NEWS LETTER ISSN 1687-1421, Volume 17, Number 3, September 2006

EURASIP NEWS LETTER ISSN 1687-1421, Volume 17, Number 3, September 2006

European Association for Signal Processing

Newsletter, Volume 17, Number 3, September 2006

Contents

EURASIP MESSAGES President’sMessage ...... 1 2006 EURASIP AdCom Elections Bios and Photos of the Candidates ...... 3 2006 EURASIP AdCom Elections Form ...... 6 EURASIP Secretary-Treasurer’s Report 1st July, 2005–30th June, 2006 ...... 7

OTHER NEWS FaceRecognitionHomepage ...... 9

STUDENT ACTIVITIES JuniorFacultyPositioninAppliedMathematicsforMultimedia ...... 11

SHORT TUTORIALS DigitalSatelliteCommunicationTechniques:AReview ...... 12 Testbeds and Rapid Prototyping in Wireless System Design ...... 32

EURASIP (CO-)SPONSORED EVENTS Report on the 48th International Symposium ELMAR-2006 focused on Multimedia Signal Processing and Communications, 07-09 June 2006, Zadar, Croatia ...... 51 Report on XI International Conference SPECOM ’2006 ...... 54 CalendarofEvents ...... 56 Call for Papers: The Fourth IASTED International Conference on SignalProcessing,PatternRecognition,andApplications ...... 57 CallforPapers:AdvancedConceptsforIntelligentVisionSystems ...... 58 Call for Papers: 6th EURASIP Conference EC-SIPMCS 2007 ...... 59 Call for Papers: 1st International Conference on Semantic and Digital Media Technologies (Samt 2006) ...... 60 Call for Papers: 15th European Signal Processing Conference (EUSIPCO 2007) ...... 61 Call for Papers: International Symposium on Signal Processing and its Applications (ISSPA 2007) ...... 62 Call for Papers: 13th International Conference on Systems, Signals & Image Processing (IWSSIP ’06) ...... 63 Call for Papers: International Workshop on Multimedia Content Representation, ClassiﬁcationandSecurity(IWMRCS) ...... 64 Call for Papers: An ISCA Tutorial and Research Workshop on Non-Linear SpeechProcessing(NOLISP’07) ...... 65 Call for Papers: 26th Picture Coding Symposium (PCS 2007) ...... 66 Call for Papers: 2007 International Waveform Diversity & Design Conference ...... 67 Call for Papers: 8th International Workshop on Image Analysis for Multimedia Interactive Services (WIAMIS 2007) ...... 68 Call for Papers: 2006 International Workshop on Acoustic Echo and Noise Control (IWAENC 2006) ...... 69 Call for Papers: International Conference on Artiﬁcial Neural Network(ICANN’06) ...... 70 Call for Papers: The International ITG / IEEE Workshop on Smart Antennas (WSA 2007) ...... 71

EURASIP JOURNALS EURASIPJournalonAppliedSignalProcessing ...... 72 EURASIPJournalonAudio,Speech,andMusicProcessing ...... 74 EURASIPJournalonBioinformaticsandSystemsBiology ...... 75 EURASIPJournalonEmbeddedSystems ...... 77 EURASIP Journal on Wireless Communications and Networking ...... 78 SignalProcessing ...... 79 SignalProcessing:ImageCommunication ...... 80 SpeechCommunication ...... 81

EURASIP Journal on Applied Signal Processing Abstracts of EURASIP JASP, Volume 2006, Regular Issue ...... 83 Abstracts of EURASIP JASP, Volume 2006, Special Issue on AdvancesinMultimicrophoneSpeechProcessing ...... 94 Abstracts of EURASIP JASP, Volume 2006, Special Issue on DesignMethodsforDSPSystems ...... 102 Abstracts of EURASIP JASP, Volume 2006, Special Issue on PerformanceEvaluationinImageProcessing ...... 112 Abstracts of EURASIP JASP, Volume 2006, Special Issue on AdvancedSignalProcessingTechniquesforBioinformatics ...... 121 EURASIPJASPCallforPapers ...... 128

EURASIP Journal on Wireless Communications and Networking Abstracts of EURASIP JWCN, Volume 2006, Regular Issue ...... 136 Abstracts of EURASIP JWCN, Volume 2006, Special Issue on QualityofServiceinMobileAdHocNetworks ...... 143 EURASIPJWCNCallforPapers ...... 150

EURASIP Journal on Embedded Systems Abstracts of EURASIP JES, Volume 2006, Regular Issue ...... 156 Abstracts of EURASIP JES, Volume 2006, Special Issue on Processing with High Complexity: Prototyping and Industrial Design ...... 157 EURASIPJESCallforPapers ...... 161

EURASIP Journal on Bioinformatics and Systems Biology Abstracts of EURASIP JBSB, Volume 2006, Regular Issue ...... 163

EURASIP Journal on Audio, Speech, and Music Processing ...... 164 EURASIP Journal on Bioinformatics and Systems Biology ...... 165 EURASIP Journal on Embedded Systems ...... 166 International Journal of Biomedical Imaging ...... 167 EURASIP Journal on Information Security ...... 168

EURASIP Book Series on Signal Processing and Communications EURASIPBookSeriesonSignalProcessingandCommunications ...... 169 SignalProcessingfortheAcousticHuman-MachineInterface ...... 170 AdvancesonNonlinearSignalandImageProcessing ...... 171 GeneticandEvolutionaryComputationforImageProcessingandAnalysis ...... 172 AdvancesinSignalTransforms:TheoryandApplications ...... 173

How to Become a EURASIP Member ...... 174 EURASIP Membership Application ...... 176

EURASIP MESSAGES

President’s Message

This is my last President’s message, since my two-years term ends in September and the new Adcom, that will result from the elections in the general assembly meeting in Florence, will elect the new EURASIP President. From now on and for the next two years I will serve in the Adcom from the past-President’s position. Over the period of these two years a lot has taken place and of course a lot more could have taken place. My most important and valuable experience was to watch my relationship with the other members of the Adcom: Ferran Marques, Marc Moonen, Markus Rupp, Paulo Correira and Fulvio Gini to be transformed from that of a professional cooperation into a friendship. If there is one thing that I gained over this period is this enriching feeling and I would like to thank them publicly for all their work and dedication to EURASIP. Nothing would have happened without their constant effort and support. The Adcom took a number of critical decisions and initiatives during the last two years, responding to the needs of our scientific community in a fast changing world. Five new journals were launched: Journal on Wireless Networks and Networking, Jour- nal on Bioinformatics and Systems Biology, Journal on Embedded Systems, Journal on Au- dio, Speech and Music Processing, Journal on Information Security. For all of them, the Open Access publishing model was adopted. As I have commented in previous messages, this publishing model, that fully exploits the current technological advances without sacrificing quality, guarantees a much higher visibility to a published article. Statistical studies have shown that the average citation index per article increases, since, obviously, it can be downloaded freely by anybody. Also, such a model, seems to be socially more fair, since colleagues from less developed countries and less privileged universities, with limited library resources can have better access to the published literature. We have come to a new agreement with ELSEVIER, one of our two publishers. The EURASIP Adcom will have the full control on the scientific aspects for all our journals. The Editorial Board members will serve for a period of time and they will be appointed jointly by the EIC and the Adcom. This regular change will guarantee that new people with enthu- siasm and new ideas will be given the opportunity to serve as Associate Editors, offering the scientific “vitality” that a journal needs in our days, where new areas appear and flourish in a very short time. The Editors in Chief will also be serving for a limited period of time and it will be the Adcom’s main responsibility to guarantee the high scientific competence and respect that an EIC must have, in order to serve the scientific community successfully. All EUSIPCO Proceedings, since 1996, are now Open Access. At any time, only the Pro- ceedings of the last EUSIPCO will be freely available to members only. All previous EU- SIPCO proceedings will be free to download by anybody. We feel that such a policy will make EUSIPCOs more attractive, due to the higher visibility a published article can now enjoy. A new award, the “Athanassios Papoulis” award has been decided to be granted, in order to honor colleagues whose work had a major impact in various aspects on SP education. 2 EURASIP Messages

The award will be offered for the first time next year, and it is intended to be a very presti- gious one. It will be offered only on demand, every time there is an exceptional candidate and not on a regular period of time. Our web system for electronic registration and journal subscription is now in full operation after overcoming the unavoidable “baby” diseases! Closing this note I would like to thank all the EURASIP officers, the EIC and all our Associate Editors whose volunteering work keeps EURASIP alive and vibrant. Last but not least I would like to thank all of you who honored me with your vote that gave me the privilege to serve EURASIP over all these years. It was such a rich experience.

