ICASSP 97, Table of Contents
1997 International Conference on Acoustics, Speech and Signal Processing

Title: Space-Time Processing for Wireless Communications
Authors: 
Arogyaswami Paulraj, Stanford University
Volume: 1, Page: 1
Abstract:
This paper reviews space-time signal processing in mobile wireless
communications. Space-time processing refers to the signal processing
performed in the spatial and temporal domain on signals received at
or transmitted from an antenna array, in order to improve performance
of wireless networks. We focus on antenna arrays deployed at the base
stations since such applications are of current practical interest.
**
Title: Variability Of Performance In Video Coding
Authors: 
Don Pearson, University of Essex
Volume: 1, Page: 5
Abstract:
Modern video compression techniques exhibit variability of performance
as a function of time. Studies are reported of viewers reactions to
this variability, which indicate a sensitivity to particular features.
Some interesting conclusions emerge for future work in video coding.
**
Title: Expert Summaries
Authors: 
Renato de Mori, C.E.R.I.
Hermann Ney, University of Technology (RWTH), Aachen
Hans Georg Musmann, University of Hannover
Rama Chellapa, University of Maryland
Mark J.T. Smith, Georgia Institute of Technology
John R. Treichler, Applied Signal Technology, Inc.
Georgios B. Giannakis, University of Virginia
Michael D. Zoltowski, Purdue University
Volume: 1, Page: 9
Abstract:
Leading experts in their fields summarize the most relevant new ideas
from submitted papers in the fields of speech processing, digital signal
processing, image and multidimensional signal processing, and statistical
and array processing.
**
Title: Expanding Team Experiences in DSP Education
Authors: 
Delores Etter, University of of Colorado
Geoffrey C. Orsak, George Mason University
Volume: 1, Page: 11
Abstract:
Since practicing engineers work in multidisciplinary teams, it is important
that universities provide as many teaming experiences as possible.
In this paper, we present some of the advantages and disadvantages
of traditional teaming approaches.  We then present issues related
to virtual teaming - the teaming of students from geographically distributed
locations.  Virtual teaming adds a new dimension to the teaming experiences
that universities can provide to students to better equip them for
the environment in which they will work in research positions and in
industry.  Experiences with a three-year program in virtual teaming
between the University of Colorado and George Mason University will
be presented.
**
Title: Interactive Classroom For DSP/Communication Courses
Authors: 
Huseyin Abut, School of Applied Science
Yusuf Ozturk, San Diego State University
Volume: 1, Page: 15
Abstract:
In this study, we present a new classroom environment to conduct digital
signal processing and communication systems courses. Key features of
the model are the collaborating instructor embracing students, a smart
classroom equipped with a `Whiteboard` and advanced telecommunication
networks, electronic textbook, and other resources, World Wide Web
(WWW), Matlab, and other on-line tools. The underlying assumptions
of the educational process are teambuilding instead of independent
learning, collaborating/ supervising instructor, lateral curriculum
instead of a vertical curriculum, and idea-to-product design concept.
We will present a sample lecture in the proposed interactive classroom,
where the concept of eye diagrams in regenerative repeaters will be
presented from the first author`s text using matlab and WWW.
**
Title: Experiences in Teaching DSP First in the ECE Curriculum
Authors: 
James H. McClellan, Georgia Institute of Technology
Ronald W. Schafer, Georgia Institute of Technology
Mark A. Yoder, Rose-Hulman Institute of Technology
Volume: 1, Page: 19
Abstract:
In this paper we describe experiences gained from teaching an introductory
electrical engineering course based on digital signal processing rather
than the traditional first course in analog circuit theory.  We will
discuss our motivation for teaching DSP first, before covering analog
circuits and systems. We will describe the style of the course and
point out difficulties, as well as advantages, in this organization
of basic material.  Finally, we will make some comments about extending
this approach to encompass a wider range of students from other disciplines.
**
Title: Analog Signal Processing: A Replacement for the Sophomore-Level
Circuit Analysis Course
Authors: 
David C. Munson, University of Illinois
Volume: 1, Page: 23
Abstract:
A new undergraduate curriculum in electrical engineering has been adopted
by the Department of Electrical and Computer Engineering at the University
of Illinois.  Major changes have been incorporated, including a redistribution
of the circuits and signal processing topics within the curriculum.
After giving an overview of the new curriculum, this paper focuses
on a new, required sophomore-level course on analog signal processing.
This course combines material from the traditional course on circuit
analysis with material on continuous-time signals and systems. Students
completing this course can study digital signal processing as first-semester
juniors, which leaves ample time for more advanced signal and image
processing courses in future semesters.
**
Title: Re-engineering The Electrical Engineering Curriculum
Authors: 
Sanjit K. Mitra, University of California
Volume: 1, Page: 27
Abstract:
Three specific programs are suggested to modify the electrical engineering
curriculum to keep up with the dramatic technological developments
of recent years.  One of the programs is a five-year combined BS/MS
program which permits the student to specialize in more than one field.
The second proposal is to restructure the BS program into a multi-track
program.  The third one is an internship-in-industry program to provide
the student with a meaningful and valuable real-world design experience
before graduation.
**
Title: Structural Subband Decomposition: A new Concept in Digital Signal
Processing
Authors: 
Sanjit K. Mitra, University of California
Volume: 1, Page: 31
Abstract:
This paper introduces the concept of structural subband decomposition
of sequences, a generalization of the polyphase decomposition of sequences,
and outlines a number of applications of this concept, such as efficient
FIR filter design and implementation, adaptive filtering, and fast
computation of discrete transforms.
**
Title: A New Algorithm for the Generalized Eigenvalue Problem
Authors: 
Knut Huper, University of Wurzburg
Uwe Helmke, University of Wurzburg
Volume: 1, Page: 35
Abstract:
The problem of finding the generalized eigenvalues and eigenvectors
of a pair of real symmetric matrices A and B, with B>0, can be viewed
as a smooth optimization problem on a smooth manifold.  We present
a cost function approach to the generalized eigenvalue problem which
is posed on the product of the n-sphere and Euclidian space R.  The
critical point set of this cost function is studied. An algorithm is
presented based on constrained optimization. A proof of local quadratic
convergence is given.
**
Title: A lattice structure for perfect reconstruction linear time varying
filter banks with all pass analysis banks
Authors: 
Soura Dasgupta, University of Iowa
Chris Schwarz, University of Iowa
Minyue Fu, University of Newcastle
Volume: 1, Page: 39
Abstract:
We consider a multi-input, multi-output lattice realization for linear
time-varying analysis banks which are all pass.  Such a realization
has been given for LTI systems; and under certain conditions, we show
how it generalizes to the LTV case.  Moreover, our implementation is
simpler than the existing LTI version.  Finally, we describe the anticausal
inverse of a lattice realization which is used in the synthesis bank.
**
Title: Algorithm Design for Structured Systems: Application to Pole Placement
Authors: 
Steffen Paul, Technical University of Munich
Josef A. Nossek, Technical University of Munich
Volume: 1, Page: 43
Abstract:
Numerical algorithms for signal processing and control are quite often
constructed by intuition.  When the system to be designed contains
algebraic or other invariants, then these constraints can be exploited
to find appropriate transformations. The transformations in system
theory are usually Lie groups.  One has to find Lie groups which are
consistent with the invariants.  We show, how this point of view can
be applied to construct pole placement algorithms for symmetric and
skew-symmetric realizations.  However, Lie group theory only reveals
the appropriate transformations but is not able to reduce the design
process to a trivial task.  The problem discussed here does also show
this limitation.
**
Title: Actions of noncompact groups and algorithm design: A case study
Authors: 
Klaus Diepold, IDT
Rainer Pauli, Technical University of Munich
Volume: 1, Page: 47
Abstract:
Numerical matrix computations involving actions of noncompact transformation
groups are known to produce numerical problems since the elements of
the pertaining matrix representations are inherently unbounded. In
this case study we analyse numerical problems occuring in a class of
algorithms that is based on actions of the pseudo-orthogonal group
 O_n,m -- a group that is noncompact (hyperbolic geometry) and well
established in signal processing (Schur methods). As a major result,
it is shown how to exploit the additional degrees of freedom in defining
coordinate frames in a Grassmannian setting in order to impose an a
priori bound on the norm of the transformation matrices. This way,
numerically disastrous situations can be circumvented systematically.
Hence, it becomes possible to develop modified algorithms which exhibit
superior numerical performance for a large class of problems based
on e.g. hyperbolic transformations.
**
Title: Discretization Issues for the Design of Optimal Blind Algorithms
Authors: 
Rodney A. Kennedy, Australian National University
Deva K. Borah, Australian National University
Zhi Ding, Auburn University
Volume: 1, Page: 51
Abstract:
The performance and complexity of blind algorithms in a digital receiver
is dependent on the prefilter prior to discretization of the received
continuous time signal and the sampling rate. This paper shows that
symbol spaced blind equalization algorithms are in general sub-optimal,
since a matched filter cannot be used. We show that, for fractionally
spaced equalizers, the prefilter can be a general low-pass filter and
does not need to be matched to the unknown channel. This flexibility
on choosing the prefilter can result in different discrete time models
with different complexities for the signal processing algorithms to
follow. As for example, a simpler whitening filter design which is
needed for the success of several important blind equalization algorithms
can be realized using this flexibility.
**
Title: Continuous-Time Envelope-Constrained Filter Design via Laguerre
Filter and H_(infinity) Optimization Methods
Authors: 
Zhuquan Zang, ATRI, Curtin University of Technology
Antonio Cantoni, ATRI, Curtin University of Technology
Koklay Teo, ATRI, Curtin University of Technology
Volume: 1, Page: 55
Abstract:
Envelope-constrained filtering is concerned with the design of a time-invariant
filter to process a given input signal such that the noiseless output
of the filter is guaranteed to lie within a prespecified output mask.
In this paper, using Laguerre filters and  H_(infinity) optimization
techniques, the continuous-time envelope-constrained filter design
problem has been reformulated and solved as a constrained  H_(infinity)
model-matching problem.  To illustrate the effectiveness of the design
method, a numerical example is presented which deals with the design
of an equalization filter for a digital transmission channel.
**
Title: Local adaptive algorithms for information maximization in neural
networks, and application to source separation
Authors: 
Jeroen Dehaene, K.U.Leuven
Nanayaa Twum-Danso, Harvard University
Volume: 1, Page: 59
Abstract:
Information theoretic criteria for neural network adaptation laws have
recently become an important focus of attention.  We consider the problem
of adaptively maximizing the entropy of the outputs of a deterministic
feedforward neural network with real valued stochastic input signals,
as considered by Bell and Sejnowski. We give a new explanation for
the relevance of output information (entropy) maximization for source
separation applications and reinterpret Bell and Sejnowski's approach
in a more general context of probability density estimation. This insight
is the basis for a generalization of the approach, and we consider
a family of gradient based algorithms.
**
Title: Quick Aggregation of Markov Chain Functionals via Stochastic Complementation
Authors: 
Kutluyil Dogancay, University of Melbourne
Vikram Krishnamurthy, University of Melbourne
Volume: 1, Page: 63
Abstract:
The paper presents a quick and simplified aggregation method for a
large class of Markov chain functionals based on the concept of stochastic
complementation. Aggregation results in a reduction in the number of
Markov states by grouping them into a smaller number of aggregated
states, thereby producing a considerable saving on computational complexity
associated with maximum likelihood parameter and state estimation for
hidden Markov models. The importance of the proposed aggregation method
stems from the ease with which Markov chains with a large number of
states can be aggregated. Three Markov chain functionals which have
widespread use are considered to illustrate the application of our
aggregation method.
**
Title: A rank preserving flow algorithm for quadratic optimization problems
subject to quadratic equality constraints
Authors: 
John B. Moore, Systems Engineering, ANU
Danchi Jiang, Dept. MAE. Chinese University
Volume: 1, Page: 67
Abstract:
This paper concerns quadratic programming problems subject to quadratic
equality constraints such as arise in broadband antenna array signal
processing and elsewhere. At first, such a problem is converted into
a semidefinite programming problem with a rank constraint.  Then, a
rank preserving flow is used to accommodate the rank constraint. The
associated gradient formulas are carefully developed.  The convergence
of the resulted algorithm is also guaranteed. Our approach is demonstrated
by a numerical experiment.
**
Title: VERBMOBIL: The Combination Of Deep And Shallow Processing For
Spontaneous Speech Translation
Authors: 
Thomas Bub, DFKI
Wolfgang Wahlster, DFKI
Alex Waibel, Carnegie Mellon University
Volume: 1, Page: 71
Abstract:
Verbmobil is a speech-to-speech translation system for spontaneously
spoken negotiation dialogs. The actual system translates 74.2% of spontaneously
spoken German input. We give an overview of the Verbmobil system. After
the introduction of the Verbmobil scenario and the unique constraints
of the project, we describe the underlying system architecture and
its realization. The progress that was achieved on the end-to-end translation
rate owes much to the increase of the word recognition rate from 45%
in 1993 to 87% in 1996. But in order to achieve the envisaged coverage
on the incertain speech recognizer output, deep and shallow approaches
to the analysis and transfer problem had to be combined.
**
Title: Prosodic Processing and its Use in Verbmobil
Authors: 
Heinrich Niemann, University of Erlangen
Elmar Noth, University of Erlangen
Andreas Kiessling, University of Erlangen
Ralf Kompe, University of Erlangen
Anton Batliner, L.M.-Univ. Munchen
Volume: 1, Page: 75
Abstract:
We present the prosody module of the VERBMOBIL speech-to-speech translation
system, the world wide first complete system, which successfully uses
prosodic information in the linguistic analysis.  This is achieved
by computing probabilities for clause boundaries, accentuation, and
different types of sentence mood for each of the word hypotheses computed
by the word recognizer. These probabilities guide the search of the
linguistic analysis. Disambiguation is already achieved during the
analysis and not by a prosodic verification of different linguistic
hypotheses. So far, the most useful prosodic information is provided
by clause boundaries. These are detected with a recognition rate of
94%. For the parsing of word hypotheses graphs, the use of clause boundary
probabilities yields a speed-up of 92% and a 96% reduction of alternative
readings.
