speech is sent with no disturbance during double
talk.
Paper

SS.4.4
IDENTIFYING THE TRUE ECHO PATH IMPULSE RESPONSES
IN STEREOPHONIC ACOUSTIC ECHO CANCELLATION

Fabrice Amand*, Andr Gilloire**, Jacob Benesty***

* CEFRIEL, Politecnico di Milano, Via Emanueli, 15, 20126 - Milano, Italy
email: amand@mailer.cefriel.it

** CNET LAA/TSS/CMC Technopole Anticipa, 2 Avenue Pierre Marzin, 22307
Lannion Cedex, France
e-mail: gilloire@lannion.cnet.fr

*** Lucent Technologies, Bell Labs Innovations, 600 Mountain Avenue, Murray
Hill, NJ 07974, USA

ABSTRACT
A fundamental problem in stereophonic acoustic echo cancellation for teleconferencing
is the possibility to identify the true impulse responses of the acoustic
echo paths. This problem arises from the correlation between the two signals
picked up in the remote room. We demonstrate by simple theoretical considerations
and experiments that in real situations, due to the characteristics of
the acoustic environment in the remote room, the identified impulse responses
converge to the true echo path impulse responses.
Paper

SS.4.5
ANALYSIS OF TWO STRUCTURES FOR COMBINED ACOUSTIC ECHO CANCELLATION AND NOISE REDUCTION

Yann Guelou, Abdelkrim Benamar, Pascal Scalart

France Telecom CNET LAA/TSS/CMC
benamar@lannion.cnet.fr, scalart@lannion.cnet.fr

This paper addresses the problem of speech enhancement in the context of GSM hands-free
radiotelephony where the signal to be transmitted is corrupted by background noise
and echo signals. We analyze possible schemes for combined acoustic echo cancellation
(AEC) and noise reduction (NR) devices. Considering two AEC algorithms and one NR
device, we show that the overall performances obtained by these schemes are greatly
dependent on the intrinsic behaviour of the considered AEC algorithms. These results
are confirmed by informal listening tests presented in that contribution.

Paper

SS.4.6
PERFORMANCE OF ADAPTIVE DEREVERBERATION TECHNIQUES USING DIRECTIVITY CONTROLLED
ARRAYS

C. Marro*, Y. Mahieux*, K. U. Simmer**

*FRANCE TELECOM - CNET LAA/TSS/CMC Technopole Anticipa, 2 avenue Pierre
Marzin 22307 Lannion Cedex - FRANCE

**Houpert Digital Audio, Wiener Str 5, D-28359 Bremen, GERMANY

e-mails: marro@lannion.cnet.fr - mahieux@lannion.cnet.fr - u.simmer@proaudio.de

ABSTRACT:
The use of optimal postfiltering has been previously proposed to increase
the performance of microphone arrays. In this paper, an analysis of the
postfilter shows that its behaviour is closely related to the one of the
array. This is illustrated by considering a typical videoconferencing
context. The results we provide demonstrate that the use of a directivity
controlled array is a requirement to ensure a sufficient robustness of
the whole system. It is also shown that the dereverberation performed
by the postfilter is limited and that its main interest lies in a significant
reduction of the acoustic echo even in the double talk case. This attractive
property depends on the whole design of the array including its placement
versus the acoustic echo sources.
Paper

SS.4.7
A HANDS-FREE PHONE SYSTEM BASED ON A PARTITIONED FREQUENCY-DOMAIN ADAPTIVE
ECHO CANCELER

Pius Estermann and August Kaelin

Swiss Federal Institute of Technology Zurich, Switzerland,
esterman@isi.ee.ethz.ch