Sergios Theodoridis President EURASIP MESSAGES

2006 EURASIP AdCom Elections Bios and Photos of the Candidates

Jean-Luc Dugelay received the Ph.D. degree in Computer Science in 1992 from the University of Rennes. Doctoral research was carried out, from 1989 to 1992, at the France Telecom Research Laboratory in Rennes (formerly CNET-CCETT). He then joined the Institut Eurcom (Sophia Antipolis), where he is currently a Professor in the Department of Multimedia Com- munications. His research interests are in the area of multimedia signal processing and communications; including security imaging (i.e., watermarking and biometrics), facial image analysis and talking heads. He is an author or coauthor of more then 70 publications that have appeared as journal papers or proceeding articles, 3 book chapters, and 3 international patents. He gave several tutorials on digital watermarking (co-authored with F. Petitcolas from Microsoft Re- search), biometrics (co-authored with J.-C. Junqua from Panasonic Research) at major conferences. He has been an invited speaker and/or member of the program committee of several scientiﬁc conferences and workshops. He was technical co-chair and organizer of the fourth IEEE workshop on Mul- timedia Signal Processing (Cannes, October 2001. His group is involved in several national projects (SEMANTIC 3D, BIOBIMO) and European NoEs related to multimedia and security (SIMILAR, ECRYPT, BIOSECURE). Jean-Luc Dugelay is a senior member of the IEEE Signal Processing Society, and served as associate editors for several journals (IEEE Trans. on IP, IEEE Trans. on MM, EURASIP JASP). He is currently the Editor in Chief of the International Journal on Image and Video Process- ing (http://www.hindawi.com/journals/ijivp/). Homepage:http://www.eurecom.fr/∼dugelay e-mail: [email protected]

Bulent¨ Sankur has received his B.S. degree in Electrical Engineering at Robert College, Istanbul and completed his graduate studies at Rensselaer Polytechnic Institute, New York, USA. He has been teaching at Bogazii (Bosphorus) University in the Department of Electric and Electronic En- gineering. His research interests are in the areas of Digital Signal Process- ing, Image and Video Compression, Biometry, Cognition and Multimedia Systems. He has established a Signal and Image Processing laboratory at Bogazic¸i University and he has been publishing in diverse areas. Dr. Sankur has held visiting positions at University of Ottawa, Technical University of Delft, and Ecole Nationale Superieure´ des Tlcommications, Paris. He also served as a consultant in several private and government institutions. He was the chairman of ICT’96: International Conference on Telecommunications and EUSIPCO’05: The European Conference on Signal Processing as well as technical chairman of ICASSP’00. 4 EURASIP Messages

Fulvio Gini received the Doctor Engineer (cum laude) and the Research Doctor degrees in electronic engineering from the University of Pisa, Italy, in 1990 and 1995 respectively. In 1993 he joined the Department of Ingegneria dellInformazione of the University of Pisa, where he is an Associate Professor since October 2000. From July 1996 through January 1997, he was a visiting researcher at the Department of Electrical Engineering, University of Vir- ginia, Charlottesville. He is an Associate Editor for the IEEE Transactions on Signal Processing and a Member of the EURASIP JASP Editorial Board. He was corecipient of the 2001 IEEE AES Societys Barry Carlton Award for Best Paper. He was recipient of the 2003 IEE Achievement Award for outstanding contribution in signal processing and of the 2003 IEEE AES Society Nathanson Award to the Young Engineer of the Year. He is a Member of the Signal Processing Theory and Methods (SPTM) Tech- nical Committee (TC) of the IEEE Signal Processing Society and a Member of the Sensor Array and Multichannel (SAM) TC. He is also Member of the SPTM Awards sub-committee of the IEEE Signal Processing Society. He is a Member of the Administrative Committee of the EURASIP Society and the Award Chairman. He is Technical co-Chairman of the 2006 EURASIP Signal and Image Process- ing Conference (EUSIPCO), Florence, Italy, September 2006, and the Technical co-Chairman of the 2008 IEEE Radar Conference, Rome, Italy, May 2008. He is the guest co-editor of the special section of the Journal of the IEEE Signal Processing Society on Special Topics in Signal Processing on “Adap- tive Waveform Design for Agile Sensing and Communication,” to be published in 2007. He is the guest editor of the special section of the IEEE Signal Processing Magazine on “Knowledge Based Systems for Adaptive Radar Detection, Tracking and Classiﬁcation,” published in the January 2006 issue. He is Guest co-editor of the special issue of the EURASIP Signal Processing journal on “New trends and ﬁndings in antenna array processing for radar,” appeared in Vol. 84, 2004. His research interests include modeling and statistical analysis of radar clutter data, non-Gaussian signal detection and estimation, parameter estimation and data extraction from multichannel interferometric SAR data. He authored or co-authored two book chapters, more than 70 journal papers and 70 conference papers.

Beatrice Pesquet-Popescu received the engineering degree in telecommunications from the “Politehnica” Institute in Bucharest in 1995 (first ranked over 600 students) and the Ph.D. thesis from the Ecole Normale Superieure´ de Cachan in 1998 (with honours). In 1998 she was a Research and Teach- ing Assistant at Universite´ Paris XI and in 1999 she joined Philips Re- search France, where she worked during two years as a research scientist, then project leader, in scalable video coding. Since Oct. 2000 she is an Associate Professor in multimedia at the Ecole Nationale Suprieure des Tel´ ecommunications´ (ENST). In October 2005 she received the “Habilita- tion a` Diriger des Recherches” (French diploma necessary to get a Professor position). Her current research interests are in scalable and robust video coding, adaptive wavelets and multimedia applications. She actively participates to the MPEG standardization activities on scalable video coding (joint team with ITU for the next Scalable Video Coding extension of MPEG- 4/AVC and the exploration activities on wavelet-based scalable video coding, for which she also served as co-AhG chair). Beatrice Pesquet-Popescu is an EURASIP member and a Senior Member IEEE. Awards EURASIP gave her a “Best Student Paper Award” in the IEEE Signal Processing Workshop on Higher-Order Statistics in 1997, and in 1998 she received a “Young Investigator Award” granted by the French Physical Society. At Philips Research, she received the Bronze Inventor Medal (after one year in the company, granted for the number of patents filled in the last five years). She also received, together with Deepak Turaga and Mihaela van der Schaar, the 2006 IEEE Circuits and President’s Message 5

Systems Society Circuits and Systems for Video Technology Transactions Best Paper Award for her paper on “Complexity Scalable Motion Compensated Wavelet Video Encoding”. Professional Activities Member of the IEEE Signal Processing Society (SPS) Multimedia Signal Processing (MMSP) Technical Committee (TC).

Søren Holdt Jensen was born in Haderslev, Denmark in 1964. He received the M.Sc. degree in electrical engineering from Aalborg University, Aalborg, Denmark in 1988, and the Ph.D. degree in signal processing from the Tech- nical University of Denmark, Lyngby, Denmark in 1995. He is currently an Associate Professor at Aalborg University. Before joining the Department of Electronic Systems of Aalborg University, he was been with the Telecommu- nications Laboratory of Telecom Denmark Ltd., Copenhagen, Denmark; the Electronics Institute of the Technical University of Denmark; the Scientiﬁc Computing Group of the Danish Computing Center for Research and Edu- cation (UNI•C), Lyngby, Denmark; the Electrical Engineering Department of Katholieke Universiteit Leuven, Leuven, Belgium; and the Center for PersonKommunikation (CPK) of Aalborg University. His research interests span the areas of statistical signal processing, speech and audio processing, multimedia technology, digital communications, and satellite based navigation. He has supervised more than 50 Ph.D./M.Sc. thesis candidates, and served as a technical consultant to industry and the Serious and Organized Crime Agency of the Danish National Police in the area of speech communications and enhancement. He has published widely within applied signal processing and is coauthor of the textbook A Software-Deﬁned GPS and Galileo Receiver: Single-Frequency Approach (Birkhuser, 2006). Dr. Jensen is Senior Member of the IEEE, a former Chairman of the IEEE Denmark Section, and Founder, Interim Chairman, and Chairman of the IEEE Denmark Sec- tion’s Signal Processing Chapter. He is serving as Associate Editor for the IEEE TRANSACTIONS ON SIGNAL PROCESSING and Member of the Editorial Board of EURASIP Journal on Applied Signal Processing. He has also guest-edited two special issues for the EURASIP Journal on Applied Signal Processing on Anthropomorphic Processing of Audio and Speech and Digital Signal Process- ing in Hearing Aids and Cochlear Implants. At this year’s European Signal Processing Conference (EUSIPCO-2006) he is Area Chair for speech and audio processing. At Aalborg University he has been Member of the Study Board for Electronics and Information Technology and is currently Member of the Academic Council for the Faculty of Engineering and Science. He has been Head of Aalborg Uni- versity’s Ph.D. program on Wireless Communications, and is currently Member of the Management Board and PhD Study Director of the university’s Doctoral School of Technology and Science with around 400 Ph.D. students enrolled. In 2004 he received the Teacher of the Year Award in Electronics and Information Technology as voted by the student body. EURASIP MESSAGES

2006 EURASIP AdCom Elections Form

Deadline for Mail voting: 28 August 2006 (for return address see below) Deadline for Direct voting: 6 September 2006 (General Assembly Meeting at EUSIPCO 2006)

Please cross up to three of the candidates and mail it to the AdCom’s Secretary

Prof. Paulo Lobato Correia Instituto de Telecomunicacoes Instituto Superior Tecnico 1049-001 Lisboa, Portugal

Jean-Luc Dugelay Bulent¨ Sankur Fulvio Gini Beatrice Pesquet-Popescu Søren Holdt Jensen EURASIP MESSAGES