**
Title: The Language Components in Verbmobil
Authors: 
Hans Ulrich Block, Siemens AG
Volume: 1, Page: 79
Abstract:
This paper gives an overview over the main problems and their solutions
in the language components of the Verbmobil speech translation system.
Interpretation of spontaneously spoken language has to take into account
that syntax and semantics differ from written language, that punctuation
is missing, that accent and intonation have effects on the meaning
and the translation, that the output of the speech recognizer may be
noisy and that speakers produce errors due to distraction. The Verbmobil
interpretation and translation components try to attack these problems
by means of a grammar for spoken language, heavy use of prosodic information,
a syntactic search on word hypothesis graphs and a shallow robust fall
back translation device that is used in case the "deep" translation
fails.
**
Title: The Karlsruhe-Verbmobil Speech Recognition Engine
Authors: 
Michael Finke, University of Karlsruhe
Petra Geutner, University of Karlsruhe
Hermann Hild, University of Karlsruhe
Thomas Kemp, University of Karlsruhe
Klaus Ries, University of Karlsruhe
Martin Westphal, University of Karlsruhe
Volume: 1, Page: 83
Abstract:
Verbmobil, a German research project, aims at machine translation of
spontaneous speech input. The ultimate goal is the development of a
portable machine translator that will allow people to negotiate in
their native language. Within this project the University of Karlsruhe
has developed a speech recognition engine that has been evaluated on
a yearly basis during the project and shows very promising speech recognition
word accuracy results on large vocabulary spontaneous speech.  In this
paper we will introduce the Janus Speech Recognition Toolkit underlying
the speech recognizer. The main new contributions to the acoustic modeling
part of our 1996 evaluation system -- speaker normalization, channel
normalization and polyphonic clustering -- will be discussed and evaluated.
Besides the acoustic models we delineate the different language models
used in our evaluation system: Word trigram models interpolated with
class based models and a separate spelling language model were applied.
As a result of using the toolkit and integrating all these parts into
the recognition engine the word error rate on the German Spontaneous
Scheduling Task (GSST) could be decreased from 30% word error rate
in 1995 to 13.8% in 1996.
**
Title: An Experiment On Korean-To-English And Korean-To-Japanese Spoken
Language Translation
Authors: 
Jae-Woo Yang, ETRI
Jun Park, ETRI
Volume: 1, Page: 87
Abstract:
We have implemented a Korean-to-English and Korean-to-Japanese spoken
language translation system prototype. The system can translate speech
in travel planning domain with 5,000 word vocabulary.  In our prototype,
we concentrate on how to transfer the intention of a user to the partner
in spite of current limitation of spoken language processing technology.
We measured the end-to-end performance of the prototype to test whether
the output of the system is understandable using a subjective measure.
We also used an objective measure to evaluate the system performance
and found that it generates coherent result with the subjective test.
The test result shows that the user can understand the output even
in the case that the system cannot translate speech correctly. Thus
it is important to provide even partially correct translation output
to the user, in order not to neglect the possibility that the user
can infer the intended message using the context and his/her intelligence.
**
Title: Multilingual Person to Person Communication at IRST
Authors: 
Bianca Angelini, IRST
Mauro Cettolo, IRST
Anna Corazza, IRST
Daniele Falavigna, IRST
Gianni Lazzari, IRST
Volume: 1, Page: 91
Abstract:
This paper refers to a machine-mediated person-to-person multilingual
communication system.  Stress is put on robustness, that is the ability
of the system to preserve communication even in presence of the variability
and errors typical of spoken language systems. The statistical approach
is adopted not only at the acoustic level, but also for the linguistic
processing. Therefore, while an overview of the global architecture
will be briefly introduced, the focus will be put on the acoustic recognizer
and the understanding module. Experimental evaluations complete the
presentation.
**
Title: Fast Word-Graph Generation For Spontaneous Conversational Speech
Translation
Authors: 
Tohru Shimizu, ATR-ITL
Harald Singer, ATR-ITL
Yoshinori Sagisaka, ATR-ITL
Volume: 1, Page: 95
Abstract:
This paper introduces the latest advances in research at ATR on speech
translation for spontaneous conversations, especially focusing on speech
recognition efforts. For recognition, we employ a word search technique
that generates moderate sized word graphs in real-time. To cope with
a variety in length of utterances, e.g., word, phrase, sentence fragment,
sentence, and concatenated sentences in spontaneous speech, we have
adopted a two pass search strategy that uses variable-order word n-gram
statistics in the first stage and task dependent language constraints
in the second stage. This strategy is evaluated using the ``ATR Travel
Arrangement'' corpus.
**
Title: JANUS-III: Speech-to-Speech Translation in Multiple Languages
Authors: 
Alon Lavie, Carnegie Mellon University
Alex Waibel, Carnegie Mellon University
Lori Levin, Carnegie Mellon University
Michael Finke, Carnegie Mellon University
Donna Gates, Carnegie Mellon University
Marsal Gavalda, Carnegie Mellon University
Torsten Zeppenfeld, Carnegie Mellon University
Puming Zhan, Carnegie Mellon University
Volume: 1, Page: 99
Abstract:
This paper describes JANUS-III, our most recent version of the JANUS
speech-to-speech translation system.  We present an overview of the
system and focus on how system design facilitates speech translation
between multiple languages, and allows for easy adaptation to new source
and target languages.  We also describe our methodology for evaluation
of end-to-end system performance with a variety of source and target
languages.  For system development and evaluation, we have experimented
with both push-to-talk as well as cross-talk recording conditions.
To date, our system has achieved performance levels of over 80% acceptable
translations on transcribed input, and over 75% acceptable translations
on speech input recognized with a 75-90% word accuracy. Our current
major research is concentrated on enhancing the capabilities of the
system to deal with input in broad and general domains.
**
Title: State-Transition Cost Functions and an Application to Language
Translation
Authors: 
Hiyan Alshawi, AT&T Labs
Adam L. Buchsbaum, AT&T Labs
Volume: 1, Page: 103
Abstract:
We define a general method for ranking the solutions of a search process
by associating costs with equivalence classes of state transitions
of the process. We show how the method accommodates models based on
probabilistic, discriminative, and distance cost functions, including
assignment of costs to unseen events.  By applying the method to our
machine translation prototype, we are able to experiment with different
cost functions and training procedures, including an unsupervised procedure
for training the numerical parameters of our English-Chinese translation
model.  Results from these experiments show that the choice of cost
function leads to significant differences in translation quality.
**
Title: Hybrid language processing in the Spoken Language Translator
Authors: 
Manny Rayner, SRI International
David M. Carter, SRI International
Volume: 1, Page: 107
Abstract:
The paper presents an overview of the Spoken Language Translator (SLT)
system's hybrid language-processing architecture, focussing on the
way in which rule-based and statistical methods are combined to achieve
robust and efficient performance within a linguistically motivated
framework. In general, we argue that rules are desirable in order to
encode domain-independent linguistic constraints and achieve high-quality
grammatical output, while corpus-derived statistics are needed if systems
are to be efficient and robust; further, that hybrid architectures
are superior from the point of view of portability to architectures
which only make use of one type of information. We address the topics
of ``multi-engine'' strategies for robust translation; robust bottom-up
parsing using pruning and grammar specialization; rational development
of linguistic rule-sets using balanced domain corpora; and efficient
supervised training by interactive disambiguation.  All work described
is fully implemented in the current version of the SLT-2 system.
**
Title: Finite-State Speech-to-Speech Translation
Authors: 
Enrique Vidal, DSIC UPV
Volume: 1, Page: 111
Abstract:
A fully integrated approach to Speech-Input Language Translation in
limited-domain applications is presented.  The mapping from the input
to the output language is modeled in terms of a  finite state translation
model which is learned from examples of input-output sentences of the
task considered.  This model is tightly  integrated with standard acoustic-phonetic
models of the input language and the resulting global model directly
supplies, through Viterbi search, an optimal output-language sentence
for each input-language utterance.  Several extensions to this framework,
recently developed to cope with the increasing difficulty of translation
tasks, are reviewed.  Finally, results for a task in the framework
of hotel front-desk communication, with a vocabulary of about 700 words,
are reported.
**
Title: An Experimental Bidirectional Japanese/English Interpreting Video
Phone System Using Internet.
Authors: 
Shoji Hiraoka, MRIT
Masakatsu Hoshimi, MRIT
Kenji Matsui, CRL
Jean-Claude Junqua, STL
Volume: 1, Page: 115
Abstract:
In this paper we report on an experimental bidirectional Japanese/English
interpreting video phone system using Internet. We particularly emphasize
the motivation for this work, the task, and the experiments conducted.
Using in house technology developed both in Japan and in the United
States, we demonstrated an Internet home shopping application where
an American shop assistant and a Japanese customer engaged in task-directed
dialogues, using their native languages. The experiments showed that
when users are familiar with the application language, a natural interaction
can be obtained.
**
Title: From Neural Networks to Neural Strategies
Authors: 
Christian Goerick, Ruhr-Univ. Bochum
Bernhard Sendhoff, Ruhr-Univ. Bochum
Werner von Seelen, Ruhr-Univ. Bochum
Volume: 1, Page: 119
Abstract:
Artificial neural network have evolved from their biologically inspired
roots to a well established means to solve a broad spectrum of engineering
problems. The embedding into modern statistics has provided the necessary
theoretical foundation for challenging engineering tasks, such as advanced
real-time image and signal processing. These are exemplary demonstrations
for the applicability of this approach to complex information processing.
However, the large number of applications must not obscure the fact
that there are some major unsolved problems concerning neural networks.
There are still no satisfactorily constructive ways to determine the
optimal structure (elements as well as organization) or the learning
and evaluation dynamics. The ongoing research addresses these problems.
In addition to pursuing this direction, one can ask, what other lessons
we can learn from biology concerning complex information processing.
Our goal in this paper is to sketch a possible way from neural networks
to more comprehensive neural strategies.
**
Title: Neural And Traditional Techniques In Diagnostic ECG Classification.
Authors: 
Rosaria Silipo, DSI
Giovanni Bortolan, DSI
Volume: 1, Page: 123
Abstract:
Neural and traditional techniques have been compared for the particular
task of automatic ECG analysis. A large validated ECG database has
been used. Statistical methods, neural architectures with supervised
and unsupervised learning, and a neuro-fuzzy architecture have been
considered. The results from the connectionist approach are always
at least comparable with those coming from more traditional classification
methods. But the best performances have been obtained by the combination
of the connectionist with the fuzzy approach.
**
Title: Unsupervised Learning for Blind Source Separation: an Information-Theoretic
Approach
Authors: 
Dragan Obradovic, Siemens, Munchen
Gustavo Deco, Siemens, Munchen
Volume: 1, Page: 127
Abstract:
This paper provides a detailed and rigorous analysis of the two commonly
used methods for redundancy reduction: Linear Independent Component
Analysis (ICA) and Information Maximization (InfoMax). The paper shows
analytically that ICA based on the Kullback-Leibler information as
a mutual information measure and InfoMax lead to the same solution
if the parameterization of the output nonlinear functions in the latter
method is sufficiently rich. Furthermore, this work briefly discusses
the alternative redundancy measures not based on the Kullback-Leibler
information distance and Nonlinear ICA. The practical issues of applying
ICA and InfoMax are also discussed.
**
Title: Applications of Neural Blind Separation to Signal and Image Processing
Authors: 
Juha Karhunen, Helsinki University of Technology
Aapo Hyvarinen, Helsinki University of Technology
Ricardo Vigario, Helsinki University of Technology
Jarmo Hurri, Helsinki University of Technology
Erkki Oja, Helsinki University of Technology
Volume: 1, Page: 131
Abstract:
In blind source separation one tries to separate statistically independent
unknown source signals from their linear mixtures without knowing the
mixing coefficients. Such techniques are currently studied actively
both in statistical signal processing and unsupervised neural learning.
In this paper, we apply neural blind separation techniques developed
in our laboratory to extraction of features from natural images and
to separation of medical EEG signals. The new analysis method yields
features that describe the underlying data better than for example
classical principal component analysis. We briefly discuss difficulties
related with real-world applications of blind signal processing, too.
**
Title: Communications and Neural Networks: Theory and Practice
Authors: 
Mark D. Plumbley, KCL
Volume: 1, Page: 135
Abstract:
In this paper we shall see that neural networks and communications
are interlinked in a number of ways, towards the goal of efficient
communication of information. One concrete example of this is the use
of neural networks to ensure efficient use of communication channels,
through connection admission control in ATM networks. In addition,
however, efficient communication is also important within a decision
making system such as a neural network. Finally we examine what type
of neural network solutions are suggested by this approach.
**
Title: Robust Vector Quantization by Competitive Learning
Authors: 
Joachim M. Buhmann, University of Bonn
Thomas Hofmann, University of Bonn
Volume: 1, Page: 139
Abstract:
Competitive neural networks can be used to efficiently quantize image
and video data. We discuss a novel class of vector quantizers which
perform noise robust data compression. The vector quantizers are trained
to simultaneously compensate channel noise and code vector elimination
noise. The training algorithm to estimate code vectors is derived by
the maximum entropy principle in the spirit of deterministic annealing.
We demonstrate the performance of noise robust codebooks with compression
results for a teleconferencing system on the basis of a wavelet image
representation.
**
Title: Recognizing faces from a new viewpoint
Authors: 
Thomas Vetter, Max-Planck-Institut, Tubingen
Volume: 1, Page: 143
Abstract:
A new technique is described for recognizing faces from new viewpoints.
From a single 2D image of a face synthetic images from new viewpoints
are generated and compared to stored views. A novel 2D image of a face
can be computed without knowledge about the 3D structure of the head.
The technique draws on prior knowledge of faces based on example images
of other faces seen in different poses and on a single generic 3D model
of a human head.  The example images are used to learn a pose-invariant
shape and texture description of a new face. The 3D model is used to
solve the correspondence problem between images showing faces in different
poses. The performance of the technique is tested on a date set of
200 faces of known orientation for rotations up to 90 degree.