Providing means for hands-free conversation is of great interest for
industry and is still a current research topic. In this paper, a
partitioned frequency-domain adaptive FIR filter is applied in a
hands-free phone system to provide echo compensation.  It is optimally
designed in such a way that it approaches the tracking behavior of the
Recursive Least-Squares (RLS) algorithm, and it is combined
with a new adaptive step-size control in order to cope with varying
far-end/local speaker situations. Its performance is demonstrated by
means of real speech signals. Assuming a loudspeaker-room-microphone
impulse response duration of 3500 taps, an increase in the critical
gain of 14dB has been obtained (for each phone) by using an adaptive
echo canceler with 1152 taps.
Paper

SS.4.8

ECHOCOMPENSATION AND NOISE SUPPRESSION FOR
SPEECH RECOGNITION APPLICATIONS

Dr. Walter Stammler, Matthias Schulz, Frank Scheppach

Daimler-Benz Aerospace AG, Sensor Systems
Woerthstrasse 85, D-89077 Ulm, Germany
phone: +49 731 3925631, fax: +49 731 3927144
e-mail: scheppf@vs-ulm.dasa.de

ABSTRACT
This contribution deals with the role and the performance
of echocompensation and noise suppression, when used in
combination with speech recognition systems.
For two applications of interest (speech control in car or
via telephone) there are quite significant differences to
classical echocompensation and noise suppression for telephone
conferences. It will be pointed out, how the systems are
structured, what performance can be achieved and how realtime
solutions are looking like.
Paper

SS.4.9
HANDSFREE SPEAKING FOR COMMUNICATION TERMINALS

Hans J. Matt and Michael Walker
ALCATEL TELECOM, Lorenzstr. 10, D-70435 Stuttgart, Germany
Tel: +49-711-869-32246 and -32556; Fax: +39-711-869-32302
e-mail: hmatt@rcs.sel.de  and  mwalker@rcs.sel.de

Abstract
In this paper some considerations for the realisation of
a most natural handsfree speaking are presented. Its essential
features comprise full duplex operation, speech loudness
well adapted to the user's environment, background noise
suppression and cancellation of line echoes. Furthermore
its algorithms be able to work properly even under severe
weaknesses caused by low cost components to allow the realisation
of economic products.

Paper

SS.5.1
Title: AN IMPROVED FULLY PARALLEL STOCHASTIC GRADIENT ALGORITHM
FOR SUBSPACE TRACKING

Authors: Jeroen Dehaene (*), Marc Moonen (+), Joos Vandewalle (+)

Affiliation:  (*) Harvard University, Pierce Hall, Cruft lab 311,
29 Oxford street, Cambridge MA 02138, U.S.A.
(+) Katholieke Universiteit Leuven, E.E Dept. (ESAT),
K. Mercierlaan 94, 3001 Leuven, Belgium
email:  marc.moonen@esat.kuleuven.ac.be

Abstract:
A new algorithm is presented for principal component analysis and subspace
tracking, which improves upon classical stochastic gradient based
algorithms (SGA) as well as
several other related algorithms that have been presented in the literature.
The new algorithm is based on and inherits its main properties from a
continuous-time algorithm, closely related to the QR flow.
It gives the same estimates as classical SGA algorithms
but requires only O(N.k) operations per update instead of
O(N.k.k), where N is the dimension of the input vector
and k is the number of principal components to be estimated.
A parallel version with O(k) parallelism (processors) and
throughput O(1/N) and is straightforwardly derived.
A fully parallel version, with throughput independent of the
problem size O(1), may be obtained at the expense of O(N.N)
Paper

SS.5.2
A MINIMAL, GIVENS ROTATION BASED FRLS LATTICE ALGORITHM

Francois Desbouvries and Phillip A. Regalia

Departement Signal & Image
Institut National des Telecommunications
9 rue Charles Fourier
91011 Evry cedex, France
desbou@int-evry.fr,  regalia@int-evry.fr