EURASIP Secretary-Treasurer’s Report 1st July, 2005–30th June, 2006

The opening balance on the 1st of July 2005 was as speciﬁed in the following table, in Euros (€). The currency conversion considered (30th June 2006) was: 1CHF= 0,63942 €

Opening balance (1st July, 2005) €€ Current accounts: EURO account 45328,88 CHF account 13768,02 total 59096,90 Savings accounts: EURO Money Market account 32813,94 total 32813,94

Total available 91910,84 Loanstobereimbursed: EUSIPCO ’2005 15000,00 EUSIPCO ’2006 15000,00 ISSM ’2005 1000,00 total 31000,00 Total 122910,84

The main EURASIP account movements during the ﬁnancial period considered are docu- mented in the following two tables, for income and expenses, respectively:

Income: €€ Membership (incl. Journal subscriptions) 61016,28 Donations/review charges: EUSIPCO ’2005 12500,00

Total income 73516,28 Reimbursed loans: EUSIPCO ’2005 15000,00

Total 88516,28 8 EURASIP Messages

Expenses: €€ Elsevier (various concepts) 21678,44 Hindawi (various concepts, incl. Newsletter) 54432,95 EURASIP Awards 4977,35 Administrative expenses 11374,31 Taxes, bank costs, interests, currency conversions −345, 47 Total expenses 92117,58

The closing balance on the 30th of June 2006 is as speciﬁed in the table below:

Closing balance (30th June, 2006) €€ Current accounts: EURO account 39974,88 CHF account 14965,58 total 54940,46 Savings accounts: EURO Money Market account 33369,08

Total available 88309,54 Loanstobereimbursed: EUSIPCO ’2006 15000,00 ISSM ’2005 1000,00 total 16000,00 Pending payments 88309,54 Elsevier funding 10000,00 Hindawi award funding 2000,00 total 12000,00 ISSM ’2005 Loss 1000,00

Total 115309,54 OTHER NEWS

Face Recognition Homepage

www.face-rec.org

Over the last ten years or so, face recognition has become a popular area of research in computer vision and one of the most successful applications of image analysis and understanding. Because of the nature of the problem, not only computer science researchers are interested in it, but neuroscientists and psychologists also. It is the general opinion that advances in computer vision research will provide useful insights to neuroscientists and psychologists into how human brain works, and vice versa. Since face recognition is an important research area, it is compulsory to give prospective researchers a place to start on the Internet. Face Recognition Homepage aims to provide scientists with the relevant information in the area of face recognition. This web-site is intended to be an information pool for the face recognition community. Its goal is to provide an entry point for novices as well as a centralized information resource to concentrate face recognition and related scientiﬁc eﬀorts. The materials currently published on the web-site are what every researcher in this area should read. Web-site is designed to be a portal, so it contains numerous links to other research groups, links to relevant journals, books, databases etc. Another issue also worth mentioning is that this is a non-commercial, purely academic web-site. 10

Face Recognition Homepage provides the following information: • general information about face recognition problems; • over 50 links to research groups (from more than 20 countries) that work in the area of face recognition; • general papers and standards that deal with the face recognition (over 15 PDFs), as well as cognitive vision / psychology / neuroscience papers (over 10 PDFs); • new papers published in the last six months in high impact factor journals with most recent advances in face recognition (usually more than 50 papers); • links with a short description of more than 25 face databases often used by researchers; • face recognition algorithms (image-based and video-based), divided into 16 sub- categories, with a few most representative papers for each algorithm (total: more than 60 PDFs); • list of upcoming and past conferences, where face recognition is the conference topic; • high impact factor journals which scope covers face recognition (23 links); information about some journal special issues (5 links); books on face recognition (4 links); related books (9 links); • links to 15 vendors developing face recognition technology that participated in FRVT 2002 or FRGC; • other related links (26 links).

Please feel free to inform all your colleagues who might be interested in this web-site. If you like the web-site yourself, please feel free to put a link to it on your web-page as well. Hopefully, this eﬀort will increase the number of researchers in this area and improve the state-of-the-art in face recognition. Feel free to send any comments, suggestions and corrections that could improve Face Recognition Homepage to Prof. Mislav Grgic ([email protected]) and Mr. Kresimir Delac ([email protected]), University of Zagreb, Faculty of Electrical Engineering and Computing, Unska 3 / XII, HR-10000 Zagreb, Croatia.

www.face-rec.org STUDENT ACTIVITIES

Junior Faculty Position in Applied Mathematics for Multimedia

The Multimedia Department of the Institut Eurecom´ invites applications for an assistant professor position (Maˆıtre de Confrences, in French). We seek a junior candidate who has recently obtained a PhD and who can provide evidence of high potential in both research and teaching. The successful applicant is expected to have a strong background on applied mathematics for multimedia and/or multimedia signal processing. Example areas of interest include but are not limited to: fusion theory in multimodal systems, probabilistic approaches in multimedia applications, optimization processes, data classification, pattern recognition. Excellent candidates in areas outside those targeted may be considered. Candidates must have a PhD in the domain of electrical engineering, computer science or applied mathematics. Excellent skills of speaking and writing in English are mandatory, knowledge of French is welcome. He/She should contribute to enforce existing research within the department which includes important activities in image and video processing, indexing, watermarking and affective computing. The department is involved in several national and European projects related to these domains of expertise. The position is open from 1 July 2006. The search will continue until the position is filled. Applicants should send a resume, a letter of motivation (including a paragraph on a tentative research plan) together with names and addresses of three references to:

Professor Bernard Merialdo, Head, Multimedia Communications Dept. Institut Eurecom 2229 route des Cretes,ˆ B.P. 193 06904 Sophia Antipolis Cedex Fax. : +33 (0)4 93 00 82 00 Or by email to [email protected]

Eurecom´ is a teaching and research institute, with an international Master program in telecommunications, and a reputation of excellence in research. It is located in Sophia An- tipolis, an active technopole on the French Riviera, gathering a large number of research units of leading multi-national corporations in the information technologies and biotech- nology sectors, as well as other leading research and teaching institutions. A multinational population and the unique geographic location provide an outstanding quality of life. http://www.eurecom.fr/mm/welcome.en.htm SHORT TUTORIALS

Digital Satellite Communication Techniques: A Review

A. Vanelli-Coralli Dipartimento di Elettronica, Informatica, Sistemistica, UniversitadiBologna,` Viole Risorgimento, 2, Bologna, Italy

G. E. Corazza Dipartimento di Elettronica, Informatica, Sistemistica, UniversitadiBologna,` Viole Risorgimento, 2, Bologna, Italy

C. Bazile CNES, 31055 Touleuse Cedex, France

P. B i t h a s Institute for Space Applications and Remote Sensing, National Observatory of Athens, 15236 Palaia Pentli, Athens, Greece

S. Cioni Facolta` di Ingegnria, Universita` di Bologna, Viole Risorgimento, 2 Bologna, Italy

L. Castanet CERT, ONERA, BP 4025, 31055 Toulouse Cedex 4, France

A. Duverdier CNES, 31055 Touleuse Cedex, France

W. Gappmair Institute fur¨ Kommunikations netze und Satellilenkommunikation, Technische Universitat¨ Graz, Inﬀeldgasse 12, 8010 Graz, Austria

M. Luglio Department of Electronic Engineering, University of Rome, Tor Vergata, Via del Politecnico, 1-00133 Rome, Italy

P.T. Mathiopolous Institute for Space Applications and Remote Sensing, National Observatory of Athens, 15236 Polaia, Pentli, Athens, Greece Digital Satellite Communication Techniques: A Review 13

C. Mosquera Department of Signal Theory and Communications, University of Vigo, Vigo (Pontevedra) 36310, Spain

L. S. Ronga CNIT, University of Florence, 50139 Firenze, Italy

A. A. Rontogiannis Institute for Space Applications and Remote Sensing, National Observatory of Athens, 15236 Polaia, Pentli, Athens, Greece

S. Scalise Institute for Communications and Navigation, DLR, 82230 Wessling, Germany

M. A. Vazquez-Castro´ Departament de Telecommunicacio´ de Sistemes, Universitat Autonoma de Barcelona, 08193 Bellaterra, Spain

1. Introduction A panoramic view on the study and design of digital satellite communication links is the focus of this paper. The theoretical foundations of our universe of discourse are probability, statistics, electromagnetic field propagation, communications theory, and information theory, and although it is true that considerations on fundamental limits, bounds, criteria, algorithms, complexity, and implementation details are not dissimilar from those used for terrestrial digital communications, it is also undeniable that there are certain peculiarities of satellite links that make it essential to have specific tools and approaches. Two keywords help to describe in a nutshell our design objectives: efficiency,which should be intended both in terms of spectrum efficiency and in terms of power efficiency, and flexibility, which is an unavoidable necessity for designing physical layers which not only can adapt dynamically to time varying channel conditions, but that can also be used in generic and somewhat unspecified systems. In other words, for a given bandwidth the information rate must be maximized in conjunction with the minimization of the signal- to-noise ratio (SNR) required to achieve the specified quality of service, in harmony with the experienced channel conditions. In this tutorial paper we revisit all the essential elements that allow the engineer to tackle the design of a digital satellite communication link, including a characterization of the propagation channel, a review of the state-of-the-art in coding and modulation techniques, and an overview over the ancillary but fundamental blocks for parameter estimation, synchronization, distortion countermeasure, diversity, and fade mitigation. The paper 14 Short Tutorials also includes a discussion on the trends in multiuser communications and software radio. A good list of bibliographic references will serve the reader as a guide to further exploration.