**
Title: Hybrid Optimization of Feedforward Neural Networks for Handwritten
Character Recognition
Authors: 
Wolfgang Utschick, Technical University of Munich
Josef A. Nossek, Technical University of Munich
Volume: 1, Page: 147
Abstract:
An extension of a feedforward neural network is presented. Although
utilizing linear threshold functions and a boolean function in the
second layer, signal processing within the neural network is real.
After mapping input vectors onto a discretization of the input space,
real valued features of the internal representation of pattern are
extracted. A vectorquantizer assigns a class hypothesis to a pattern
based on its extracted features and adequate reference vectors of all
classes in the decision space of the output layer. Training consists
of a combination of combinatorial and convex optimization. This work
has been applied to a standard optical character recognition task.
Results and comparison to alternative approaches are presented.
**
Title: Reading Checks with multilayer graph transducter networks
Authors: 
Yann LeCun, AT&T Labs
Leon Bottou, AT&T Labs
Yoshua Bengio, AT&T Labs
Volume: 1, Page: 151
Abstract:
We propose a new machine learning paradigm called Multilayer Graph
Transformer Network that extends the applicability of gradient-based
learning algorithms to systems composed of modules that take graphs
as input and produce graphs as output.  A complete check reading system
based on this concept is described.  The system combines convolutional
neural network character recognizers with graph-based stochastic models
trained cooperatively at the document level.  It is deployed commercially
and reads million of business and personal checks per month with record
accuracy.
**
Title: Neural Networks For Process Control In Steel Manufacturing
Authors: 
Martin Schlang, Siemens AG, Munich
Einar Broese, Siemens AG, Munich
Bjoern Feldkeller, Siemens AG, Munich
Otto Gramckow, Siemens AG, Munich
Michael Jansen, Siemens AG, Munich
Thomas Poppe, Siemens AG, Munich
Clemens Schaeffner, Siemens AG, Munich
Guenter Soergel, Siemens AG, Munich
Volume: 1, Page: 155
Abstract:
Neural Networks are particularly suitable for the approximation of
non-linear time-variant functions. Due to their learning capabilities,
they have proven useful in control applications for complex industrial
processes. In collaboration with the Corporate Research and Development
Department, the Siemens Industrial and Building Systems Group developed
Neural Network applications for the steel industry, resulting in a
more economic use of resources and an improvement of productivity.
At this time Siemens has installed more than 100 neural nets world
wide at different plants.
**
Title: A Neuro-Dynamic Programming Approach to Admission Control in ATM
Networks: The Single Link Case
Authors: 
Peter Marbach, LIDS, MIT
John N. Tsitsiklis, LIDS, MIT
Volume: 1, Page: 159
Abstract:
We are interested in solving large-scale Markov Decision Problems.
The classical method of Dynamic Programming provides a mathematical
framework for finding optimal solutions for a given Markov Decision
Problem. However, for Dynamic Programming algorithms become computationally
infeasible when the underlying Markov Decision Problem evolves over
a large state space. In recent years, a new methodology, called Neuro-Dynamic
Programming, has emerged which tries to overcome this ``curse of dimensionality''.
We present how Neuro-Dynamic Programming can be applied to the Admission
Control Problem for a single link in an ATM environment. Based on results
obtained through Neuro-Dynamic Programming, we derive a heuristic ``Threshold''
policy. Performances of the policies obtained through Neuro-Dynamic
Programming are compared with a policy which always accepts a customer
when the required resources are available.
**
Title: Issues In Measuring The Benefits Of Multimodal Interfaces
Authors: 
James L. Flanagan, Rutgers University
Ivan Marsic, Rutgers University
Volume: 1, Page: 163
Abstract:
Multimedia interfaces are rapidly evolving to facilitate human/machine
communication.  Most of the technologies on which they are based are,
as yet, imperfect.  But, the interfaces do begin to allow information
exchange in ways familiar and comfortable to the human--principally
through natural actions in the sensory dimensions of sight, sound and
touch.  Further, as digital networking becomes ubiquitous, the opportunity
grows for collaborative work through conferenced computing. In this
context the machine takes on the role of mediator in human/machine/human
communication--the ideal being to extend the intellectual abilities
of humans through access to distributed information resources and collective
decision making.  The challenge is how to design machine mediation
so that it extends, not impedes, human abilities.  This report describes
evolving work to incorporate multimodal interfaces into a networked
system for collaborative distributed computing.  It also addresses
strategies for quantifying the synergies that may be gained.
**
Title: Multimodal Interfaces for Multimedia Information Agents
Authors: 
Alex Waibel, Carnegie Mellon University
Bernhard Suhm, Carnegie Mellon University
Minh Tu Vo, Carnegie Mellon University
Jie Yang, Carnegie Mellon University
Volume: 1, Page: 167
Abstract:
When humans communicate they take advantage of a rich spectrum of cues.
Some are verbal and acoustic. Some are non-verbal and non-acoustic.
Signal processing technology has devoted much attention to the recognition
of speech, as a single human communication signal. Most other complementary
communication cues, however, remain unexplored and unused in human-computer
interaction. In this paper we show that the addition of non-acoustic
or non-verbal cues can significantly enhance robustness, flexibility,
naturalness and performance of human-computer interaction. We demonstrate
computer agents that use speech, gesture, handwriting, pointing, spelling
jointly for more robust, natural and flexible human-computer interaction
in the various tasks of an information worker: information creation,
access, manipulation or dissemination.
**
Title: Smart Rooms, Desks, and Clothes
Authors: 
Alex Pentland, MIT Media Lab
Volume: 1, Page: 171
Abstract:
We are working to develop smart networked environments that can help
people in their homes, offices, cars, and when walking about. Our research
is aimed at giving rooms, desks, and clothes the perceptual and cognitive
intelligence needed to become active helpers.
**
Title: Human Machine Interaction by Voice and Gesture
Authors: 
Nikil Jayant, Bell Laboratories
Volume: 1, Page: 175
Abstract:
Voice and gesture represent fundamental and universal modalities in
interhuman communication. With recent advances in automatic methods
of speech recognition and synthesis, human-machine interaction by voice
is rapidly becoming a technological and commercial reality. Although
less mature and deployed, gesture recognition by machine is becoming
reliable enough to be considered as a serious supplement to the voice
interface between humans and machines.
**
Title: Audio-Visual Interaction in Multimedia Communication
Authors: 
Tsuhan Chen, AT&T Labs - Research
Ram R. Rao, Georgia Institute of Technology
Volume: 1, Page: 179
Abstract:
To many people, the word "multimedia" simply means the combination
of various forms of information: text, speech, music, images, graphics
and video.  What is often overlooked is the interaction among these
forms.  In this paper, we will present our recent results in exploiting
the audio-visual interaction that is very significant in multimedia
communication.  The applications include lip synchronization, joint
audio-video coding, and person verification.  We will present the enabling
technologies, including audio-to-visual mapping and facial image analysis,
for these applications.  Our results show that the joint processing
of audio and video provides advantages that are not available when
audio and video are studied separately.
**
Title: LIP Motion Modeling and Speech Driven Estimation
Authors: 
F. Lavagetto, University of Genova
S. Lepsoy, University of Genova
C. Braccini, University of Genova
S. Curinga, University of Genova
Volume: 1, Page: 183
Abstract:
Recent advances in joint acoustical/visual analysis for model-based
lip motion synthesis is presented. The 2D lip motion field is modeled
as a linear combination of a low dimensional motion basis computed
through Principal Component Analysis (PCA). The vector of PCA coefficients
is expressed as a function of a limited set of articulatory parameters
which describe the external appearance of the mouth.  The acoustical
processing estimates these articulatory parameters from the direct
analysis of the speech waveform based on a neural processing stage,
i.e. through a bank of Time Delay Neural Networks. The achieved results
have been subjectively evaluated by visualizing the estimated motion
on a wire-frame mouth template presented in synchronization with speech.
The experiments carried out so far deal with single-speaker trained
TDNNs and with single-speaker PCA, but suitable algorithms for generalizing
the techniques are currently under investigation.
**
Title: Voice Source Localization for Automatic Camera Pointing System
in Videoconferencing
Authors: 
Hong Wang, PictureTel
Peter Chu, PictureTel
Volume: 1, Page: 187
Abstract:
This paper describes the voice source localization algorithm used in
the PictureTel automatic camera pointing system (LimeLight-TM, Dynamic
Speech Locating Technology).  The system uses an array of 46cm wide
and 30cm high, which contains 4 microphones, and is mounted on top
of the monitor. The three dimensional position of a sound source is
calculated from the time delays of 4 pairs of microphones. In time
delay estimation, the averaging of signal onsets of each frequency
band is combined with phase correlation to reduce the influence of
noise and reverberation. With this approach, it is possible to provide
reliable three dimensional voice source localization by a small microphone
array.  Post processing based on a priori knowledge is also introduced
to eliminate the influences of reflections from furniture such as tables.
Results of speech source localization under real conference room conditions
will be given. Some system related issues will also be discussed.
**
Title: Video interface for spatiotemporal interactions based on multi-dimensional
video computing
Authors: 
Akihito Akutsu, NTT Human Interface Laboratories
Yoshinobu Tonomura, NTT Human Interface Laboratories
Hiroshi Hamada, NTT Human Interface Laboratories
Volume: 1, Page: 191
Abstract:
Because digital video is becoming increasingly important for the networked
multimedia society, the audio-visual access environment should allow
us to do more than just passively watch.  We propose a new video user
interface concept made possible by multi-dimensional video computing.
Multi-dimensional video computing offers a framework for analyzing
a video, creating new structures, and restyling and visualizing the
video according to the user's demands.  The video interface visualizes
video content and context structure comprehensibly to allow us to access
the spatiotemporal information in videos intuitively. In this paper,
we introduce our research activities toward a video interface based
on the information extracted from the video. New video interfaces called
VideoBrowser, PanoramaVideo, and VideoJigsaw are described.
**
Title: Indexing and Search of Multimodal Information
Authors: 
Alexander G. Hauptmann, Carnegie Mellon University
Howard D. Wactlar, Carnegie Mellon University
Volume: 1, Page: 195
Abstract:
The Informedia Digital Library Project allows full content indexing
and retrieval of text, audio and video material.  The integration of
speech recognition, image processing, natural language processing and
information retrieval overcomes limits in each technology to create
a useful system.  In order to answer the question how good speech recognition
has to be in order to be useful and usable for indexing and retrieving
speech recognizer generated transcripts, some empirical evidence is
presented that illustrates the degradation of information retrieval
at different levels of speech accuracy.  In our experiments, word error
rates up to 25% did not significantly impact information retrieval
and error rates of 50% still provided 85 to 95% of the recall and precision
relative to fully accurate transcripts in the same retrieval system.
**
Title: Acoustic Indexing for Multimedia Retrieval and Browsing
Authors: 
Steve J. Young, Cambridge University Engineering Dept
Jonathan T. Foote, Cambridge University Engineering Dept
Gareth J.F Jones, Cambridge University Engineering Dept
Karen Sparck Jones, Cambridge University Computer Lab
Martin G. Brown, ORL Ltd
Volume: 1, Page: 199
Abstract:
This paper reviews the Video Mail Retrieval (VMR) project at Cambridge
University and ORL.  The VMR project began in September 1993 with the
aim of developing methods for retrieving video documents by scanning
the audio soundtrack for keywords.  The project has shown, both experimentally
and through the construction of a working prototype, that speech recognition
can be combined with information retrieval methods to locate multimedia
documents by content.  The final version of the VMR system uses pre-computed
phone lattices to allow extremely rapid word spotting and audio indexing,
and statistical information retrieval (IR) methods to mitigate the
effects of spotting errors.  The net result is a retrieval system that
is open-vocabulary and speaker-independent, and which can search audio
orders of magnitude faster than real time.
**
Title: Broadcast News Transcription
Authors: 
Francis Kubala, BBN
Hubert Jin, BBN
Long Nguyen, BBN
Richard Schwartz, BBN
Spyros Matsoukas, Northeastern University
Volume: 1, Page: 203
Abstract:
In this paper we describe our recent work on automatic transcription
of radio and television news broadcasts.  This problem is very challenging
for large vocabulary speech recognition because of the frequent and
unpredictable changes that occur in speaker, speaking style, topic,
channel, and background conditions.  Faced with such a problem, there
is a strong tendency to try to carve the input into separable classes
and deal with each one independently. In our early work on this problem,
however, we are finding that the rewards for condition-specific techniques
are disappointingly small.  This is forcing us to look for general,
robust, and adaptive algorithms for dealing with extremely variable
data.  Herein, we describe the BBN BYBLOS recognition system configured
to handle off-line transcription and we characterize the speech contained
in the 1996 DARPA Hub-4 testbed.  On the partitioned development test
set, we achieved a 29.4% overall word error rate.
**
Title: Image/Speech Processing that Adopts an Artistic Approach -Toward
Integration of Art and Technology-
Authors: 
Ryohei Nakatsu, ATR-MIC
Volume: 1, Page: 207
Abstract:
In the areas of image/speech processing, researchers have long dreamed
of producing computer agents that can communicate with people in a
human-like way.  Although the non-verbal aspects of communications,
such as emotions-based communications, play very important roles in
our daily lives, most research so far has concentrated on the verbal
aspects of communications and has neglected the nonverbal aspects.
To achieve human-like agents we have adopted a two-way approach. 1.
To provide agents with nonverbal communications capability, engineers
have started research on emotions recognition and facial expressions
recognition.  2. Artists have begun to design and generate the reactions
and behaviors of agents, to fill the gap between real human behaviors
and those of computer agents.
**
Title: Noise Cancelling for Microphone Arrays
Authors: 
Jens Meyer, Darmstadt University of Technology
Carsten Sydow, SIEMENS AG
Volume: 1, Page: 211
Abstract:
In this paper an application of the noise cancelling method for suppression
of noise of a microphone array system is discussed. First an overview
of the noise cancelling approach is given. This is followed by a description
of the employment of the method in a realized microphone array system.
The limiting factors are described and theoretical limits of the noise
suppression are derived. Experimental results, which are obtained in
a realistic environment, are presented. The results show, that depending
on the recording situation the noise cancelling approach applied to
a microphone array system leads to a significant enhancement of the
signal to noise ratio of the array output signal.