Abstract:
We propose a new Givens rotation based
least-squares lattice algorithm.
Based on spherical trigonometry principles, this
algorithm turns out to be a normalized version of the fast QRD-based
least-squares lattice filter, introduced independently by
Ling and Proudler et al.
In constrast to those algorithms, the storage requirements of
the new algorithm are minimal (in the system theory sense).
From this, we show that the new algorithm satisfies
the backward consistency property, and hence enjoys stable
error propagation.
Paper

SS.5.3
A HIGHLY PARALLEL MULTICHANNEL FAST QRD-LS ADAPTIVE ALGORITHM

Athanasios A. Rontogiannis and Sergios Theodoridis

Department of Informatics
Division of Communications  and Signal Processing
University of Athens
GR-157 71 Zografou, GREECE
e-mail:{tronto,stheodor}@di.uoa.gr

A new fast multichannel QR decomposition (QRD) least squares (LS) adaptive
algorithm is presented in this paper. The algorithm deals with the general
case of channels with different number of delay elements and is based
exclusively on numerically robust orthogonal Givens rotations. The new
scheme processes each channel separately and as a result it comprises scalar
operations only. Moreover, the proposed algorithm is implementable on a
very regular systolic architecture and offers substantially reduced
computational complexity compared to previously derived multichannel fast
QRD schemes.
Paper

SS.5.4
Increasing the Performance of the LMS algorithm using an Adaptive Preconditioner.

I. K. Proudler, I.D. Skidmore, and J.G. McWhirter.

Rm. E506, DRA, St. Andrews Road, Malvern, Worcestershire, WR14 3PS, UK.
Tel. +44 1684 894228 Fax. +44 1684 896502
e-mail: proudler@signal.dra.hmg.gb

In this paper we outline a technique for increasing the convergence rate
of the LMS algorithm by means of a preconditioning filter which reduces
the eigenvalue spread of the input signal. Specifically we use a low order
linear prediction lattice filter followed by a tapped-delay-line as the
preconditioner. Some computer simulations are provided to demonstrate
the increased convergence rate of the new algorithm. (c) British Crown
Paper

SS.5.5
Stabilizing the LFTF algorithm by leakage control

Bernhard Nitsch and Stephan Binde

Institut fuer Netzwerk- und Signaltheorie
Fachgebiet Theorie der Signale
Merkstrasse 25
Germany

To stabilize the FTF algorithm the accumulation of numerical errors can be
prevented by introducing a leakage factor in the equation system.
In state space description the leakage factor causes a reduction of the eigenvalues of the linearized
error system matrix. By an appropriate choice of the leakage factor the eigenvalues can be
transformed into the unit circle of the z-plane resulting in a stable round-off error system.
The structure of the linearized error system matrix shall be analysed and
by comparing the Leakage FTF algorithm (LFTF) with the Stabilized FTF algorithm (SFTF) and the
NLMS algorithm in a real-time environment the success of this method is shown.
Paper

SS.5.6
PAST INPUT RECONSTRUCTION IN
BACKWARD CONSISTENT FAST LEAST-SQUARES ALGORITHMS

Phillip A. Regalia

Departement Signal & Image
Institut National des Telecommunications
9, rue Charles Fourier
F-91011 Evry cedex France
e-mail: regalia@galaxie.int-evry.fr

Abstract:
We present an analytic solution to the past input reconstruction
problem, which consists in describing all past input sequences which
would give rise to a given set of variables in fast least-squares algorithms,
whenever the variables in question are reachable.
Paper

SS.5.7

ASYMPTOTIC ANALYSIS OF THE UNDERDETERMINED RECURSIVE LEAST-SQUARES ALGORITHM

Authors: B. Baykal, O. Tanrikulu and A. G. Constantinides

Affiliation:    Signal Processing and Digital Systems Section
Dept. of EE. Eng., Imperial College of Sci., Tech. and Med.,
London SW7 2BT, UK, Email: b.baykal@ic.ac.uk