2. Propagation channel A good understanding of the impairment introduced by the propagation channel is of paramount importance in the design of a satellite-based communications system. The relevant sources of channel impairment depend upon several factors, among them the considered scenario (e.g., mobile versus fixed terminals), the carrier frequency, and the employed access scheme. For instance, effects like shadowing and blockade are typical of mobile reception and do not apply to fixed reception. On the other hand, atmospheric attenuation phenomena, can be safely disregarded at L- (1-2 GHz) and S-Band (2–4 GHz) frequencies, whereas they start to become relevant at higher frequency bands such as Ku (10–12 GHz), Ka (20–30 GHz) and above. Finally, the contribution of interference to achievable signal- to-noise-plus. Interference ratio (SNIR) will not be forgotten. The modelling of satellite channels can follow either a deterministic or a statistical approach. Deterministic channel models provide the knowledge of the channel impulse response or transfer function with a very high accuracy, at the expense of a large computational complexity. Although in recent times algorithms have been proposed that reduce the model complexity, their application to satellite systems is still practically unfeasible, due to the vast area covered by a single satellite beam. On the other hand, statistical channel models, represent the relevant propagation phenomena, that is, distance-dependent attenuation, diffraction, absorption, and scatter- ing, by means of suitable statistical distributions. In general, the received signal contains a direct component, and a number of multiple paths, which are usually identified as the diffuse component. Fading effects, that is, variations of the signal amplitude and phase with the time, are typically divided into long-term fading and short-term fading. The long-term fading, commonly referred to as shadowing, models the attenuation caused by absorption and diffraction of the orography and large obstacles, such as hills, buildings, trees, and so forth. In order to measure a significant power level variation due to long-term fading, the mobile terminal must typically travel through several hundreds of wavelengths. No phase rotation is usually associated to shadowing. The short-term fading models variations in the signal due to constructive and destructive interference in the sum of multiple rays, mainly caused by reflections over surrounding surfaces. In this case, power fluctuations are mea- surable over distances comparable to the signal wavelength. Since statistical models based on a single distribution apply to wide-sense stationary (WSS) fading conditions, they are thus unable to properly model nonstationary transitions, for example, due to the terminal moving from urban to suburban environment, or to a change in elevation angle for a non- geostationary link. In order to be able to describe these variations in the statistical nature of the channel, multistate statistical models have been introduced. Assuming in general that an underlying Markov process determines the channel state transitions, each state in the Markov chain is described by a single-distribution model. When the signal bandwidth Bs is larger than the coherence bandwidth Bc of the channel, diffuse components may have a significant delay with respect to the direct component. In this case time dispersion associated to the presence of diffuse components will be taken into Digital Satellite Communication Techniques: A Review 15 account. By discretizing the diffuse components, the channel impulse response can be written as a sum of several impulsive components, each characterized by a certain amplitude, delay (with respect to direct ray), frequency shift, and phase rotation. In addition to the effects of mobility, satellite signals at frequencies above 10 GHz are perturbed when passing through the troposphere due to various phenomena. Four kinds of effects have to be considered: attenuation caused by atmospheric gases, clouds, and pre- cipitation, scintillation, depolarization, and increase of the antenna temperature in the receiving Earth stations (ESs). In turn, ionospheric effects can be considered as negligible for these frequency bands. Among the relevant tropospheric effects, rain attenuation and scintillation are the most remarkable ones. Rain specific attenuation (in dB/km) increases with the quantity of water and can be determined from the rainfall rate R (in mm/h). At frequency bands below Ku, this parameter is sufficient to calculate rain attenuation because all droplets are smaller than the wavelength. On the contrary, for frequencies higher than 20 GHz, the drop size distribution can have an impact on attenuation: two different distributions are able to produce two different rain attenuations for the same rain rate. However, statistical information on drop size distribution is typically not available. Therefore, statistical models are used to characterize rain attenuation, with the rain rate as input parameter. Tropospheric scintillation is caused by small-scale refractive index inhomogeneities induced by atmospheric turbulence along the propagation path. In satellite links, the scintillation effects are mainly attributed to strong turbulence in clouds; they are higher for very cloudy sky conditions, in particular in the presence of cumulus clouds, and in summer, around noon. Nonetheless, the scintillation effects may also appear in clear sky and in rainy conditions.

3. Coding in satellite communications One of the main countermeasures in the hands of the designer to counteract satellite channel impairment is forward error correction (FEC). The first practical application of FEC was in deep-space communications. In fact, in the mid-1960’s, NASA engineers understood that deep-space communications and coding is a “marriage made in heaven” [1]. Since then, channel coding techniques have been developed not only in relation to deep space communications, but also for commercial applications, where FEC is instrumental to ensure the required quality of services. However, unlike applications to deep space communications, FEC techniques for commercial services, in particular for mobile satellite systems, must deal with constraints such as limited receiver computational capabilities, power consumption limitation, and implementation cost. In this framework, different standardization bodies have opted for different FEC techniques, with the aim of striking a good tradeoff between performance and impact on the receiver terminal. The ETSI standard for UMTS and S-UMTS [2] recommends a turbo encoder with two 8-states recursive systematic convolutional (RSC) components and coding rate of 1/3. A rate of 1/2 is also available by appropriate puncturing of the parity bits. The related transfer function is given by 3 = g1(D) = 1+D + D G(D) 1, 1, 2 3 , (1) g0(D) 1+D + D 16 Short Tutorials

where g1(D) denotes the feedforward polynomial and g0(D) the feedback polynomial,bothin octal form of the constituent RCS code components. The DVB-RCS standard [3], adopted by ETSI, recommends for the return channel an 8-states duo-binary turbo code. The need of puncturing is less crucial in case of duo-binary turbo codes. This is because the RSC encoder components of rate 2/3 allow less redundant symbols to be discarded, in contrast to classical turbo codes with RSC encoder components of rate 1/2. In this case, it is considered that in satellite broadcasting applications the small ATM packets (ranging from 12 to 216 bytes) require, apart from a very low frame error rate (around 10−7 and less), a trellis termination of the RSC encoder components without the need of tail bits, if possible, so as not to waste extra bandwidth on flushing bits. DVB- RCS has solved this difficulty via a technique called circular trellis or tail biting [4], that is, the encoder retrieves the initial state at the end of the encoding process, such that the decoder can be initialized to any state and finish to this state in a circular way. The other main characteristic of this code is that DVB-RCS has proposed to realize the internal interleaving in two steps. The first one performs intrasymbol permutations, which increases the minimum distance of the code. The second interleaving is as for classical binary turbo codes and performs intersymbol permutations, so as to reduce the correlation of symbols during the iterative decoding process. The major disadvantage is that the 8-states duo-binary turbo decoder is around 30% more complex than the 16-states binary variant. However, if the comparison is per bit, then the duo-binary turbo decoder is simpler by 35%. The latest FEC implementation for satellite commercial applications regards the choice of LDPC codes for DVB-S2 [5]. LDPC codes are in fact able to provide performance comparable to or, in certain cases, better than turbo codes, while maintaining a very flexible and high-speed decoder implementations. Interestingly, also the Consultative Committee for Space Data Systems (CCSDS) has been investigating the use of LDPC codes in both deep-space and near-earth high data-rate missions [6] since 2001. In the DVB-S2 standard the FEC encoding is performed in three steps: an outer systematic BCH encoding, an inner LDPC encoding, and a block bit-interleaving to accommodate four possible constellations: QPSK, 8-PSK, 16-APSK, 32-APSK. The FEC scheme produces 64 800 or 16 200 bit-long codewords with eleven possible rate ranging from 1/4to9/10. The combination of the different code rates and the different constellations makes it possible to optimize spectral efficiency as transmission conditions evolve. DVB-S2 LDPC code belongs to the extended irregular repeat-accumulate code family, where the sparse component of the parity check matrix is quasicyclic, which makes the encoding process very simple.