**
Title: A Microphone Array System for Speech Recognition
Authors: 
Kenji Kiyohara, NTT Human Interface Labs.
Yutaka Kaneda, NTT Human Interface Labs.
Satoshi Takahashi, NTT Human Interface Labs.
Hiroaki Nomura, NTT Human Interface Labs.
Junji Kojima, NTT Human Interface Labs.
Volume: 1, Page: 215
Abstract:
This paper proposes a microphone array system which realizes the following
important functions for speech recognition: i) SNR improvement, ii)
flat spectrum response for an arbitrary speaker position, and iii)
speech period detection in noisy speech. This microphone array system
features time delay estimation using pre-whitening signal processing,
delay-and-sum array weighted optimally, and speech period detection
based on the level difference (called MLD) between signals before and
after array processing. Word recognition experiments performed in the
presence of crowd noise demonstrate greater robustness of the proposed
system against noise than the system with conventional directional
microphone and speech period detection method.
**
Title: Strategies for combining Acoustic Echo Cancellers and Adaptive
Beamforming Microphone Arrays
Authors: 
Walter Kellermann, FH Regensburg
Volume: 1, Page: 219
Abstract:
New concepts for efficient combination of acoustic echo cancellation
(AEC) and adaptive beamforming microphone arrays (ABMA) are presented.
By decomposing common beamforming methods into a time-invariant part,
which the AEC can integrate, and a separate time-variant part, the
number of echo cancellers is minimized without rendering the system
identification problem more difficult. Methods for controlling the
interaction of ABMA and AEC are outlined and implementations for typical
microphone array applications are discussed briefly.
**
Title: A Steerable and Variable First-Order Differential Microphone Array
Authors: 
Gary W. Elko, Acoustics Research Department
Anh-Tho Nguyen Pong, Speech Processing Software and Technology Research
Volume: 1, Page: 223
Abstract:
A new first-order differential microphone array with an infinitely
steerable and variable beampattern is described. The microphone consists
of 6 small pressure microphones flush-mounted on the surface of a 3/4"
diameter rigid nylon sphere.  The microphones are located on the surface
at points where included octahedron vertices contact the spherical
surface. By appropriately combining the three Cartesian orthogonal
pairs with simple scalar weightings, a general first-order differential
microphone beam (or beams) can be realized and directed to any angle
in 4(pi) steradian space. A working real-time version has been created
and measured results from this microphone are shown.  This microphone
should be useful for surround sound recording/playback applications
and to virtual reality audio applications.
**
Title: Microphone Array based Speech Recognition with Different Talker-Array
Positions
Authors: 
Maurizio Omologo, ITC-IRST
Marco Matassoni, ITC-IRST
Piergiorgio Svaizer, ITC-IRST
Diego Giuliani, ITC-IRST
Volume: 1, Page: 227
Abstract:
The use of a microphone array for hands-free continuous speech recognition
in noisy and reverberant environment is investigated. An array of eight
omnidirectional microphones was placed at different angles and distances
from the talker.  A time delay compensation module was used to provide
a beamformed signal as input to a Hidden Markov Model (HMM) based recognizer.
A phone HMM adaptation, based on a small amount of phonetically rich
sentences, further improved the recognition rate obtained by applying
only beamforming. These results were confirmed both by experiments
conducted in a noisy and reverberant environment and by simulations.
In the latter case, different conditions were recreated by using the
image method to reproduce synthetic versions of the array microphone
signals.
**
Title: Acoustic Source Location In A Three-Dimensional Space Using Crosspower
Spectrum Phase
Authors: 
Piergiorgio Svaizer, ITC-IRST
Marco Matassoni, ITC-IRST
Maurizio Omologo, ITC-IRST
Volume: 1, Page: 231
Abstract:
A microphone array can be used to locate a dominant acoustic source
in a given environment. This capability is successfully employed to
locate an active talker in teleconferencing or other multi-speaker
applications.  In this work the source location is obtained in two
steps: 1) a Time Difference Of Arrival (TDOA) computation between the
signals of the array; 2) an ``optimal'' source location based on the
interchannel delay estimates and on a geometrical description of the
sensor arrangement. The Crosspower Spectrum Phase technique was used
for TDOA estimation, while a Maximum Likelihood approach was followed
to derive the source coordinates. Source location experiments in a
three-dimensional space were performed by means of an array of 8 microphones.
For this purpose both a loudspeaker and a real talker were used to
collect data in a large noisy and reverberant room.
**
Title: Superdirective Microphone Array for a Set-Top Videoconferencing
System
Authors: 
Peter Chu, PictureTel
Volume: 1, Page: 235
Abstract:
In set-top videoconferencing, the complete videoconferencing system
fits unobtrusively on top of the television. The microphone sound pickup
system is one of the most important functional blocks with constraints
of small size, high performance, and low cost. Persons speaking several
feet away from the system must be picked up satisfactorily while noise
generated internally in the system by the cooling fan and hard drive,
and noise generated externally from air conditioning and nearby computers
must be attenuated. In this paper, a three microphone superdirective
array is described which meets these constraints. An analog highpass
and lowpass filter are used to merge two of the microphone signals
to form a single channel, so that a single stereo A/D converter is
required to process the three microphone signals. The microphone signals
are then linearly combined so as to maximize the signal-to-noise ratio,
resulting in nulls steered toward nearby objectionable noise sources.
**
Title: Simultaneous Echo Cancellation and Car Noise Suppression Employing
a Microphone Array
Authors: 
Matttias Dahl, University of Karlskrona/Ronneby
Ingvar Claesson, University of Karlskrona/Ronneby
Sven Nordebo, University of Karlskrona/Ronneby
Volume: 1, Page: 239
Abstract:
This paper presents a method to simultaneously perform 20~dB acoustic
echo cancellation and 15-20~dB speech enhancement using an adaptive
microphone array combined with spectral subtraction. Primarily intended
for handsfree telephones in automobiles, the microphone array system
simultaneously emphasizes the near-end talker and suppresses the handsfree
loudspeaker and the broadband car noise. The array system is based
on a fast and efficient on-site calibration and can be used in other
situations such as conventional speaker phones.
**
Title: Analytical Evaluation of a Self-calibrating Microphone Array
Authors: 
Sven Nordholm, University of Karlskrona/Ronneby
Ingvar Claesson, University of Karlskrona/Ronneby
Volume: 1, Page: 243
Abstract:
This paper gives an analytical description of an adaptive microphone
array which facilitates a simple built-in calibration to the environment
and instrumentation.  The scheme offers several advantages, such as
a simple calibration procedure and reduced target signal distortion.
The analysis employs noncausal Wiener filters yielding compact and
effective theoretical suppression limits.
**
Title: Microphone Array Response to Speaker Movements
Authors: 
Yves Grenier, ENST
Sofiene Affes, INRS-Telecommunications
Volume: 1, Page: 247
Abstract:
Matched filtering and adaptive beamforming are both necessary for efficient
speech dereverberation and noise reduction by microphone arrays. This
can be achieved by the identification of impulse responses. In this
contribution, we show that adaptive microphone arrays are sensitive
to identification errors of impulse responses, particularly due to
speaker movements. We prove that adjusted matched-filtering and permanent
tracking of impulse responses are also necessary. The proposed microphone
array responds well to these requirements under realistic conditions.
**
Title: A Digital Processing System for Source Location and Sound Capture
by Large Microphone Arrays
Authors: 
Harvey F. Silverman, Brown University
William R. Patterson, Brown University
James L. Flanagan, Rutgers University
Daniel Rabinkin, Rutgers University
Volume: 1, Page: 251
Abstract:
The Huge Microphone Array(HMA) project started in February 1994 to
design, construct, and test a real-time 512-microphone array system
and to develop algorithms for use on it.  Analysis of known algorithms
showed that signal-processing performance of over 6 Gigaflops would
be required; at the same time, there was a need for portability, i.e.,
fitting into a small van.  These tradeoffs and many others have led
to a unique design in both hardware and software.  This paper presents
the design and its justifications.  Performance data for a few important
algorithms relative to usage of processing-capability, response latency,
and difficulty of programming are discussed.
**
Title: 3-D Unitary ESPRIT for Joint 2-D Angle and Carrier Estimation
Authors: 
Martin Haardt, Siemens
Josef A. Nossek, Technical University of Munich
Volume: 1, Page: 255
Abstract:
It is essential for an efficient frequency and time slot allocation
procedure in future mobile communication systems using space division
multiple access (SDMA) to determine the mobiles that are spatially
well separated from one another. Thus, once a mobile desires to initiate
a call, precise knowledge of the 2-D arrival angles of its dominant
wavefronts is required.  In this application, 3-D Unitary ESPRIT for
joint 2-D angle and carrier estimation offers an efficient way to handle
such mobile access requests since it provides efficient high-resolution
measurements of the spatial characteristics of the wireless channel,
even if only a small number of antennas is available at the base station.
Automatic pairing of the 3-D estimates is achieved via a new simultaneous
Schur decomposition (SSD) of three real-valued, non-symmetric matrices.
In general, the SSD enables an R-dimensional extension of Unitary ESPRIT
(R greater-or-equal-to 3) to estimate several undamped R-dimensional
modes or frequencies along with their correct pairing in multidimensional
harmonic retrieval problems.  Here, we present a Jacobi-type method
to calculate the SSD.  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.
**
Title: Quality enhancement of coded and corrupted speeches in GSM mobile
systems using residual redundancy
Authors: 
Thomas Hindelang, Technical University of Munich
Wen Xu, Technical University of Munich
Christian Erben, Technical University of Munich
Volume: 1, Page: 259
Abstract:
There is often residual redundancy remaining in coded speech data,
even if a powerful speech codec (e.g. the full rate coder used in GSM
mobile communications) is employed. By using such redundancy together
with the information provided by the channel decoder, such as soft
output (L-value), the number of channel bits inverted by the decoder,
or a cyclic redundancy check, the bit error rate can be further reduced
and a more graceful degradation of speech quality can be achieved,
especially under bad channel conditions. In this paper, we report on
the study with regard to this aspect for GSM full rate speech transmission
and error concealment. The algorithms developed can be easily implemented
with a currently available DSP designed for GSM mobile phones.
**
Title: Pilot Assisted Coherent DS-CDMA Reverse-Link Communications with
Optimal Robust Channel Estimation
Authors: 
Fuyun Ling, Motorola
Volume: 1, Page: 263
Abstract:
Optimal pilot assisted estimation of communication channels is considered
for coherent cellular and PCS CDMA reverse link communications. Both
pilot symbol and pilot channel based schemes are described and the
optimal estimators for these two schemes are analyzed. Relative mean
square estimation error (RMSEE) and optimal power allocation between
data and pilot signals are derived based on the analysis. Finally,
simulation results are given to show the reverse link performance can
be significantly improved by using the pilot assisted coherent communication
instead of non-coherent schemes for CDMA reverse link.
**
Title: A new Frequency Estimator applied to Burst Transmission
Authors: 
Christian Bergogne, Telecom Paris, Alcatel Telspace
Michel Bousquet, ENSAE
Philippe Sehier, Alcatel Telspace
Volume: 1, Page: 267
Abstract:
In TDMA communications systems using all feedforward sychronization
techniques, the quality of data decoding strictly depends on the estimation
accuracy of the synchronization parameters (timing, carrier phase/frequency
and preamble detection) extracted from the received signal. The frequency
offset estimation is the most critical point. Indeed, an inaccurate
frequency estimation can cause cycle slips and then errors during decoding.
In this paper, we propose a new frequency estimator, analytically derived
from the Maximum Likelihood principle and optimized thanks to variance
simulations. Its performance is compared to the Cramer Rao Bound.
**
Title: Unified Specification of Control and Data Flow
Authors: 
Thorsten Groetker, ISS, RWTH Aachen
Rainer Schoenen, ISS, RWTH Aachen
Heinrich Meyr, ISS, RWTH Aachen
Volume: 1, Page: 271
Abstract:
Many signal processing systems use event driven mechanisms - typically
based on finite state machines (FSMs) - to control the operation of
computationally intensive (data flow) parts. The state machines in
turn are often fueled by external inputs as well as by feedback from
the signal processing portions of the system.  Packet-based transmission
systems are a good example for such a close interaction between data
and control flow.  For an efficient design flow it is of crucial importance
to be able to model and analyze the complete functionality of the system
within one single design environment.  Therefore, we developed a computational
model that integrates the specification of control and data flow by
combining the notion of data flow graphs with event driven process
activation.
**
Title: Reconfigurable Processing: The Solution to Low-Power Programmable
DSP
Authors: 
Jan M. Rabaey, University of California at Berkeley
Volume: 1, Page: 275
Abstract:
One of the most compelling issues in the design of wireless commu-
nication components is to keep power dissipation between bounds. While
low-power solutions are readily achieved in an application- specific
approach, doing so in a programmable environment is a sub- stantially
harder problem. This paper presents an approach to low- power programmable
DSP that is based on the dynamic reconfigura- tion of hardware modules.
This technique has shown to yield at least an order of magnitude of
power reduction compared to traditional instruction-based engines for
problems in the area of wireless com- munication.
**
Title: DSP Cores for Moblile Communications: Where are we going ?
Authors: 
Gerhard P. Fettweis, Technical University of Dresden
Volume: 1, Page: 279
Abstract:
Digital signal processors (DSPs) have become a key component for the
design of communications ICs. Application customization leads to key
market advantages but also to enormous problems of having too many
different DSPs and their software development tools. First, by analysis
of the problem open issues are pointed out. Then, a possible solution
named CATS is presented, which allows for customization without the
generation of too much heterogeneity in hardware and tools.