Abstract: The asymptotic analysis of the Underdetermined Recursive
Least-Squares (URLS) algorithm is performed. In particular, the behaviour
of the weight-error correlation matrix is investigated and the misadjustment
is calculated. For highly correlated input signals the misadjustment is shown
to be inversely proportional to the minimum eigenvalue of the underdetermined
order autocorrelation matrix. Simulations are included to justify the
conclusions.
Paper

SS.5.8
ROBUSTNESS AND CONVERGENCE OF ADAPTIVE SCHEMES IN BLIND EQUALIZATION
AND NEURAL NETWORK TRAINING

Ali H. Sayed and Markus Rupp

Ali H. Sayed, Department of Electrical and Computer Engineering,
University of California, Santa Barbara, CA 93106--9560,
sayed@ece.ucsb.edu
Markus Rupp, Wireless Technology Research  Dept.,
Lucent Technology, 791 Holmdel-Keyport Rd.,
Holmdel NJ 07733--0400, rupp@lucent.com

We pursue a time-domain feedback analysis of adaptive schemes with
nonlinear update relations. We consider commonly used algorithms in blind
equalization and neural network training and study their performance
in a purely deterministic framework. The derivation employs insights from
system theory and feedback analysis, and it clarifies the combined effects of
the step-size parameters and the nature of the nonlinear functionals on the
convergence and robustness performance of the adaptive schemes.
Paper

SS.5.9
MULTI-CHANNEL ADAPTIVE FILTERING APPLIED TO MULTI-CHANNEL ACOUSCTIC ECHO
CANCELLATION

Jacob Benesty (1), Pierre Duhamel (2), Yves Grenier (2)

(1) Lucent Technologies, Bell Labs Innovations, New Jersey, USA,
jb@research.att.com
(2) ENST, Dept. Signal, 46 rue Barrault, 75634 Paris Cedex 13, France
duhamel@sig.enst.fr, grenier@sig.enst.fr

This paper presents some new ways of deriving multi-channel (M-C) adaptive
algorithms in the context of M-C acoustic echo cancellation (AEC). We first
discuss the M-C identification problem which occurs in such systems by
distinguishing between the ideal case where the adaptive filters have the very
length of the impulse responses of the distant room and the real case.

These properties also explain some problems encountered with classical M-C
least mean squares (LMS) algorithm: straightforward generalization of the LMS
algorithm and the affine projection algorithm (APA) to the M-C case are easily
obtained. However, the resulting algorithms do not take into account the
cross-correlation between the input signals (such the M-C RLS algorithm),
hence  they do converge very slowly. Based on an original writing of the M-C
recursive  least squares (RLS) algorithm, we obtain useful properties that are
used to  overcome this problem, and we derive efficient algorithms in terms of
convergence rate.
Paper

SS.5.10
A NEW FREQUENCY DOMAIN EQUALIZER FOR
CHANNELS WITH LONG IMPULSE RESPONSE

Kostas Berberidis (*) and Jacques Palicot (#)

(*)  Computer Technology Institute (C.T.I.)
P.O. Box 1122
26110 Patras, GREECE
E-mail: berberid@cti.gr

(#)  C.C.E.T.T., SRA/DCS
4 rue du Clos Courtel
35512 Cesson Sevigne Cedex, FRANCE
E-mail: palicot@ccett.fr

ABSTRACT:

In this paper a recently introduced block Decision
Feedback Equalizer (DFE) is further studied and developed.
Moreover it is shown that the new technique is particularly
suitable for channel equalization in applications involving
channels with medium up to long impulse response.
The new equalizer, which is totally implemented in the
frequency domain, offers remarkable savings in computational
complexity as compared to the conventional time domain DFE.
Moreover the new technique results in a Symbol Error Rate
which is always lower (or much lower) with respect to that of
the existing frequency domain linear equalization techniques.