4. Modulation schemes Digital modulation is a process which transforms a digital symbol into a modulated signal suitable for transmission over a communication channel. According to this modulation process the information is carried by a sinusoidal signal s(t) = a(t)sin 2πf(t)t + φ(t) (2) and by modifying one or more of three parameters of s(t): amplitude a(t), frequency f (t), and phase φ(t). By encoding the information in one of these parameters, keeping the other Digital Satellite Communication Techniques: A Review 17 two unchanged, three basic modulation schemes exist [7]. (i) Amplitude shift keying (ASK): the digital symbol is encoded in the amplitude a(t)of the signal. A simple example of ASK is the on/off modulation with digital 1 and 0 represented by signal with amplitude A and no signal, respectively. (ii) Frequency shift keying (FSK): the digital symbol is encoded in the frequency f (t). The signal with carrier frequency f1 represents digital 1 and the signal with carrier frequency f2 represents digital 0. A simple FSK modulator consists of a switch and two oscillators with frequencies f1 and f2. (iii) Phase shift keying (PSK): the digital symbol is encoded in the phase changes, φ(t), of the modulated signal. Advanced modulation schemes can be derived by combining basic modulation schemes to comply with transmission requirements, for example, higher bandwidth efficiency. Many combinations and variations of basic modulation techniques exist. Some modulation schemes can be derived from others and some can be represented as successors of the basic modulation types. For satellite communication, the modulation schemes can be appropriately classified according to their sensitivity on nonlinear distortion into two generic group of signals, namely, (i) constant and (ii) variable envelope modulation schemes. The variable envelope modulation schemes include two basic types: phase and amplitude modulated signals. The PSK modulation schemes, for example, binary phase shift keying (BPSK), quadrature phase shift keying (QPSK), 8-PSK, and 16-PSK, have constant envelope without baseband filtering. However, for all practical applications filtering is being employed, and thus they become variable modulation schemes [7]. The multilevel quadrature amplitude modulation (QAM) is a combination of amplitude and phase basic modulations. The in (I)andquadrature(Q) channels of QAM signal can be superimposed either coherently, that is, without delay between components, or with a delay of one half of the symbol interval between the components, which is known as an offset or staggered QAM. For satellite systems, special attention should be paid to the QPSK signal and its derivations. It can be represented as a “4-QAM” or “4-PSK” signal. If the Q-channel of QAM signal is delayed by half of the symbol interval, regarding to the I-channel, the orthogonal quadrature phase shift keying (OQPSK) is obtained. Further- more, if I-andQ-channels are shaped by sine function, the minimum shift keying (MSK) modulation scheme is obtained. The MSK signal, which is a constant envelope modulation scheme, can be also represented as an FSK signal with continuous phase and modulation index h = 1/2[8]. Assuring the smooth phase transition from symbol interval to symbol interval significantly decreases the amount of spectrum outside the main spectrum lobe. This type of modulation is referred to as continuous phase modulation (CPM). By choosing different pulse shape, varying modulation index h and alphabet size M, a great variety of CPM schemes can be obtained [8]. One of the most commonly used criteria for evaluating the performance of modulation schemes is the average bit error probability (ABEP). Assuming ideal channel state information, additive white Gaussian noise (AWGN) channel, and perfect synchronization, after 18 Short Tutorials

Nyquist filtering and ideal sampling, the received kth sample is yk = gk + nk where gk and nk are the received symbol and noise samples, respectively. They are in general complex numbers, represented as gk = gI,k + jgQ,k and nk = nI,k + jnQ,k. In most cases the ABEP can be obtained in closed-form in terms of the Gaussian Q function, [9, equation (26.2.3)]. On the contrary, for arbitrary modulation schemes, the derivation of a closed-form expression for the ABEP is a very difficult task. As a result analytical bounding techniques are proved to be an insuperable engineering tool for the design and the performance evaluation of codes and modulation schemes [10]. Another important satellite channel for mobile communications is the fading channel [11]. Assuming again ideal channel state information and Rk to be the kth channel envelope of a flat fading channel, after filtering and sampling, the baseband received complex signal can be expressed as

yk = Rkgk + nk. (3)

N0 is a random variable (RV) such that the conditional symbol error probability Pes(s) is provided as RV as well. Following the approach based on the probability density function (pdf), the expected value of Pes(s) can be derived by averaging with respect to the pdf of s. 2 In case of fading channels and alleviating index k, γs = R Es/N0 is a random variable (RV) such that the conditional symbol error probability Pes(γs) is an RV as well. Following the approach based on the probability density function (pdf), the expected value of Pes(γs) can be derived by averaging with respect to the pdf of γs, that is, ∞ = | Pes γs Pes γs γ fγs (γ)dγ. (4) 0 An alternative and efficient method to study the average bit error probability (ASEP) is to use the moment-generating-function-(mgf-) based approach. By following this approach and using an available formula for the γs mgf of the fading channel under consideration, the ASEP can be derived for the most important fading channels including the Rayleigh, Nakagami-m [12], the Rice [13] and the Weibull [14]. As far as the Ricean distribution is concerned, it is often used to model propagation paths consisting of one strong direct line- of-sight (LoS) signal and many random reflected and usually weaker signals. Such fading environments are typically observed in microcellular and mobile satellite radio links [11]. It is worth mentioning that the optimal maximum-likelihood detector for Ricean fading channels has been derived for the first time in [15]andlaterin[16]. This work has shown for the first time that the optimal receiver structure for the Rician fading channels consists ofabankofmultipledifferential detectors (MDD). The application of the Weibull fading model to satellite communications has been investigated in [17].

5. Parameter estimation and synchronization To allow proper detection and decoding of the received signal, a number of parameters must be estimated [18], such as the sampling instants, the carrier (frequency/phase) reference, and the time epochs corresponding to the start of frames or code words. All these ancillary Digital Satellite Communication Techniques: A Review 19 tasks are commonly considered as synchronization issues, and must be especially designed to minimize their impact on the ﬁnal performance of the decoding. The expected signal- to-noise ratio (SNR) of the signal is of paramount importance for the design of the corresponding synchronization algorithms. This is especially relevant in systems with strong coding protection, such as DVB-S2 or S-UMTS, for which SNR values can be very low. Un- der these circumstances many synchronization algorithms work properly only under the guidance of known symbols, frequently referred to as pilot symbols. However, whenever possible, blind algorithms which are able to cope with the noise and the randomness of the underlying symbols are preferred to maximize the system throughput. Let us consider the following received baseband signal model to illustrate the required estimation tasks [19, 20]: j(θ0+2πf0t) r(t) = Ae ckg t − kT − 0T + n(t). (5) k

The symbols are denoted by {ck}, the received signalling pulse as g(t), and the noise is given by n(t). The extraction of the symbols needs the knowledge, up to a given error, at least of the following values.

(i) Symbol timing 0. Its estimation is known as timing recovery. (ii) Carrier frequency f0. This value is often estimated in two stages, ﬁrstly to get a coarse estimate leaving a residual error which will be corrected in a second stage. Both processes are jointly known as frequency recovery. (iii) Carrier phase θ0. The estimation of this value is referred to as phase recovery. (iv) Amplitude level A. One or more gain control stages are necessary to keep the amplitude of the signal at predetermined levels.

In some other applications, such as adaptive coding and modulation adopted by DVB-S2, 2 2 2 2 it is necessary to estimate also the SNR value, given by A E{|ck| }g(t) /E{|n(t)| }.For time division multiplexing systems where symbols are grouped in packets known as frames, it is also necessary to perform the frame epoch recovery; this process is known as frame synchronization. Similarly, spread spectrum systems need to establish the relative position of the spreading code.

6. Distortion countermeasures The satellite channel introduces linear distortion to the transmitted signal due to linear filtering, shadowing, and multipath propagation which is also typical of terrestrial communications. A peculiarity of satellite systems, is the need to maximally exploit on-board resources, that is, power, by driving the high power-amplifier (HPA) at or near its saturation point, thus introducing nonlinear distortion on the satellite link. Indeed, this is one of the main sources of performance degradation in satellite communication systems, which prevent the use of bandwidth efficient high-order modulation schemes. To study the effect of the HPA on the overall system performance and design appropriate distortion countermeasure techniques, HPA operation is frequently modeled by the so-called Saleh model [21], 20 Short Tutorials whose amplitude-to-amplitude conversion and amplitude-to-phase conversion characteristics are given by the following expressions: α ρ2 = αaρ Φ = p A(ρ) 2 , (ρ) 2 , (6) βaρ +1 βpρ +1 where αa, αp, βa, βp are constant coefficients, and subscripts a and p denote amplitude and phase. Countermeasures for nonlinear distortion are usually divided into predistortion and equalization. Predistortion refers to preprocessing of the signal entering the HPA, so that the response of the overall system is linear. On the other hand, equalization counteracts the distortion introduced by the channel, by processing the signal at the receiver side. The operation of the ideal predistorter is described by the following generic relation: y = H P(x) = cx, (7) where x is the transmitted signal, P(·) represents the predistorion function, H(·)accounts for the overall channel distortion, and c is a constant gain. A commonly used approach for predistortion is the so-called look-up-table (LUT) method [22]. Rather than computing in real-time the inverse characteristic of the channel, which requires a considerable amount of computational power, a number of coefficients can be computed offline and stored into a LUT. Another approach is to select a particular nonlinear model for the channel and attempt to design the predistorter directly as the inverse of this model [23]. In general, this method is difficult to implement and results in an approximation of the inverse system, which may be unacceptable in practice. Alternatively, a nonlinear model can be assumed for the predistorter itself: after a training phase the model can be adjusted to mimic the inverse operation of the channel. A number of nonlinear models can be employed for this purpose, for example, the Volterra model [24] and the neural network (NN) model [25]. Equalization techniques for nonlinear satellite channels have been studied in open scientific literature. The optimum ML receiver for a nonlinear bandlimited satellite channel under a general formulation is presented in [26]. However, this approach cannot be applied in practice due to its high computational complexity and the need of channel state information at the receiver. Suboptimal reduced computational complexity implementations have been designed based on a specific nonlinear model, for example, the Volterra [27] or the NN [28] model. Nonlinear equalization methods, which do not require a training sequence and thus offer effective use of the available bandwidth, have been presented (blind equalization methods), for example, [29, 30]. Another very promising technique already applied for linear channels, and that promises for significant gains also in nonlinear satellite channels, is turbo equalization [31]. Finally, predistortion techniques can also be applied in conjunction with precoding instead of equalization, as shown in [32].