**
Title: DSPs in Mobile Communication in the United States
Authors: 
Sanjay Kasturia, Bell Labs, Lucent Technologies
Colin Warwick, Bell Labs, Lucent Technologies
Volume: 1, Page: 283
Abstract:
The mobile communication industry in the United States is undergoing
major changes. Auctioning of additional spectrum will lead to more
service providers and will significantly increase competition. Service
providers are likely to customize the services they offer to differentiate
themselves from others. We will discuss possible technologies for differentiation
of services and the implications of these on the requirements for embedded
DSPs. In the US, supporting the customization in the absence of a single
industry wide standard, and the high likelihood of at least three widely
used air interfaces will significantly challenge the ability of the
industry to serve the phone needs of all service providers.  The need
for customization in the context of multiple standards, will create
strong pressure to significantly improve the code development environment
for DSPs. This also implies evolution to architectures that are more
friendly to developers.
**
Title: FRIDGE: An Interactive Code Generation Environment for HW/SW CoDesign
Authors: 
Markus Willems, ISS, RWTH Aachen
Volker Bursgens, ISS, RWTH Aachen
Thorsten Grotker, ISS, RWTH Aachen
Heinrich Meyr, ISS, RWTH Aachen
Volume: 1, Page: 287
Abstract:
Digital mobile systems are sensitive to power consumption, chip size
and costs. Therefore they are realized using fixed-point architectures,
either dedicated HW or fixed-point processors.  On the other hand,
system design starts from a floating-point description. These requirements
have been the motivation for FRIDGE, a design environment for the specification,
evaluation and implementation of fixed-point systems. FRIDGE offers
a seamless design flow from a floating-point description to a fixed-point
implementation. Within this paper we focus on the FRIDGE-concept of
an interactive, automated transformation of floating-point programs
written in ANSI-C into fixed-point specifications, based on an interpolative
approach. Since HW and SW implementations of the same functionality
in general require different fixed-point specifications, the design
time reductions that can be achieved by using FRIDGE make it a key
component for an efficient HW/SW-CoDesign.
**
Title: Staying Ahead of the Game In Silicon for Digital Mobile Communications
Authors: 
Ravi Subramanian, Synopsys Inc.
Marc Barberis, Synopsys Inc.
Herbert Dawid, Synopsys GmbH
Volume: 1, Page: 291
Abstract:
While the mobile communication electronics industry's appetite grows
for ever more functions and ever higher levels of integration, the
complexity of these large designs is creating a discontinuity in the
method by which these systems are designed. In this paper, we will
take a close look at what is causing the design discontinuity, and
how new design technologies are being used to design advanced digital
communications systems for portable and wireless communication applications.
We will examine how system-level design tools closely tied to silicon
design implementation and verification technologies are enabling the
creation of digital communications ICs in record time. We take several
examples of commercially available silicon solutions designed using
these methodologies- a G.721 ADPCM speech codec for cordless telephony
and a complete variable-rate digital-video broadcast receiver for the
DVB-S broadcast standard.
**
Title: Approximation of Optimal Step Size Control for Acoustic Echo Cancellation
Authors: 
Christiane Antweiler, RWTH Aachen
Jorn Grunwald, RWTH Aachen
Holger Quack, RWTH Aachen
Volume: 1, Page: 295
Abstract:
One of the most widely used gradient-based adaptation algorithms is
the so called normalized least mean square (NLMS) algorithm. The rate
of convergence, misadjustment and noise insensitivity of the NLMS-type
algorithm depend on the proper choice of the step size parameter, which
controls the weighting applied to each coefficient update.  Different
step size methods have been proposed to improve the convergence of
NLMS-type filters, while preserving the steady-state performance. The
step size methods considered here use either a step size parameter
which varies with time or a separate, tap-individual step size for
each filter tap. The derivation of the respective step size methods
is based on different optimization criteria.  In this paper a step
size parameter is proposed satisfying a combined optimization criterion
leading to a time variant and individual step size parameter. The realization
aspects of the new concept are discussed for an acoustic echo control
application as an example.
**
Title: Subband stereo echo canceller using the projection algorithm with
fast convergence to the true echo path
Authors: 
Shoji Makino, NTT Human Interface Labs
Klaus Strauss, NTT Human Interface Labs
Suehiro Shimauchi, NTT Human Interface Labs
Yoichi Haneda, NTT Human Interface Labs
Akira Nakagawa, NTT Human Interface Labs
Volume: 1, Page: 299
Abstract:
This paper proposes a new subband stereo echo canceller that converges
to the true echo path impulse response much faster than conventional
stereo echo cancellers. Since signals are bandlimited and downsampled
in the subband structure, the time interval between the subband signals
become longer, so the variation of the crosscorrelation between the
stereo input signals becomes large. Consequently, convergence to the
true solution is improved. Furthermore, the projection algorithm, or
affine projection algorithm, is applied to further speed up the convergence.
Computer simulations using stereo signals recorded in a conference
room demonstrate that this method significantly improves convergence
speed and almost solves the problem of stereo echo cancellation with
low computational load.
**
Title: A Better Understanding and an Improved Solution to the Problems
of Stereophonic Acoustic Echo Cancellation
Authors: 
Jacob Benesty, Bell Labs
Dennis R. Morgan, Bell Labs
M. Mohan Sondhi, Bell Labs
Volume: 1, Page: 303
Abstract:
Teleconferencing systems employ acoustic echo cancelers (AECs) to reduce
echos that result from coupling between the loudspeaker and microphone.
To enhance the sound realism, two-channel audio is necessary. However,
in this case (stereophonic sound) the acoustic echo cancellation problem
is more difficult to solve because of the necessity to uniquely identify
two acoustic paths. In this paper, we explain these problems in detail
and give an interesting solution which is much better than previously
known solutions. The basic idea is to introduce a small nonlinearity
into each channel that has the effect of reducing the interchannel
coherence while not being noticeable for speech due to self masking.
**
Title: Comparison of three post-filtering algorithms for residual acoustic
echo reduction
Authors: 
Valerie Turbin, CNET
Andre Gilloire, CNET
Pascal Scalart, CNET
Volume: 1, Page: 307
Abstract:
We consider an acoustic echo control system composed of a short conventional
acoustic echo canceller combined with a post-filter in a teleconference
context. The post-filter is implemented in an open-loop structure in
the frequency domain, which provides good adaptive performance and
flexibility for the choice of the post-filter length. Three post-filtering
algorithms are compared in terms of residual echo attenuation and near-end
speech distortion. The effect of the post-filter length is also examined.
Our study confirms that the post-filtering approach provides high residual
echo attenuation. Moreover, it appears that the distortion of the near-end
speech can be controlled by choosing appropriately the post-filter
length.
**
Title: Audio Coding Using Sinusoidal Excitation Representation
Authors: 
Wen-Whei Chang, National Chiao-Tung University
De-Yu Wang, National Chiao-Tung University
Li-Wei Wang, National Chiao-Tung University
Volume: 1, Page: 311
Abstract:
Most LPC-based audio coders employ simplistic noise-shaping operations
to perform psychoacoustic control of quantization noise. In this paper,
we report on new approaches to exploiting perceptual masking in the
design of adaptive quantization of LPC excitation parameters. Due to
its localized spectral sensitivity, sinusoidal excitation representation
is preferred to spectrally flat signals for use in excitation modeling.
Simulation results indicate that the proposed multisinusoid excited
coder can deliver high quality audio reproduction at the rate of 72
kb/s.
**
Title: Optimum Bit Allocation and Decomposition for High Quality Audio
Coding
Authors: 
Xiang Wei, University of Central Lancashire
Martyn J. Shaw, University of Central Lancashire
Martin R. Varley, University of Central Lancashire
Volume: 1, Page: 315
Abstract:
Current audio compression schemes are capable of reducing the per channel
bit rate of high quality audio signals from 16 bits per sample to around
2-4 bits per sample.  In these schemes, knowledge of psychoacoustics
is utilised and a uniform or nonuniform frequency decomposition method
is used.  In this paper we derive the optimum bit allocation to achieve
the highest perceptual quality under a fixed bit rate, for an arbitrarily
decomposed, critically sampled, filter bank.  The resultant optimum
bit allocation gives rise to a shaped reconstruction noise floor approximately
parallel to the masking threshold level.  Perceptual coding gain is
defined and should be maximized for an optimum decomposition performed
by the filter bank.  Optimum band splitting is discussed and it is
pointed out that decomposition in the manner of critical band splitting
does not lead to optimal performance.
**
Title: The D5 Lattice Quantization For A 64 KBit/S Low-Delay Subband
Audio Coder With A 15 KHz Bandwidth
Authors: 
Karine Hay, ENST-Br, Dept. SC.
S. Saoudi, ENST-Br, Dept. SC.
L. Mainard, CCETT, Servive RCS/SDA
Volume: 1, Page: 319
Abstract:
A new method for coding generic audio signals at 64 kbit/s in the 20-15000
Hz bandwidth with a low delay is presented. It combines subband coding,
Low Delay CELP algorithm and cascaded filterbanks. Our earlier works
shown that, when using an equal bit rate on each subband, the resulting
audio quality was not appropriate. We propose here a new technique
based on lattice quantization to avoid the search complexity of the
statistical vector quantization. It allows an adaptive bit rate allocation
in each subband. Experimental results assessing the validity of the
proposed method are also presented.
**
Title: An Experimental Audio Codec Based on Warped Linear Prediction
of Complex Valued Signals
Authors: 
Aki Harma, Helsinki University of Technology
Unto K. Laine, Helsinki University of Technology
Matti Karjalainen, Helsinki University of Technology
Volume: 1, Page: 323
Abstract:
Bark-scale warped linear prediction [WLP] is a very potential core
for a monophonic perceptual audio codec.  In the current paper the
WLP scheme is extended for processing complex valued signals (CWLP).
Three different methods of converting a stereo signal to one complex
valued signal are introduced.  The philosophy behind the coding scheme
is to integrate some aspects of modern wideband audio coding (e.g.
perceptuality and stereo signal processing) into one computational
element in order to find a more holistic and economic way of processing.
**
Title: High Quality Low Complexity Scalable Wavelet Audio Coding
Authors: 
William Kurt Dobson, U.S. Robotics
Jiankan Jack Yang, U.S. Robotics
Kevin J. Smart, U.S. Robotics
Feng Kathy Guo, U.S. Robotics
Volume: 1, Page: 327
Abstract:
This paper presents an audio coder for real-time multimedia applications.
To achieve high quality at low bit rate, the audio coder uses a wavelet
packet decomposition to transform the audio data into the wavelet domain,
and a psychoacoustic model is used to minimize quantization noise.
The wavelet packet decomposition tree structures were chosen in a way
to closely mimic the critical bands in a psychoacoustic model. Instead
of determining the masking thresholds in the Fourier domain, the wavelet
coefficients are used to drive the psychoacoustic model directly. Most
of the standard industrial sampling frequencies are supported by this
coder. An efficient bit rate control scheme was designed such that
the audio coder operates at virtually any desired bit rate level. The
audio coder achieves near perceptually lossless quality at or below
80 kb/s for most audio sources. Real-time encoding/decoding is possible
by using only a fraction of a Pentium or faster CPU.
**
Title: An Efficient Tonal Component Coding Algorithm For MPEG-2 Audio
NBC
Authors: 
Yuichiro Takamizawa, NEC Corporation
Masahiro Iwadare, NEC Corporation
Akihiko Sugiyama, NEC Corporation
Volume: 1, Page: 331
Abstract:
This paper proposes a tonal component coding algorithm for a codec
that employs a transform followed by Huffman coding, such as MPEG-2
Audio NBC (Non-Backward Compatible).  After the input audio signal
is mapped onto a frequency domain, the proposed algorithm withdraws
local maximum components that degrade coding efficiency. By this withdrawal,
the flatness of the spectrum increases and the efficiency in Huffman
coding is improved. The withdrawn components are encoded separately
as side information. When the frequency resolution of the time/frequency
mapping is high, this algorithm works more effectively since local
maximum samples appear more frequently with such a mapping. Simulation
results show that this algorithm achieves as much as 11% bit reduction
per frame and improves the coding efficiency in 41% of all the audio
frames.
**
Title: Spectral Amplitude Warping (SAW) for Noise Spectrum Shaping in
Audio Coding
Authors: 
Roch Lefebvre, University of Sherbrooke
Claude Laflamme, University of Sherbrooke
Volume: 1, Page: 335
Abstract:
In this paper, we present a new approach to shape the coding noise
in speech and audio coders.  The approach, called Spectral Amplitude
Warping (SAW), consists essentially of a pre- and post-processing which
apply a non-linear transformation to the signal short-term spectrum
prior to, and after, encoding.  Since it is possible to view SAW as
a separate entity from the coder, the noise shaping capability of an
existing coder can be improved without modifying the coder itself.
Using SAW as a pre- and post-process to the G.722 wideband speech coding
standard, it was found in an informal listening test that the quality
of the 64 kb/s operating mode can be achieved at only 48 kb/s. The
price to be paid is an additional delay.
**
Title: A fast noise-scaling algorithm for uniform quantization in audio
coding schemes
Authors: 
Carlos A. Serantes, Universidad de Vigo
Antonio S. Pena, Universidad de Vigo
Nuria Gonzalez-Prelcic, Universidad de Vigo
Volume: 1, Page: 339
Abstract:
A new bit assignment algorithm is presented. Its goals are the simultaneous
assignment on all subbands in a few steps of an iterative calculus,
the use of memory to achieve a better speed of convergence and the
consideration of a deformable error curve. The basis of the algorithm
is discussed and also other considerations that are likely to arise
in practice. Finally, an example of performance is given.
**
Title: Pyramid Vector Coding for high quality audio compression
Authors: 
Daniele Cadel, Cefriel
Giorgio Parladori, Alcatel Telecom
Volume: 1, Page: 343
Abstract:
Target of this work is the high quality audio coding at low bit rate.
It will be shown how the Pyramid Vector Coding (PVC) can conveniently
replace the classical Huffman Coding technique in audio compression
systems, giving also an advantage in the bit allocation procedure.
The compression performances can be further improved by fixing an upper
limit value of the vector components.