-------------------------------
Paper

SS.6.1
NONLINEAR FUZZY FILTERS: AN OVERVIEW

Fabrizio Russo

D.E.E.I. -  University of Trieste,
Via A. Valerio 10, Trieste I-34127, Italy
Tel.: +39-40-6763015, FAX : +39-40-6763460,
E-mail: rusfab@univ.trieste.it

Emergent techniques based on Fuzzy Logic have successfully entered the
area of nonlinear filters.  Indeed, a variety of methods have been recently
proposed in the literature which are able to perform detail-preserving
smoothing of noisy image data yielding better results than classical operators.
The aim of this paper is to present a selection of the most significant
contributions in this field focussing on their similarities and differences.
A brief introduction to the theory of fuzzy sets and systems is presented
in order to make these results available to non-fuzzy researchers too.
Paper

SS.6.2
DATA-DEPENDENT FILTERING BASED ON
IF-THEN RULES AND ELSE RULE

Akira Taguchi and Tomoaki Kimura
Department of Electrical and Electronic Engineering
Musashi Institute of Technology
Setagaya-ku, Tokyo 158, Japan
Tel: +81 3 3703 3111; Fax: +81 3 5707 2174
e-mail: taguchi(@ipc.musashi-tech.ac.jp

ABSTRACT

We have proposed fuzzy filters based on local
characteristics, in order to remove additive noise
while preserving signal edges. Fuzzy filters were
constructed by only IF-THEN rules. This paper
shows a novel fuzzy filter which is constructed by
not only IF- THEN rules but also ELSE rule. A lot
of IF-THEN rules which have the same
consequent, can be integrated into one ELSE rule.
As a results, introducing the ELSE rule can realize
increasing the local characteristics for the fuzzy
filter without increasing the number of IF-THEN
rules.
Paper

SS.6.3
FUZZY CELL HOUGH TRANSFORM

Vassilios Chatzis and Ioannis Pitas

Department of Informatics
University of Thessaloniki, 54006 Thessaloniki, GREECE
Tel, fax: +30-31-996304
e-mail: pitas@zeus.csd.auth.gr

In this paper a new variation of Hough Transform
is proposed. It can be used to detect shapes or curves in an image,
with better accuracy, especially in noisy images.
It is based on a fuzzy split of the Hough Transform parameter
space. The parameter space is split into fuzzy cells which
are defined as fuzzy numbers. This fuzzy split of the
parameter space provides the advantage to use the uncertainty
of the contour points location, which is increased
when noisy images have to be used. Moreover,
the computation time is slightly increased by this method,
in comparison with classical Hough Transform.
Paper

SS.6.4
FUZZY CENTER WEIGHTED MEDIAN FILTERS

Akira Taguchi and Nobunori Izawa
Department of Electrical and Electronic Engineering
Musashi Institute of Technology
Setagaya-ku, Tokyo 158, Japan
Tel: +81 3 3703 3111; Fax: +81 3 5707 2174
e-mail: taguchi@ipc.musashi-tech.ac.jp

ABSTRACT

Stack filters are a class of nonlinear filters, first introduced by
Wedent et. al. Stack filters perform well in many situations
where linear filters fail. Stack filters include rank order filters,
morphological filters and weighted median filters. The stack
filter is defined by a Boolean function. The output of Boolean
functions is restricted two values (i.e., "0" or "1"). Intuitively,
one would expect better performance for stack filters, if the
output of Boolean functions is defined from 0 to 1
continuously. We call this Boolean functions fuzzy Boolean
functions. We discuss about fuzzy center weighted median
(FCWM) filters which is one of the simplest fuzzy stack
filters in this paper. Two design methods are shown in this
paper.
Paper

SS.6.5
A FUZZY EXPERT SYSTEM FOR LOW LEVEL IMAGE SEGMENTATION

Mauro Barni*, Sandro Rossi*, Alessandro Mecocci**

*Department of Electronic Engineering, University of Florence
Via di Santa Marta, 3 - 50139 Firenze, ITALY