7. Diversity and fade mitigation techniques As described in Section 2, satellite communications are characterized by time varying propagation conditions, due to either mobility of the terminal or change in atmospheric Digital Satellite Communication Techniques: A Review 21 conditions, which lead the satellite communication system to operate in deep fading conditions (e.g., 10–20 dB below the average signal level). Various methods exist to counteract these effects among which the most interesting for satellite communications are satellite, site, frequency and polarization diversity, and adaptive modulation and coding (ACM) techniques. Satellite diversity provides coverage of the same area from more than a single satellite. By means of satellite diversity, a user can potentially exploit different satellites inside its field of view in order to reduce the probability of paths to the satellites being blocked by natural or artificial obstacles. Although this is the main motivation for satellite diversity, there is also an impact of satellite diversity on system capacity, which cannot be neglected in the evaluation of the advantages and disadvantages of this diversity technique [33]. The site diversity rationale is that the link availability increases because the joint outage probability of two stations located at a certain distance is much lower than the single site case. As main drawbacks, the station must be duplicated (at least a part of the equipment) and a dedicated terrestrial link must be set up. This concept can be implemented with a more sophisticated and evolutionary architecture including a set of cooperating stations, working separately with the same satellite [34]. In this case if one of the stations needs assistance, its traffic flow can be rerouted to one of the others, selected on the basis of proper algorithms. Frequency diversity is based on a common resource shared among the Earth stations, utilized for those links which need assistance because of the local high value of rain fading. A frequency slot represents the common resource, of course allocated at a lower range than the slot utilized in normal conditions. In fact, with frequency diversity we use a higher band to carry most of the traffic and a lower band (typically much smaller than the former) to assist those links in the time intervals during which the rain attenuation overcomes a certain threshold. As a matter of fact the link at the higher frequency is dimensioned with a greater outage probability, allowing a smaller rain margin. When the rain attenuation overcomes the rain margin, the link will work at the lower frequency. Polarization diversity refers to the use of two orthogonally polarized receiving antennas. In most cases, this is the simplest way to realize the diversity concept. In contrast to space diversity, the antennas need not to be multiplied. In the case of a reflector antenna the primary feed only must be more complex than its nondiversity counterpart. Dual-polarized radiators must be used in array or patch antennas. An economic advantage is that in either case antennas are virtually co-located, making the whole setup cheaper than with two distinct antennas. In contrast to frequency diversity, polarization diversity does not need a wider frequency band than a nondiversity system. On the other hand, the number of diversity routes is definitely two—in contrast to space or frequency diversity in which any number of routes are in principle available. Moreover, polarization diversity is slightly less effective than its counterparts. In conventional communication systems, to overcome deep attenuation effects and minimize the outage probability, or conversely, to guarantee a reliable transmission for a large percentage of the time, the communication link is designed according to the worst channel conditions, thus wasting power or capacity resources when good propagation conditions are experienced, for example, transmission in clear sky. ACM techniques can 22 Short Tutorials provide significant capacity gains by allowing the optimization of transmission configuration, that is, modulation order and channel code rates, according to the measured link SNIR. Interestingly, ACM has been adopted by the DVB-S2 standard [5]. Finally, in the recent years, multiple-input-multiple-output (MIMO) is attracting the interest of the satellite communication research community as a key technology for the next generation of satellite communications systems. The MIMO approach is an extension of the space diversity technique, exploiting the diversity which comes forth from multiple and uncorrelated channel paths from the transmitter and the receiver. To exploit the potentially large capacity of MIMO in fading channels, a considerable amount of research has been focused on space time coding (STC) which plays on the domain of both antenna and time diversity [35]. The most diffused families of STC techniques are based on (STBC space time block code) [36] or (STTC space time trellis code) [37].

8. Multiuser satellite communications As any other wireless communication system, satellite communication systems face the problem of sharing the system resources among different users and services. Current wireless communications systems have been developed driven by the classical information theory that follows a point-to-point philosophy. Point-to-point communications occur between a single transmitter and a single receiver. Multiuser communications occur among fixed and/or mobile users. Multiuser communications in a satellite scenario usually refer to a star topology consisting of one or more gateways and a large number of small user terminals. Gateways communicate with the user terminals through the forward link and terminals may communicate with the gateways through the return link, both links having a multiuser uplink channel and a multiuser downlink channel. Broadly speaking multiuser communications may refer either to a one-way communication as radio or TV broadcasting or to a two-way multiuser communication where each user is interested in a specific message. In general, multiuser communications usually refer to the latter case. A multiuser system achieves a specific point in the capacity region depending on how the multiuser channel is shared by the users, which depends on the multiple access technique or medium-access control (MAC) protocol used. MAC protocols are designed to coordinate multiuser transmissions as well as eventual retransmissions or resolution of collisions. Note that in general the uplink multiuser channel is usually called multiple-access channel while the multiple access for the downlink channel is usually referred to as multiplexing.

8.1. MAC techniques for satellite systems Satellite systems pose some major constraints on a MAC protocol performance such as long propagation delay, remote control of on-board processing capabilities, or power limitation that precludes the use of some protocols developed for a terrestrial scenario. For satellite networks, multiple access occurs in the uplink of the return link (return path) only. A classification according to the most efficiently supported types of traffic follows: connection- oriented multiple access, which establishes a dedicated channel (or circuit) for the duration of a transmission. In general, dedicated radio resources are orthogonalor non-orthogonal Digital Satellite Communication Techniques: A Review 23 along available axes: time, frequency, and/or code. This solution is appropriate for communications that require data to be transmitted in real-time. This type of communication is different to both contention oriented communications and packet switching, which divides messages into packets and sends each packet individually. Packet switching is more efficient for the so-called elastic traffic for which some amount of delay is acceptable. IP trafficis connectionless and packet oriented, and in a wireline network it is transmitted through packet switching. However, IP traffic is sent over a satellite link after having established a connection in order to access the satellite system air-interface. Examples are frequency division multiple access (FDMA), time division multiple access (TDMA), code division multiple access (CDMA), direct sequence CDMA (DS-CDMA), frequency hopping CDMA (FH-CDMA), space division multiple access (SDMA), orthogonal frequency division multiple access (OFDMA), multifrequency-time division multiple access (MF-TDMA), or ultra- wideband (UWB) contention-oriented multiple access, by which terminals access the common shared channel without any previous assignment. Random access techniques are usually employed in networks where most sources are bursty since they are easy to implement and adapt well to traffic fluctuations. However, collisions may result in a waste of capacity. The most representative random multiple access protocols for satellite communications are Random access techniques, based on the ALOHA protocol such as narrowband ALOHA or spread ALOHA [38], and contention-based reservation, based on a channel reservation, made via a subchannel to avoid collision. demand assignment multiple access (DAMA),by which the network allocates system resources on a demand assigned basis. With DAMA, satellite capacity is dynamically assigned to capacity requests coming from users and therefore capacity is made available when needed. DAMA multiple access is also referred to as bandwidth on demand (BoD). It is possible to allow distributed control, in which capacity requests are broadcasted so as each station decides on whether to demand capacity or not. The two most commonly DAMA techniques used in satellite communications systems are packet reservation media access (PRMA) and combined free-demand assignment multiple access (CF-DAMA).

8.2. Multiuser capacity From a theoretic information point of view, it is possible to find the limits of achievable sets of rates of a multiuser system that any resource management strategy is able to obtain. Such limits are different for the uplink and the downlink multiuser channel. Both channels can be seen as a dual of the other. The downlink channel can also be referred to as broadcast channel (BC). The BC capacity is also known for the degraded version of the channel, that is, when the received powers can be ordered. In general, it is assumed that power can be adapted to channel conditions. For satellite systems however, it is common to assume constant power given the limitations imposed by the on-board amplifier. Adaptation however has been assumed for (space division multiplexing SDM) satellite communications, also called multibeam systems. In a multibeam satellite system, adaptivity is enabled at transmission level by allowing a range of feasible spectral efficiencies (corresponding to different combinations of modulation formats and coding protection levels). The uplink channel can also be referred to as multiple-access channel (MAC) and capacity regions can be defined and obtained as shown in [39, 40]. 24 Short Tutorials

8.3. Capacity enhancement techniques Multiuser detection (MUD) is an effective technique for counteracting cochannel interference (CCI) effects in time-division multiple-access (TDMA) systems and multiple-access interference (MAI) in code-division multiple access (CDMA) systems. Multiuser detection has been proved to yield good performance in CDMA code sectored multibeam networks and heavy interference situations. The enhanced single-user detection for DS-CDMA downlink schemes consist of chip level multipath equalizers, multipath interference can- cellers, and symbol level adaptive interference mitigating filters. The chip level equalizers operate on the assumption that if orthogonal Walsh-Hadamard codes are used in the downlink then the only source of interference comes from multipath propagation that destroy the orthogonality of the codes, hence if equalizers are used to restore the orthogonality of the codes, interference can be reduced. Another technique called turbo MUD gets inspira- tion from the “Turbo Principle” of the exchange of soft information between the multiuser detector and the channel decoders iteratively for joint multiuser detection and channel decoding. Multiuser communications based on satellite systems is a currently active research topic.