**
Title: Subband Audio Coding with Synthesis Filters Minimizing a Perceptual
Criterion
Authors: 
Karine Gosse, ENST Paris
Francois Moreau de Saint-Martin, CCETT
Xavier Durot, CCETT
Pierre Duhamel, ENST Paris
Jean-Bernard Rault, CCETT
Volume: 1, Page: 347
Abstract:
The design of filter banks for source coding purposes classically relies
on the perfect reconstruction (PR) property. However, several recent
studies have shown that taking the quantization noise into account
in the design could yield noticeable reduction of the mean square reconstruction
error. The purpose of this study is to show that perceptual improvement
can also be obtained in the particular audio coding context by relaxing
the PR constraint. In this context, the mean square error is not relevant
any more, and we define a new perceptual distortion criterion, making
use of a simplified ear model, the MPE (Mean Perceptual Error). Then,
synthesis filters are optimized so as to minimize this MPE. Finally,
this MMPE (Minimum MPE) filter bank is included in an audio coding
scheme. Compared to the corresponding PR filter bank-based scheme by
the means of POM (Perceptual Objective Measure), they show an improved
audio quality.
**
Title: New Results in Low Bitrate Audio Coding Using a Combined Harmonic-Wavelet
Representation
Authors: 
Simon Boland, Queensland University of Technology
Mohamed Deriche, Queensland University of Technology
Volume: 1, Page: 351
Abstract:
In this paper, we propose a new combined harmonic-wavelet representation
for audio where a harmonic analysis-synthesis scheme is used, first,
to approximate each audio frame as a sum of several sinusoids.  Then,
the difference between the original signal and the reconstructed harmonic
signal is analyzed using a wavelet filtering scheme.  After each step
(harmonic analysis & wavelet filtering), parameters are quantized and
encoded.  Compared to previously proposed methods, our audio coder
uses different harmonic analysis-synthesis and wavelet filtering schemes.
We use the Total Least Squares (TLS)-Prony algorithm for the harmonic
analysis-scheme, and an M-band wavelet transform for analyzing the
residual.  Altogether, our proposed coder is capable of delivering
excellent audio signal quality at encoder bitrates of 60-70 kb/s.
**
Title: Adaptive Inverse Control of Weakly Nonlinear Systems
Authors: 
Wolfgang J. Klippel, Dresden
Volume: 1, Page: 355
Abstract:
A weak nonlinear plant can be linearized and will track an input signal
if the plant is preceded by a nonlinear controller which approximates
the inverse of the plant's transfer function. Present techniques for
adjusting the controller adaptively to the plant require an additional
nonlinear adaptive filter to perform a separate system identification.
Straightforward update algorithms can not directly update the filter
parameters in the controller because the transfer function of the plant
might cause instabilities in the adaptive process. This problem is
overcome by performing additional linear filtering to the nonlinear
state vector and/or error signal. Novel filtered-A and filtered-E modifications
of the stochastic gradient based methods are presented which are capable
to update generic as well as special block-oriented nonlinear filter
architectures.
**
Title: Broadband Beamforming with Adaptive Postfiltering for Speech Acquisition
in Noisy Environments
Authors: 
Sven Fischer, Ericsson Eurolab
Karl-Dirk Kammeyer, University of Bremen
Volume: 1, Page: 359
Abstract:
In this paper the implementation of a broadband beamformer which is
built up by several harmonically nested subarrays for each octave band
combined with optimal postfiltering is described. This method has the
advantage of providing large sensor distances for the postfilter estimation
by simultaneously controlling the directivity of the array. The selection
of an optimal postfilter is discussed in detail and its estimation
based on a Nuttall/Carter method for spectrum estimation is described.
The resulting noise reduction system yields improved performance in
diffuse noise fields and no distortions in the case of coherent direct
path noise. Furthermore, the system is robust to steering misadjustment.
**
Title: Near-field Beamforming for Microphone Arrays
Authors: 
James G. Ryan, National Research Council
Rafik A. Goubran, Carleton University
Volume: 1, Page: 363
Abstract:
This paper describes the application of array optimization techniques
to improving the near-field response of an arbitrary microphone array.
The optimization exploits the differences in wavefront curvature between
near-field and far-field sound sources and is suitable for reverberation
reduction in small rooms. The optimum near-field beamformer provides
increased array gain over that obtained from a uniformly weighted delay-and-sum
beamformer.
**
Title: A Robust Adaptive Microphone Array with Improved Spatial Selectivity
and Its Evaluation in an Echoic Environment
Authors: 
Osamu Hoshuyama, NEC
Akihiko Sugiyama, NEC
Akihiro Hirano, NEC
Volume: 1, Page: 367
Abstract:
This paper presents a new robust adaptive microphone array (AMA) and
its evaluation in an echoic environment.  The proposed AMA is a generalized
sidelobe canceller equipped with a variable blocking matrix using coefficient-constrained
adaptive filters, and a multiple-input canceller using norm-constrained
adaptive filters (NCAFs).  Because the NCAFs have selective nonlinearity
in the relationship between coefficient norm and coefficient error,
the proposed AMA has better spatial selectivity than the conventional
AMA. Evaluation with real acoustic data captured in a room of 0.3-second
reverberation time shows that the noise is suppressed by 19 dB. In
subjective evaluation, the proposed AMA obtains 3.8 on a 5-point mean-opinion-score
scale.
**
Title: Tracking Multiple Talkers using Microphone-Array Measurements
Authors: 
Douglas E. Sturim, Brown University
Harvey F. Silverman, Brown University
Michael S. Brandstein, Harvard University
Volume: 1, Page: 371
Abstract:
A method for tracking the positional estimates of multiple talkers
in the operating region of an acoustic microphone array is presented.
Initial talker location estimates are provided by a time-delay-based
localization algorithm.  These raw estimates are spatially smoothed
by a Kalman filter derived from a set of potential source motion models.
Data association techniques based on the estimate clusterings and source
trajectories are incorporated to match location observations with individual
talkers.  Experimental results are presented for array recorded data
using multiple talkers in a variety of scenarios.
**
Title: A Robust Method for Speech Signal Time-Delay Estimation in Reverberant
Rooms
Authors: 
Michael S. Brandstein, Harvard University
Harvey F. Silverman, Brown University
Volume: 1, Page: 375
Abstract:
Conventional time-delay estimators exhibit dramatic performance degradations
in the presence of multipath signals.  This limits their application
in reverberant enclosures, particularly when the signal of interest
is speech and it may not possible to estimate and compensate for channel
effects prior to time-delay estimation.  This paper details an alternative
approach which reformulates the problem as a linear regression of phase
data and then estimates the time-delay through minimization of a robust
statistical error measure.  The technique is shown to be less susceptible
to room reverberation effects.  Simulations are performed across a
range of source placements and room conditions to illustrate the utility
of the proposed time-delay estimation method relative to conventional
methods.
**
Title: A Model-Based Approach to Active Noise Cancellation Using Loudspeaker
Array
Authors: 
Jie Gu, HK University of Science & Tech.
Sze Fong Yau, HK University of Science & Tech.
Volume: 1, Page: 379
Abstract:
This paper presents a new model-based adaptive noise cancellation system
using loudspeaker array and error sensor array which can be used to
reduce the noise in a specific three-dimensional region. First, open
loop system transfer functions are designed using a theoretical propagation
model. The transfer functions thus found are regarded as the nominal
values for the complete system. Second, to compensate for deviations
from the theoretical model, the transfer functions are adapted using
error measures from error sensor array by LMS algorithm. Computer simulation
results shows that our approach is effective for noise reduction in
3-D space. Experiments using real-time active noise control hardware
also confirms the performance of the system.
**
Title: Reverberant Sound Field Analysis using a Microphone Array
Authors: 
Wolfgang Tager, CNET
Yannick Mahieux, CNET
Volume: 1, Page: 383
Abstract:
The use of microphone arrays for sound pickup in reverberant environments
has been proposed by many authors. The observation on the M microphones
can be decomposed into a spatially coherent and an incoherent part.
The first one is due to perfect (plane or spherical) sound waves caused
by the direct path and specular reflections, whereas the latter is
caused by diffusion, diffraction, non-perfect reflections, electrical
and quantization noise. In this paper we firstly present a deflation
method to detect and localize spatially coherent waves from the measured
impulse responses. In a second step the filters which model the source
directivity and the reflecting materials are estimated. The model takes
into account nearfield delay, range attenuation, microphone and source
directivity as well as non trivial reflections.
**
Title: Minimisation of the Maximum Error Signal in Active Control
Authors: 
Alberto Gonzalez, UPV, Valencia
Antonio Albiol, UPV, Valencia
Stephen J. Elliott, ISVR, Southampton
Volume: 1, Page: 387
Abstract:
This paper deals with Multiple Input Multiple Output systems for active
control of acoustic signals. These systems are used when the acoustic
field is complex and therefore a number of sensors are necessary to
estimate the sound field and a number of sources to create the cancelling
field. A steepest descent iterative algorithm is applied to minimise
the p-norm of a vector composed by the output signals of a microphone
array. The existing algorithms deal with the 2-norm of this vector.
This paper describes a general framework that covers the existing systems
and then it focuses on the (infinity)-norm minimisation algorithm.
The minimax algorithm based on the (infinity)-norm minimises the output
signal which has the greatest power. It is shown by means of simulations
using measured data from a real room that the minimax algorithm leads
to a more uniform final noise field than the existing algorithms.
**
Title: Subband Active Noise Control Algorithm Based on a Delayless Subband
Adaptive Filter Architecture
Authors: 
Jeong-Hyeon Yun, Yonsei University
Dae-Hee Youn, Yonsei University
Young-Cheol Park, Samsung Electronic
Volume: 1, Page: 391
Abstract:
In this paper, a new active noise control algorithm based on a delayless
subband adaptive filter architecture is presented. Also, an on-line
system identification method implemented in the subband structure is
suggested. To implement the filtered-x LMS algorithm in the subband
structure, the secondary path transfer function is decomposed into
sets of subband functions. The two filter on-line modeling algorithm
is then applied to each subband to estimate the secondary-path transfer
function in a decomposed form. In this manner, the computational load
for the on-line system identification is reduced by a factor 3 compared
with the wideband approach. Simulation results are presented to show
the efficiency of the new ANC algorithm and the performance of the
on-line system identification scheme.
**
Title: Nonlinear Active Noise Control in a Linear Duct
Authors: 
Paul Strauch, University of Edinburgh
Bernard Mulgrew, University of Edinburgh
Volume: 1, Page: 395
Abstract:
The problem in active noise control in a linear duct is examined. Essentially,
a nonlinear inverse to a nonminimum phase actuator is proposed. The
nonlinear inverse exploits the non-Gaussian nature of some chaotic
and stochastic noise sources. The architecture of the controller is
derived using Bayesian estimation theory and is shown to be a combination
of a linear adaptive network and a radial basis function (RBF) or Volterra
series (VS) network. Because of the nonlinear nature of the controller,
the filtered-x least means square (LMS) architecture cannot be used.
Hence a modified active noise controller is proposed. Simulation results
demonstrate the improvements in performance achievable with the combined
linear and nonlinear controller.
**
Title: Fast Exact Filtered-X LMS and LMS Algorithms for Multichannel
Active Noise Control
Authors: 
Scott C. Douglas, University of Utah
Volume: 1, Page: 399
Abstract:
In some situations where active noise control could be used, the well-known
multichannel version of the filtered-X LMS adaptive filter is too computationally-complex
to implement. In this paper, we develop a fast, exact implementation
of this multichannel system whose complexity is approximately O(2L)
per filter channel, where L is the FIR filter length. In addition,
we provide a computationally-efficient method for effectively removing
the delays of the secondary paths within the coefficient updates, thus
yielding a fast implementation of the LMS adaptive algorithm for multichannel
active noise control.  Examples illustrate both the equivalence of
the algorithms to their original counterparts and the computational
gains provided by the new algorithms.
**
Title: A Novel Frequency Domain Filtered-X LMS Algorithm for Active Noise
Reduction
Authors: 
Toshifumi Kosakat, Tokyo National College of Technology
Stephen J. Elliott, University of Southampton
Christopher C. Boucher, University of Southampton
Volume: 1, Page: 403
Abstract:
A Frequency Domain implementation of the LMS Algorithm has significant
advantages. In broadband applications it is important to use the correct
window function before Fourier transformation to obtain an unbiased
estimation of the required cross correlation function and to eliminate
wrap-around effects. In the Frequency Domain Filtered-X LMS Algorithm
described in this paper, the control filter is updated in the frequency
domain as a background task, while control filtering is performed in
time domain, to minimize processing delays. The frequency domain algorithm
showed better performance than the conventional time domain algorithm
in simulations of single channel active control systems. The algorithm
is also able to improve the convergence of multiple channel systems
by compensating for the coupling between the control channels.
**
Title: Practical Supergain Head Sized Arrays
Authors: 
Dorra Masmoudi, University of Bordeaux
Dominique Dallet, University of Bordeaux
Jean Paul Dom, University of Bordeaux
Volume: 1, Page: 407
Abstract:
This paper carried out a new design of head sized sensor arrays with
a simple delay-and-sum beamforming which provides useful amounts of
directivity index with sufficient robustness to errors.  A frequency-independant
sidelobe reduction is proposed to achieve optimal frequency characteristics.
In order to obtain this control, a principle of combining multiple
level of array structures is established. Results are presented for
spherically isotropic noise. It is found that good performance can
be obtained for a head sized array by combining multiple level structures
with simple delay and sum beamformer.
**
Title: A Multichannel Compression Strategy for a Digital Hearing Aid
Authors: 
Todd Schneider, Unitron
Robert Brennan, Unitron
Volume: 1, Page: 411
Abstract:
Multi-channel compression schemes are a practical method of mapping
the wide dynamic range of speech signals into the reduced dynamic range
of hearing impaired listeners. These systems address two of the shortcomings
of single-channel compression systems: (1) the reduction of gain as
a result of narrow-band non-speech stimuli and (2) the reduction of
gain that often occurs when high-frequency speech components are followed
by intense low-frequency speech components. They also provide frequency-dependent
compression ratios that are needed by many newer supra-threshold fitting
strategies (e.g., DSL I/O). This paper presents a multichannel compression
scheme that employs an oversampled, polyphase DFT filterbank. In each
compressor channel, the gain is controlled by an adjustable combination
of a overall, dual time-constant input signal level and the individual
channel signal level that is measured with a short time-constant RMS
detector. Informal listening tests have demonstrated that the design
has very good audio quality and performs well in real-world listening
situations. The design is suited for low-power, real-time operation.