**Department of Electronic Engineering, University of Siena
Via Roma, 56 - 53100 Siena, ITALY

e-mail: barni@cosimo.ing.unifi.it

Abstract. In this paper a general purpose fuzzy expert system is presented for
low level image segmentation. By means of approximate reasoning based
on fuzzy logic, the criticality of the choice of the several thresholds and
parameters which usually must be tuned to make the expert system work properly
is reduced. More specifically, it is proved that, by keeping
constant the number of rules the expert system consists of, the fuzzy
approach permits to build a more general system, capable of giving
satisfactory results for a large number of images stemming from different
applications. The validity of the approach is demonstrated by
comparing the effectiveness of a classical expert system with that of
its corresponding fuzzy version. Upon analysis of the results, the superiority
of the fuzzy system in terms of robustness and generality comes out.
Paper

SS.6.6
INTEGRATION OF LINGUISTIC KNOWLEDGE FOR COLOUR IMAGE SEGMENTATION

T. CARRON, P. LAMBERT

Laboratoire d'Automatique et de MicroInformatique Industrielle
LAMII/CESALP - Universite de Savoie - B.P 806 - F.74016 Annecy Cedex (France)
(CNRS G1047 - Information-Signal-ImageS)
e-mail: carron@univ-savoie.fr - lambert@univ-savoie.fr

The Hue, Chroma, Intensity (HCI) space is well suited to colour images
segmentation processing. In this paper, we used fuzzy logic for integrating
specific knowledge of the Hue component. Based upon several linguistic
rules which built a symbolic cooperation between Hue and Intensity according
to Chroma, a region growing segmentation with fuzzy aggregation is proposed.
This fuzzy segmentation is compared with a technique using a Fuzzy C-Means
algorithm in different colour spaces.
Paper

SS.6.7
FUSION OF DATA FROM FUZZY INTEGRAL-BASED ACTIVE AND PASSIVE
COLOUR STEREO VISION SYSTEMS FOR CORRESPONDENCE IDENTIFICATION

Alois Knoll, Ralf Schroeder, and Andre Wolfram

University of Bielefeld, Faculty of Technology,
Department of Computer Science,
Postfach 10 01 31, D-33501 Bielefeld, Germany
e-mail: {knoll,andre}@techfak.uni-bielefeld.de

As shown in our previous work, an approach using the fuzzy-integral [3] can be
applied to solving the correspondence problem of active colour stereo vision
systems [2]. Evaluating the similarity measure derived in [2] enables the
identification of a correct match or otherwise indicates at least several
possible matches. To reduce the remaining ambiguity further, the novel approach
presented here uses data fusion techniques to make use of additional fuzzy
feature-based information gathered by passive colour stereo procedures. Our
experimental results, which are discussed in the paper, indicate that this new
approach is considerably more effective than the approach using only
intensity-based information for determining the similarity of line blocks in
colour stereo images. We conclude the paper with a discussion of the potential
of the method and directions of possible future research.
Paper

SS.6.8
FUZZY CLUSTERING OF DIGITAL IMAGES BY EXPLOITING DENSITOMETRIC AND TOPOLOGICAL
INFORMATION

M. Mari, C. Garcia and S. Dellepiane

Department of Biophisical and Electrionic Engineering (DIBE)
University of Genoa
via Opera Pia, 11a, 16145 Genova, Italy
Tel. +39 10 3532754; fax: +39 10 3532134
e-mail: silvana@dibe.unige.it

ABSTRACT

Topological features are very seldom exploited in image processing, also
due to the complexity of their extraction. Even when topological features
are used, densitometric information are usually not considered at the
same time. The simultaneous exploitation of such features, as proposed
in the paper, allows a more appropriate automatic processing of digital
images. A novel image segmentation approach is presented (based on fuzzy
clustering) that exploits topological and densitometric image features.
The novelty of such an image segmentation consists mainly in using easy
and fast computation methods, to improve the handling of any digital image,
whenever automatic segmentation or data reduction processing is required.
Paper