9. Software radio Software radio (SR), sometimes also referred to as software-defined radio (SDR), is the emerging technology paradigm on many telecommunication areas. In the presence of many radio standards the ability software-based (re)-configuration of the physical layer will produce practical benefits also for the satellite context. This section will explore the main innovations that support the adoption of software radio for both earth terminals and on-board space processing.

9.1. Antenna, analog processing, and A/D conversion The general wireless transceiver architecture can be divided into two main parts: the digital back-end and the analog front-end. These parts are divided by the domain converters. Generally we can ﬁnd an analog front-end in an SDR with a bandwidth equal to the Nyquist bandwidth of the converters. This is the optimum as converters can neither generate nor process any wider bandwidth and the analog front-end does not limit the capabilities of the converters. If the system channels are narrower than this bandwidth, further channelization (i.e., up/downconversion and sample rate conversion) is done in the digital domain. This is a typical IF digitized SDR architecture.

9.2. High-speed ADC technologies It has been proven that ADCs are more critical devices than DACs and their performance can be a real bottleneck in the system. Today’s conversion technology used for wireless RF/IF applications uses conventional purely electronic devices. State-of-art ADC sampling rates reach to hundreds Ms/s and even up to a few Gs/s providing 10–14 bit resolution. These converters are prevalently ﬂash, pipelined or folding & interpolating designs [41]. Digital Satellite Communication Techniques: A Review 25

It was demonstrated that optical pulses from a mode-locked laser source exhibit far less jitter when compared to electrical clock sources. There are efforts to build hybrid (optic- electronic) devices and fully optical ADCs. Hybrid devices use the optical pulses for sampling the input voltage and subsequently quantize it by electronic means. The samples are converted into the electrical domain and distributed among many slow (in terms of the sampling frequency) ADCs in some interleaved manner. The interleaving can be done in the time domain or in the frequency domain by using standard wavelength-division techniques. 9.3. Optimized computational structures In high-speed matched filtering, where multipliers are to be avoided at any cost, a kind of noise shaping could be performed [42]. The basic idea is to use the ΣΔ principle to reduce the resolution of the sampled signal to, say, 4 bits, such that the SNR remains very high in the band of interest and shrinks dramatically out of this band. The overall SNR coincides with the theoretically achievable SNR of 4 bit but, due to noise shaping, it does not show a white distribution. The same idea can be applied to matched filters in the transmitter path. The data source for this filter is the symbol mapper. The resolution of the data flow depends on the modulation scheme. Another simplification may be exploited if the impulse response exhibits an even sym- metry like raised-cosines, which half the number of LUT’s, which holds the precomputed filter weights. DSPs are usually high-speed Harvard architecture processors capable of doing various fixed-point and/or floating-point arithmetic operations very efficiently, integrated with on- chip I/O-controller, direct memory access (DMA) interface and memory. DSPs offer some additional flexibility in the sense that it can dynamically balance the processing power between different tasks. It can, for instance, afford a better channel equalizer if working under difficult conditions with reduced functionality in other stages. Performance comparison with FPGAs and even among different DSP families is not obvious. The highest degree of flexibility would be obtained if both the ADC and the DAC could be placed as near to the antenna as possible. In this case, only analog components, such as amplifiers and antialiasing filters, are needed. It is to be noticed, however, that satellite communications systems use carrier frequencies in the GHz range, which requires very fast AD/DA conversion. Super-Nyquist sampling is then out of scope for state-of-the-art converters, not at least, because also part of the data processing has to run at this speed. According to Figure 1, the units on the right-hand of the AD/DA conversion are denoted as hard and soft real-time signal processing. The definition of hard real-time signal processing is such that the results are generated every rising or falling edge of the sampling clock; data are read from or written to a FIFO buffer to be processed by subsequent soft realtime processing. In this context, soft real-time processing means that the calculations are finished on the average within a clock cycle without occurrence of buffer underflows or overflows. 9.4. Software radio onboard With the evolution of communication systems towards multimedia services, the required data-rates, bandwidth, and QoS get higher. In this scenario, satellites can play a major role 26 Short Tutorials

Figure 1: Software radio communications system. as they offer the unique advantages of a wide area coverage, also in areas where terrestrial infrastructures are not economically effective, and a rapid deployment. Payloads with on- board processing are particularly effective for multimedia services as they improve the link budget and decrease the transport delay when on-board packet switching is achieved. The lifetime of a communication satellite along with high deployment costs make the transparent payload solution still attractive, since all smart functions are located on the earth terminals. As the satellite lifetime increases (15 years typical for a geostationary satellite) and communication standards and services evolve faster, the technological requirements for new multimedia services cannot be predicted accurately at the payload design time. Conventional regenerative payloads provide a poor level of flexibility because of a large use of ASIC technology for the digital hardware equipment. The potential solution is to provide the payload with a reconfigurable architecture that, starting from a conventional fully transparent behavior, can host more sophisticated functions during the satellite lifetime. The space-qualified processing technologies that enable the realization of SDR-based equipment in space segment are mainly three: FPGA, DSP, and array processors.

9.5. Service and technologies in next generation terminals Future satellite applications will aim to the distribution of many types of traffic with different quality, security requirements, and transmission techniques. Starting from the upcoming DVB-S2 standard, new concepts of signal modulation adaptivity has been inserted in the space paradigm. As described in Section 3, ACM can improve dramatically the performance of a forward link satellite system. The target of the software radio technology is the implementation of a multimode and multibandwidth telecommunication system with characteristics defined by software over all protocol layers. This means a multimode radio with software dynamic characteristics defined in all protocol stack layers, included the physical one. It allows a large flexibility in terms of the particular coding, modulation, and multiple = access techniques adopted. Moreover, it grants a long system life through remote updating and reconfiguration. This issue is of key importance when the advanced transmission and reception features are located on earth terminals (transparent and semitrans- parent satellite payloads). Remote configuration capability is a relevant element enabled by SR technology. The configuration process is obtained by delivering to the remote SDR device a set of configuration commands able to properly define the physical layer of the remote device. A specific on-board processor controller is required to manage the remote configuration process. Digital Satellite Communication Techniques: A Review 27

10. Conclusions In concluding this tutorial paper, we would like to express our ﬁrm belief that the solutions and algorithms that we are able to design today are but the beginning of future evolutions, that will have to be surpassed by new ideas and innovations that we are completely unable to see today, and this is the beauty and the very reason for dwelling deep into scientiﬁc research.

Acknowledgments This work has been supported by the IST-SatNEx projects. The authors would like to thank SatNEx Partners for fruitful discussions and comments.

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[34] M. Luglio, R. Mancini, C. Riva, A. Paraboni, and F. Barbaliscia, “Large-scale site diversity for satellite communication networks,” International Journal of Satellite Communications, vol. 20, no. 4, pp. 251–260, 2002. [35] A. Naguib, N. Seshadri, and A. Calderbank, “Space-time coding and signal processing for high data rate wireless communications,” IEEE Signal Processing Magazine, pp. 76–92, 2000. [36] V. Tarokh, H. Jafarkhani, and A. R. Calderbank, “Space-time block codes from orthogonal designs,” IEEE Transactions on Information Theory, vol. 45, no. 5, pp. 1456–1467, 1999. [37] V. Tarokh, N. Seshadri, and A. R. Calderbank, “Space-time codes for high data rate wireless communication: performance criterion and code construction,” IEEE Transactions on Informa- tion Theory, vol. 44, no. 2, pp. 744–765, 1998. [38]D.B.ZelekeandM.A.Vazquez-Castro,´ “ALOHA versus single code spread ALOHA for satellite systems,” in Proceedings of 61st IEEE Vehicular Technology Conference (VTC ’05), vol. 4, pp. 2692– 2696, Stockholm, Sweden, May-June 2005. [39] A. Goldsmith, Wireless Communications, Cambridge University Press, Cambridge, UK, 2005. [40] D. Tse and P. Viswanath, Fundamentals of Wireless Communication, Cambridge University Press, Cambridge, UK, 2005. [41] P. E. Pace, Advanced Techniques for Digital Receivers, Artech House, Norwood, Mass, USA, 2000. [42] C. Dick and F. J. Harris, “Conﬁgurable logic for digital communications: some signal processing perspectives,” IEEE Communications Magazine, vol. 37, no. 8, pp. 107–111, 1999.

A. Vanelli-Coralli joined the University of Bologna in 1996 and he is currently an Assistant Profes- sor. During 2003 and 2005, he was a Visiting Scientist at Qualcomm Inc. (San Diego, Calif). He is responsible for the R&D Group of the Advanced Satellite Mobile Systems Task Force (ASMS-TF) and Vice-Chair of the R&D Group of the Integral Satcom Initiative. He has been Editor of several Special Issues on satellite communications published in international journals. He is corecipient of the Best Paper Award at IEEE ICT 2001 and at IEEE ISWCS 2005.