**
Title: Multi-Microphone Sub-band Adaptive Signal Processing for Improvement
of Hearing Aid Performance: Preliminary Test Results using Normal
Hearing Volunteers
Authors: 
Paul Shields, University of Paisley
Douglas R. Campbell, University of Paisley
Volume: 1, Page: 415
Abstract:
A system for the binaural pre-processing of speech signals for input
to a standard linear hearing aid has been proposed.  The work is based
on that of Toner & Campbell which applied the Least Mean Squares (LMS)
algorithm in sub-bands to speech signals from various acoustic environments
and signal to noise ratios (SNR). The method attempts to take advantage
of the multiple inputs to perform noise cancellation. The use of sub-bands
enables a diverse processing mechanism to be employed, where the wide-band
signal is split into smaller sub-bands, which can subsequently be processed
according to their signal characteristics. The results of a series
of intelligibility tests are presented from experiments in which acoustic
speech and noise data, generated in a simulated room was tested on
normal hearing volunteers.
**
Title: Environmental noise reduction based on speech/non-speech identification
for hearing aids
Authors: 
Kenzo Itoh, NTT HI Labs.
Masahide Mizushima, NTT HI Labs.
Volume: 1, Page: 419
Abstract:
We proposed a vary practical and useful noise reduction system that
has wide application for hearing impaired persons, such as a sound-gathering
system at a lecture hall or conference room. The system uses two basic
technologies, a speech/non-speech identification process and a new
noise reduction process. A speech/non-speech identification process
uses four characteristics of the time and frequency domains of the
input signal. In the noise reduction process, frequency weighting function
is used for basic spectral subtraction and loss control algorithm.
Various kinds of environmental noise were reduced by this system, which
showed excellent performance. Noise is further reduced by using a multi-microphone
system as an acoustic noise suppressor. The results of intelligibility
tests using persons with hearing loss show excellent noise reduction.
**
Title: Blind Separation of Multiple Speakers in a Multipath Environment
Authors: 
Russell Lambert, TRW
Anthony Bell, Salk Institute
Volume: 1, Page: 423
Abstract:
We relate information theoretic blind learning methods (infomax) and
Bussgang blind equalization methods.  The multipath extension of blind
source separation methods can be seen in the frequency domain using
FIR matrix algebra (matrices of finite impulse response filters). Three
forms of Bussgang algorithms are given. The blind serial update method
of Cardoso and Laheld is related to the infomax objective of Bell and
Sejnowski. The application emphasis is on speech separation. We demonstrate
the robustness and power of the new techniques by blindly separating
speech signals recorded in a multipath environment.
**
Title: A Single Chip 1,200 Sinusoid Real-Time Generator for Additive
Synthesis of Musical Signals
Authors: 
Fernando De Bernardinis, Dip. Ing. Informazione, Univ. Pisa
Roberto Roncella, Dip. Ing. Informazione, Univ. Pisa
Roberto Saletti, Dip. Ing. Informazione, Univ. Pisa
Pierangelo Terreni, Dip. Ing. Informazione, Univ. Pisa
Graziano Bertini, IEI-CNR, Pisa
Volume: 1, Page: 427
Abstract:
This paper presents a new hardware implementation of additive synthesis
for high quality musical sound generation. The single-chip configuration
is capable of performing 1,200 sinusoid real-time synthesis; the system
is expandable to 13,200 partials by series connecting 11 chips. Each
sinusoid is generated by a marginally stable second order IIR filter,
and its frequency, amplitude and phase can be independently specified.
The system is clocked at 60 MHz when working with a 44.1 kHz sampling
rate. Two completely independent channels are available as output,
and each sample relies on a 20 bit representation to achieve an SNR
of at least 110 dB, thanks to the internal 24 bit word length. The
IC is designed in a 0.5 (mu)m CMOS technology and has a core area of
approximately 19 mm^2.
**
Title: A Generalized Musical-Tone Generator with Applications to Sound
Compression and Synthesis
Authors: 
Carlo Drioli, University of Padova
Davide Rocchesso, University of Padova
Volume: 1, Page: 431
Abstract:
A musical-tone generator based on physical modeling of the sound production
mechanisms is presented. To the purpose of making this scheme general
for a wide class of musical instruments, the nonlinear part of the
tone-generator is modeled by a neural network. The system learns its
parameters and the nonlinearity shape by means of nonlinear identification
procedures based on waveform or spectral matching.  Two possible applications
of this model are discussed: sound compression can be obtained when
considering the system as a nonlinear predictor, while sound synthesis
can be obtained by adding control inputs to the network and by training
the system to respond as desired.
**
Title: A Singing Voice Synthesis System Based on Sinusoidal Modeling
Authors: 
Michael Macon, Oregon Graduate Institute
Leslie Jensen-Link, Momentum Data Systems
James Oliverio, Georgia Institute of Technology
Mark A. Clements, Georgia Institute of Technology
E. Bryan George, Texas Instruments, Dallas
Volume: 1, Page: 435
Abstract:
Although sinusoidal models have been demonstrated to be capable of
high-quality musical instrument synthesis speech modification, and
speech synthesis, little exploration of the application of these models
to the synthesis of singing voice has been undertaken.  In this paper,
we propose a system framework similar to that employed in concatenation-based
text-to-speech synthesizers, and describe its extension to the synthesis
of singing voice.  The power and flexibility of the sinusoidal model
used in the waveform synthesis portion of the system enables high-quality,
computationally-efficient synthesis and the incorporation of musical
qualities such as vibrato and spectral tilt variation.  Modeling of
segmental phonetic characteristics is achieved by employing a "unit
selection" procedure that selects sinusoidally-modeled segments from
an inventory of singing voice data collected from a human vocalist.
The system, called LYRICOS, is capable of synthesizing very natural-sounding
singing that maintains the characteristics and perceived identity of
the analyzed vocalist.
**
Title: Time-Scale Modification of Audio Signals with Combined Harmonic
and Wavelet Representations
Authors: 
Khaled N. Hamdy, University of Minnesota
Ahmed H. Tewfik, University of Minnesota
Satoshi Takagi, Sony Corporation
Ting Chen, Stanford University
Volume: 1, Page: 439
Abstract:
We propose a new time-scale modification method for high quality audio
signals. Our approach strives to preserve pitch and timbre.  In our
method, the signal is represented as the sum of sinusoidal components
and a residual (edges and noise). The decomposition is computed via
a combined harmonic and wavelet representation.  Time-scaling is performed
on the harmonic components and residual components separately.  The
harmonic portion is time-scaled by demodulating each harmonic component
to DC, interpolating and decimating the DC signal, and remodulating
each component back to its original frequency.  The residual portion
is time-scaled by preserving edges and relative distances between the
edges while time-scaling the stationary (noise) components between
the edges.
**
Title: A Waveguide Model for Slapbass Synthesis
Authors: 
Erhard Rank, Vienna University of Technology
Gernot Kubin, Vienna University of Technology
Volume: 1, Page: 443
Abstract:
Starting from the waveguide model for plucked strings, a new digital
signal processing model for the slapping technique on electric bassguitars
is derived. The model includes amplitude limitations for the string
at the frets and/or the fingerboard. These highly nonlinear elements
are realized by conditional reflections which depend on the local string
displacement. A model of the string dynamics for the two slapbass techniques
- knocking the string with the thumb knuckle and plucking very strong
with the index or middle finger - has been implemented both as MATLAB
and C simulations and synthesizes sounds close to the natural instrument.
**
Title: Minimum Perceptual Spectral Distance FIR Filter Design
Authors: 
Shao-Po Wu, Stanford University
William Putnam, Stanford University
Volume: 1, Page: 447
Abstract:
This paper addresses the problem of designing finite impulse response
filters which optimally approximate desired frequency responses in
the sense that they minimize a perceptual audio spectral measure. This
measure is based on a simplified auditory model similar to those used
in the area of perceptual audio quality measurement.  It is shown that
this problem can be cast as a logarithmic Chebychev approximation problem,
which can be solved efficiently using recent interior point methods.
**
Title: A Phase Interpolation Algorithm for Sinusoidal Model Based Music
Synthesis
Authors: 
Xiaoshu Qian, URI
Yinong Ding, TI
Volume: 1, Page: 451
Abstract:
This paper presents a least square quadratic phase interpolation algorithm
for sinusoidal model based music synthesis.  This algorithm uses two
additions with one parameter per data frame to generate the phase samples
of a component sine wave.  Compared with the cubic phase interpolation
algorithm proposed by McAulay and Quatieri, the proposed algorithm
is more efficient in terms of computational complexity and parameter
storage. In the meantime, it also produces smoother frequency tracks.
Unlike the existing quadratic phase interpolation algorithm, where
the phase measurements are totally ignored ("magnitude-only"), the
proposed algorithm interpolates phase in a least square sense from
both the phase and the frequency measurements at data frame boundaries.
Thus the resulting phase samples are approximately "locked" to the
measured ones. Informal listening tests on various musical instrument
tones indicate that the proposed algorithm clearly outperforms the
magnitude-only synthesis approach and is qualitatively comparable to
the cubic one.
**
Title: Analytical Approximations of Fractional Delays: Lagrange Interpolators
and Allpass Filters
Authors: 
Stephan Tassart, IRCAM
Philippe Depalle, IRCAM
Volume: 1, Page: 455
Abstract:
We propose in this paper a new point of view which unifies two well
known filter families for approximating ideal fractional delay filters:
Lagrange Interpolator Filters (LIF) and Thiran Allpass Filters.  We
achieve this unification by approximating the ideal Fourier transform
of the fractional delay according to two different Pade approximations:
series expansions and continued fraction expansions, and by proving
that both approximations correspond exactly either to the LIF family
or to the allpass delay filters family. This leads to an efficient
modular implementation of LIFs.
**
Title: Improved discrete-time modeling of multi-dimensional wave propagation
using the interpolated digital waveguide mesh
Authors: 
Lauri Savioja, Helsinki University of Technology
Vesa Valimaki, Helsinki University of Technology
Volume: 1, Page: 459
Abstract:
The digital waveguide mesh is an extension of the one-dimensional digital
waveguide technique. Waveguide meshes are used for simulation of two-
and three-dimensional wave propagation in musical instruments and acoustic
spaces. The original waveguide mesh algorithm suffers from direction-dependent
dispersion. In this paper we show that this problem may be reduced
by using an interpolated rectilinear mesh. In the analysis part we
show the analytical solution for the wave propagation speed and numerical
simulations of the magnitude response and phase speed in both the original
and the interpolated two-dimensional waveguide mesh algorithms. We
demonstrate by simulation that the wave propagation characteristics
of the proposed interpolated waveguide mesh are independent of direction
and thus the remaining errors caused by dispersion may be corrected
with a postprocessor.
**
Title: Generalized Likelihood Ratio Test for Selecting a Geo-acoustic
Environmental Model
Authors: 
Christoph F. Mecklenbrauker, RUB
Peter Gerstoft, SACLANTCEN
Pei-Jung Chung, COMNETS
Johann F. Bohme, RUB
Volume: 1, Page: 463
Abstract:
A generalized likelihood ratio test is considered for testing acoustic
environmental models with application to parameter inversion using
an acoustic propagation code.  In the following, we use the term ``hierarchy
of models'' to denote a sequence of model structures  M_1, M_2,ldots
in which each particular model structure  M_n contains all previous
ones as special cases.  We propose a combined parameter estimation
and multiple sequential test for simultaneously determining the model
order and its parameters:  given the observed data, how many parameters
should be included in the model?  The last question is important for
the order selection problem in hierarchies of models with increasing
number of parameters where the observations are corrupted by additive
noise. Monte Carlo simulations show the behaviour of the sequential
test for selecting a model order as a function of the SNR. Finally,
the test is applied to broadband data measured using a vertical array
near the island of Elba in the Mediterranean Sea.
**
Title: Tuning Genetic Algorithms for Underwater Acoustics Using  a priori
Statistical Information
Authors: 
Maria Joao Rendas, I3S/CNRS
Georges Bienvenu, Thomson Marconi Sonar
Volume: 1, Page: 467
Abstract:
In this paper we present a new technique for the evaluation/selection
procedures of genetic algorithms, to be used in the context of parameter
estimation problems. The proposed algorithm uses  a priori information
about the structure of the surface of which an extremum is being searched.
For parameter estimation problems, the availability, at each iteration
of a genetic algorithm, of a collection of samples of the ambiguity
surface of the problem, enables the determination of the correlation
between the  observed ambiguity surface (at the sampled points) and
the  predicted ambiguity surface. The consideration of this information
allows early detection of secondary extrema (which yield an ambiguity
surface which does not correlate well with the observed one) and thus
contributes to speed the convergence of the algorithm to the global
optimal values. The paper applies the proposed technique to a source
localization problem.
**
Title: Robust Beamformer Design for Broadband Matched-Field Processing
Authors: 
Kerem Harmanci, Duke University
Jeffrey L. Krolik, Duke University
Volume: 1, Page: 471
Abstract:
Matched-field beamforming has been proposed for localizing wideband
acoustic sources in uncertain underwater channels. While adaptive matched-field
beamforming provides adequate sidelobe suppression for stronger sources,
at low signal-to-noise ratios it converges to its quiescent response,
in this case the Bartlett beamformer, which has unacceptably high sidelobe
levels. In this paper, a design method is presented for reducing matched-field
non-adaptive beamformer sidelobe levels given a sufficiently large
observation time-bandwidth product. The proposed alpha-beamformer incoherently
averages narrowband matched-field beamformer output power over the
signal band after a trade-off has been performed at each frequency
to achieve better sidelobe suppression at the expense of some reduction
in gain against diffuse noise.  Simulations and results with Mediterranean
vertical array data indicate that the wideband alpha-beamformer can
provide improved sidelobe suppression versus conventional techniques.
**
Title: FASTMAP: A Fast, Approximate Maximum A Posteriori Probability
Parameter Estimator with Application to Robust Matched-Field Processing
Authors: 
Brian F. Harrison, NUWCDIVNPT
Richard J. Vaccaro, University of Rhode Island
Donald W. Tufts, University of Rhode Island
Volume: 1, Page: 475
Abstract:
In many estimation problems, the set of unknown parameters can be divided
into a subset of desired parameters and a subset of nuisance parameters.
Using a maximum a posteriori (MAP) approach to parameter estimation,
these nuisance parameters are integrated out in the estimation process.
This can result in an extremely computationally-intensive estimator.
This paper proposes a method by which computationally-intensive integrations
over the nuisance parameters required in Bayesian estimation may be
avoided under certain conditions. The propsed method is an approximate
MAP estimator which is much more computationally efficient than direct,
or even Monte Carlo, integration of the joint posteriori distribution
of the desired and nuisance parameters. As an example of its efficiency,
we apply the fast algorithm to matched-field source localization in
an uncertain environment.
**
Title: Electromagnetic Matched Field Processing for Source Localization
Authors: 
Donald F. Gingras, Naval Command, Control and Ocean Surveillance Center
Peter Gerstoft, SACLANT
Neil L. Gerr, Office of Naval Research
Christoph F. Mecklenbrauker, Vienna University of Technology
Volume: 1, Page: 479
Abstract:
Matched field processing (MFP) refers to signal and array processing
techniques in which, rather than a planewave arrival model, complex-valued
(amplitude and phase) field predictions for propagating signals are
used.  Matched field processing has been successfully applied in ocean
acoustics.  In this paper the extension of MFP to the electromagnetic
domain, i.e., electromagnetic (EM) MFP (EM-MFP) is described.  Simulations
of EM-MFP in the tropospheric setting suggest that, under suitable
conditions, EM-MFP methods can enable EM sources to be both detected/localized
and used as sources of opportunity for estimating the environmental
parameters that determine EM propagation.
**
Title: Power-Law Processors for Detecting Unknown in Signals in Colored
Noise
Authors: 
Ivars P. Kirsteins, NUWCDIVNPT
Sanjay K. Mehta, NUWCDIVNPT
John Fay, NUWCDIVNPT
Volume: 1, Page: 483
Abstract:
We propose a new non-parametric adaptive detector for detecting an
unknown broadband signal in interference consisting of non-stationary
narrowband components and a locally stationary broadband component.
An important feature of this detector is that it needs no prior information
about the signal or interference. The proposed detector is based on
the integration of the non-parametric power law detector of Nuttall
with robust narrowband interference removal and whitening using a multiple
taper spectral estimation-based technique.  Experimental results indicate
that the proposed detector outperforms conventional detectors.
**
Title: Multitarget detection/tracking of echoes with known waveform:
algorithm and applications
Authors: 
Vittorio Rampa, C.S.T.S. - C.N.R.
Umberto Spagnolini, Politecnico di Milano
Volume: 1, Page: 487
Abstract:
The Time of Delay (TOD) estimation of multiple echoes is here solved
with an iterative multitarget detection/tracking algorithm. The evaluation
of the TODs is based on their a-posteriori probability, while a first-order
Markov model is used for a-priori probability estimation. The effectiveness
of the algorithm (low false-alarm rate and robustness) is also experimentally
proven. Moreover the algorithm exhibits a better noise rejection and
an improved target resolution with respect to algorithms that perform
separate detection and tracking.
**
Title: Detection of Gaussian Bandpass Transients Under Impulsive Noise:
A Wavelet Transform Approach
Authors: 
Francisco M. Garcia, ISR - IST
Isabel M.G. Lourtie, ISR - IST
Volume: 1, Page: 491
Abstract:
In underwater acoustics, the modeling of impulsive noise ambients by
symmetric-alpha-stable laws is motivated by the generalized central
limit theorem. However, detection of stochastic signals under such
additive noise is a difficult task to implement, due to the lack of
a closed-form expression of the a-posteriori probability density function.
In this paper, we present a suboptimal detector for Gaussian bandpass
transients in impulsive noise that uses a nonlinear, memoryless prefilter
followed by a discrete wavelet transform. The resulting signals present
a Gaussian-like behavior and the decision is achieved by the comparison
of a quadratic likelihood ratio with a threshold. The tuning of the
nonlinearity parameter is performed either by looking at the receiver
operating characteristic or using the Chernoff distance, that, although
resulting in an approximate solution, is easier to compute.  Simulation
results are presented by Monte-Carlo simulation.
**
Title: Maximum Likelihood Estimator for Magneto-Acoustic Localisation
Authors: 
Gilles Dassot, LETI CEA/Grenoble
Roland Blanpain, LETI CEA/Grenoble
Claude Jauffret, GESSY, University Toulon et Var
Volume: 1, Page: 495
Abstract:
This paper is devoted to the localization of magneto- acoustic sources
moving in a straight line at a constant speed. Our technique is based
on the association of narrow band acoustic signals and magnetostatic
measurements.  First of all, we describe features that make possible
the association of magnetic and acoustic data, secondly, we show that
positioning accuracy is much improved by this association. In this
paper we focus on solving the problem with as few sensors as possible.
A geometric discussion of identifiability is proposed, as well as a
Batch Maximum Likelihood estimator whose covariance matrix asymptotically
achieves Cramer Rao Lower Bounds (CRLB).
**
Title: Barankin Bound for Source Localization in Shallow Water
Authors: 
Joseph Tabrikian, Duke University
Jeffrey L. Krolik, Duke University
Volume: 1, Page: 499
Abstract:
Matched-field methods are known to have a severe ambiguity problem.
In low signal-to-noise-ratios (SNR's), where the estimator cannot distinguish
between the ambiguity function peak near the true source location and
ambiguous ones, its mean square error deviates radically from the Cramer-Rao
lower bound (CRLB). In this paper, the Barankin bound for the source
localization problem in an uncertain shallow water environment is derived.
In particular, a method of selection of the test-points for evaluation
of the bound is presented. The bound is evaluated using a ``general
mismatch'' benchmark scenario. The results presented here predict the
threshold SNR below which the performance degrades dramatically. Channel
uncertainties in the benchmark scerario are shown to increase this
threshold SNR by as much as 3dB.
**
Title: Underwater transient signal processing: marine mammal identification,
localization, and source signal deconvolution
Authors: 
Zoi-Heleni Michalopoulou, CAMS, NJIT
Volume: 1, Page: 503
Abstract:
Processing marine-mammal signals for species classification and monitoring
of endangered marine mammals are problems that have recently attracted
attention in the scientific literature.  For classification it has
been proposed to use methods appropriate for non-stationary signals,
such as time-frequency and time-scale analysis.  This paper shows that
a factor that can significantly affect results from marine-mammal signal
processing is the impulse response of the ocean in which the signals
propagate.  The ocean is a dispersive propagation medium and, therefore,
affects the time-frequency characteristics of a propagating acoustic
signal.  Because of this distortion, feature selection should be performed
after the oceanic impulse response has been deconvolved from the recorded
signals. The paper also discusses localization of vocalizing marine
mammals using matched-field processing and shows how this becomes a
part of the deconvolution process.
**
Title: Numerical Optimization of Non-adaptive Microphone Arrays
Authors: 
Alexander Goldin, IBM Israel S&T
Volume: 1, Page: 507
Abstract:
The paper describes an application of the numerical optimization methods
for the design of non-adaptive multi-sensor arrays. The parameters
and the geometry of such arrays do not change with changes in the input
signals, and must be chosen in advance. Generally, the goal of a non-adaptive
multi-sensor array may be numerically expressed through its pattern
function which shows the gain for a signal coming from a particular
direction in space. The real pattern function depends on the geometry
of the array and on the processing which signals from every sensor
undergo. The array pattern function is non-linear and it is frequency
dependent. The geometry and the processing parameters of the multi-sensor
array are optimized to provide the minimum difference between the goal
and the real functions over a specified frequency range. Optimization
results for several goal functions for multi-microphone arrays are
provided and discussed.
**
Title: Joint Direction-of-Arrival and Array Shape Tracking for Multiple
Moving Targets
Authors: 
Jason Goldberg, Tel Aviv University
Ana Perez-Neira, UPC
Miguel Lagunas, UPC
Volume: 1, Page: 511
Abstract:
An algorithm for the joint tracking of source DOA's and sensor positions
is presented to address the problem of DOA tracking in the presence
sensor motion. Initial maximum likelihood estimates of source DOA's
and sensor positions are refined by Kalman filtering. Spatio-temporally
correlated array movement is considered. Source angle dynamics are
used to achieve correct data association. The new technique is capable
of performing well for the difficult cases of sources that cross in
angle, fully coherent sources, as well as sources of identical or vastly
different (possibly time-varying) power. Computer simulations show
that the approach is robust in the presence of array motion modeling
uncertainty and effectively reduces dependence on expensive and possibly
unreliable hardware.
**
Title: Comparison of Probabilistic Least Squares and Probabilistic Multi-Hypothesis
Tracking Algorithms for Multi-Sensor Tracking
Authors: 
Mark L. Krieg, DSTO, CSSIP, University of Adelaide
Douglas A. Gray, University of Adelaide, CSSIP
Volume: 1, Page: 515
Abstract:
A key element for successful tracking is knowing from which target
each measurement originates.  These measurement-to-target associations
are generally unavailable, and the tracking problem becomes one of
estimating both the assignments and the target states.  We present
the Probabilistic Least Squares Tracking (msPLST) algorithm for estimating
the measurement-to-target assignments and the track trajectories of
multiple targets, using measurements from multiple sensors.  This is
a different approach to that used in Probabilistic Multi-Hypothesis
Tracking (PMHT), although both algorithms employ the concept of an
extended observer containing both the target states and the measurement-to-target
assignments.  A comparison of both algorithms is made, and their performance
is evaluated using simulated data.
**
Title: Direction Finding with Imperfect Wavefront Coherence: A Matrix
Fitting Approach Using Genetic Algorithm
Authors: 
Alex B. Gershman, Ruhr University Bochum
Christoph F. Mecklenbrauker, Ruhr University Bochum
Johann F. Bohme, Ruhr University Bochum
Volume: 1, Page: 519
Abstract:
The performance of high-resolution direction finding methods degrades
in several practical situations where the wavefronts have imperfect
spatial coherence. The original solution to this problem was proposed
by Paulraj and Kailath, but their technique requires  a priori knowledge
of the matrix characterizing the loss of wavefront coherence along
the array aperture. Below, a novel solution to this problem is proposed,
which does not require  a priori knowledge of the spatial coherence
matrix. Our technique is based on the multidimensional minimization
of appropriate concentrated cost function using Genetic Algorithm (GA).
**
Title: Design Of An Optimum Wideband Active Sonar Array With Robustness
Authors: 
Saman S. Abeysekera, Curtin University of Technology
Y.H. Leung, Curtin University of Technology
Volume: 1, Page: 523
Abstract:
The use of wideband active sonar array processing to estimate the range,
velocity and bearing of a target has received much interest in the
literature recently.  Although increased attention has been focused
on wideband correlation processing for estimating range and velocity,
array directivity patterns are almost always computed and interpreted
under the narrowband signal assumption. This paper considers the target
bearing estimation problem using the wideband correlation approach.
Via this approach, it will be shown how an optimum set of array weights
can be selected for a known transmitted signal. The optimization procedure
also provides robustness against errors in the array structure.
**
Title: Multipath time-delay estimation
Authors: 
Jean Jacques Fuchs, IRISA
Volume: 1, Page: 527
Abstract:
A transmitted and known signal is observed at the receiver through
more than one path in additive noise. The problem is to estimate the
number of paths and for each of them the associated attenuation and
delay. It is a frequent problem in sonar, radar and geophysics. We
propose an algorithm that is easy to implement, that has a reasonable
computational load and seems to be able to solve the problem under
more severe conditions (lower SNR) than previous methods.
**
Title: Fast Maximum Likelihood Estimation With Multiple Signal Initialization
Authors: 
Robert B. MacLeod, NUWC
Richard J. Vaccaro, University of Rhode Island
Volume: 1, Page: 531
Abstract:
In this paper we are concerned with signal processing of acoustic signals
resulting from active transmissions by high frequency sonar systems.
These signals consist of structured interference related to propagation
effects in the media, reflections from targets, and measurement noise.
The methods herein model these signals as replicas of the transmitted
signal, scaled in amplitude and time, and delayed.  Furthermore, we
are interested in signals with `simple' time frequency profiles, such
as linear frequency modulated (LFM) or hyperbolic frequency modulated
(HFM) signals.  These signals have the underlying property that the
principle ridge of the autoambiguity function crosses the mid point
of the time-frequency plane in a smooth manner, with a simple relationship
between time delay and time scaling (frequency shifting). This paper
describes a method for estimating the delay and time scale of signal
components using fast maximum likelihood, while preserving the high
resolution property of related time delay estimation techniques.
**
Title: An Algorithm for Detecting Closely Spaced Delay/Doppler Components
Authors: 
Amir W. Habboosh, NUWCDIVNPT
Richard J. Vaccaro, University of Rhode Island
Steven M. Kay, University of Rhode Island
Volume: 1, Page: 535
Abstract:
This paper considers a method for estimating time delays, amplitudes,
and Doppler scales of a multipath signal.  The method is an extension
of work previously reported by Manickam and Vaccaro which dealt solely
with time delays and amplitudes, and extended by Habboosh and Vaccaro
to include Doppler scale.  In this paper, an algorithm is presented
for determining the size of the indicator set to reduce ill-conditioning
of the signal subspace matrix.  Simulation results are shown and comparisons
to the Cramer-Rao lower bound provided; these results show that significant
reduction in estimate variances can be achieved using the deconvolution
approach with a properly selected indicator set.
**
Title: Improvement of TDOA Measurement using Wavelet Denoising with a
Novel Thresholding Technique
Authors: 
Shi Quan Wu, The Chinese University of Hong Kong
Hing Cheung So, City University of Hong Kong
Pak Chung Ching, The Chinese University of Hong Kong
Volume: 1, Page: 539
Abstract:
In this paper, wavelet denoising is applied in time delay estimation
between signals received at two spatially separated sensors in the
presence of noise. Prior to cross correlation, each of the sensor outputs
is denoised according to a n