G. E. Corazza is a Full Professor at the University of Bologna, where he is responsible for wireless communications inside the Advanced Research Centre for Electronic Systems (ARCES). He is the Chair- man of the Integral Satcom Initiative (ISI) and of the Advanced Satellite Mobile Systems Task Force (ASMS TF), European fora on satellite communications under the auspices of the European Commis- sion and of the European Space Agency. He is Editor for Spread Spectrum for the IEEE Transactions on Communications.

P. Bi t h a s received the Diploma in electrical and computer engineering from the university of Patras, Greece, in 2003, and continues his studies towards the Ph.D. degree at the same University. Since his graduation, he joined as a Research Associate the Wireless Communication Research Group at the Institute for Space Applications and Remote Sensing (ISARS), National Observatory of Athens (NOA), Greece. Since 2004 he has been participating in SatNEx Network of Excellence.His research interests are digital communications over fading channels, diversity techniques, and mobile radio communications.

S. Cioni received the Dr.Ing. degree in telecommunication engineering and the Ph.D. from University of Bologna, Italy, in 1998 and in 2002, respectively. From March 2002 to October 2002, he was a Visit- ing Researcher at the European Space Agency (ESA) on adaptive coding and modulation (ACM) techniques for future broadband satellite networks. In 2002, he joined the ARCES, where he is currently a Post-Doc researcher. His interests include synchronization techniques, medium-access control 30 Short Tutorials resource allocation algorithms, OFDM systems, and iterative decoding techniques joint to channel parameter estimation.

L. Castanet works as Research Engineer in the radioelectrical propagation ﬁeld at ONERA, Toulouse. His main research interests are earth-space propagation (more particularly the dynamics and spatial variability of the propagation channel) and fade mitigation techniques (design and performance simulation). He has been a French expert in the COST 255 and COST 280 European projects, in which he has worked on the development and validation of advanced propagation models of the spatiotemporal variability of the propagation channel and on the design and optimization of adaptive fade mitigation techniques. At the moment, he is involved in the European Network of Excellence “SatNEx,” in which he Chairs the propagation joint action, and participate to the physical layer and access joint actions. He is also a French Representative in ITU-R Study Group 3 which deals with radiowave propagation.

W. Gappmair studied communications engineering and computer science at Graz University of Tech- nology, Austria, where he holds currently the position of a Staﬀ Scientist at the Institute of Commu- nication Networks and Satellite Communications. His main interests and research activities include information theory, channel coding, digital modulation, parameter estimation, and synchronization.

M. Luglio received the Laurea degree in electronic engineering in 1990 and the Ph.D. degree in telecommunications in 1994. From August to December 1992 he worked at Comsat Labs. From 1995 to 2004 he was a Researcher at the University of Rome “Tor Vergata” where he is now an Associate Professor. He works on designing satellite systems for multimedia services. In 2001 and 2002 he was a Visiting Professor at the University of California, Los Angeles. He teaches satellite telecommunications and signals & transmission.

P. T. Mathiopolous is currently the Director of Research at the Institute for Space Applications and Remote Sensing (ISARS) of the National Observatory of Athens (NOA), where is has established the Wireless Communications Research Group. Prior to that he was a Professor of electrical and computer engineering at the University of British Columbia in Canada, where he is now an Adjunct Professor. Over the years, he has supervised university- and industry- based R&D groups and has successfully acted as Technical Manager for large R&D Canadian and European projects. Since 1989, under his supervision, more then 30 graduate students have graduated in Canada and Greece. He has published about 150 papers in journals and international conference proceedings and is on the editorial board is of many scientiﬁc journals. He has regularly acted as a consultant for several governmental and private organizations, including the European Commission. He has delivered numerous invited presentations, including plenary lectures, and has taught many short courses all over the world.

C. Mosquera received the Undergraduate degree from the Universidad de Vigo (1992), the M.S. degree from Stanford University (1994), and the Ph.D. degree from the Universidad de Vigo (1998), all in electrical engineering. In 1999 he spent six months with the European Space Agency at ESTEC in the Netherlands. He is currently an Associate Professor at the Universidad de Vigo. His interests lie in the area of signal processing applied to communications.

L. S. Ronga received his M.S. degree in electronic engineering in 1994 and his Ph.D. degree in telecommunications in 1998 from the University of Florence, Italy. In 1997 he joined the International Com- puter Science Institute of Berkeley, California, as a Visiting Scientist. In 1998 he obtained a post-doc position in the Engineering Faculty of the University of Florence. In 1999 he joined the Italian Na- tional Consortium for Telecommunications, where he is currently the Head of Research. His interests Digital Satellite Communication Techniques: A Review 31 focus on transmission aspects for terrestrial and satellite communications, quality of service, and cross-layer aspects for wireless networks. He has been involved in COST 252 and 272 European Com- munity actions and in SatNEx Network of Excellence. He also directed several research groups for national projects and he is author of scientiﬁc papers in conferences and international journals. He is currently the editor of the EURASIP Newsletter.

A. A. Rontogiannis is a Researcher on wireless communications at the Institute for Space Applications and Remote Sensing of the National Observatory of Athens, since 2003. From 1999 to 2003 he was a Lecturer of informatics at the University of Ioannina. He holds a Ph.D. degree from the University of Athens, an M.A.S. from the University of Victoria, Canada, and a Diploma degree in electrical engineering from the National Technical University of Athens. His research interests are in the general areas of signal processing and wireless communications.

S. Scalise graduated in electronic engineering specializing in telecommunications (with honors) from the University of Ferrara, Italy, in July 1999. Since 2001, he is within the Institute for Communica- tions and Navigation, DLR, German Aerospace Center, Germany. Since October 2004, he is leading the Mobile Satellite Systems Group. His research activity deals with forward error correction and synchronization schemes for mobile satellite applications, land mobile satellite channel modelling, and link performance evaluation. He is the coauthor of many international journal and conference papers, and Chairman of the R&D Working Group of ISI (Integral SatCom Initiative) European Technology Platform.

M. A. Vazquez-Castro´ received M.S. degree in telecommunications engineering in 1994 and Ph.D. degree (cum Laude) in 1998 from University of Vigo. She has been an Assistant Professor at the Uni- versity Carlos III de Madrid, Guest Researcher at the University of Southern California and from 2002 to 2004 she was an internal Research Fellow at the European Space Agency, The Netherlands. Cur- rently she is an Associate Professor at the Universitat Autonoma´ de Barcelona leading the Wireless Communications Research Group. Her current areas of interest are satellite communications networks cross-layer design and optimization. SHORT TUTORIALS

Testbeds and Rapid Prototyping in Wireless System Design

Markus Rupp Institute of Communications and Radio Frequency Engineering, Faculty of Electrical Engineering and Information Technology, Vienna University of Technology, Gußhausstraße 25/389, 1040 Wien, Austria

Christian Mehlfuhrer¨ Institute of Communications and Radio Frequency Engineering, Faculty of Electrical Engineering and Information Technology, Vienna University of Technology, Gußhausstraße 25/389, 1040 Wien, Austria

Sebastian Caban Institute of Communications and Radio Frequency Engineering, Faculty of Electrical Engineering and Information Technology, Vienna University of Technology, Gußhausstraße 25/389, 1040 Wien, Austria

Robert Langwieser Institute of Communications and Radio Frequency Engineering, Faculty of Electrical Engineering and Information Technology, Vienna University of Technology, Gußhausstraße 25/389, 1040 Wien, Austria

Lukas W. Mayer Institute of Communications and Radio Frequency Engineering, Faculty of Electrical Engineering and Information Technology, Vienna University of Technology, Gußhausstraße 25/389, 1040 Wien, Austria

Arpad L. Scholtz Institute of Communications and Radio Frequency Engineering, Faculty of Electrical Engineering and Information Technology, Vienna University of Technology, Gußhausstraße 25/389, 1040 Wien, Austria

This tutorial paper gives an overview of requirements on testbeds and rapid prototyping suitable for MIMO transmissions. Testbeds support real-time transmissions over the air and thus allow for exper- imenting with true physical channels, including also an analog frontend. This makes the transmission process very realistic. On the other hand, rapid prototyping allows for sketching transmitter and receiver hardware architectures of future products. Thus, rapid prototyping is very close to the design of a final product, derisking its financial investment. Several experiment examples demonstrate how testbeds and prototypes can support system design significantly. Testbeds and Rapid Prototyping in Wireless System Design 33

1. Introduction Wireless products have become very popular due to low-cost devices and services. Commu- nicating in a wireless fashion is by far not simple, in particular when many users desire to communicate at the same time. The available spectrum is rather limited, making the design of a wireless multiaccess scheme very challenging. Note that in most countries the major- ity of the spectrum below 1 GHz is occupied by public TV broadcasting while only little is left for cellular and WLAN services. Although recently new bands at 3.5 GHz, 5.2 GHz and 11 GHz (in some countries also 1.5 GHz, 2.6 GHz, and 5.8 GHz) became available, moving to higher frequencies is not a good solution for most wireless connections because more transmit power is required to compensate for the much higher channel attenuation. Due to multipath propagation, the receivers become very complex, in particular when data is to be transmitted at high rates. Because of this high complexity in signal processing, the mobile communication devices suffer of low-battery life, making it a necessity to recharge the battery quite often. On top of all these problems, newer communication standards still require increased signal processing complexity, draining the battery power even more. Several studies [1, 2, 3] show that although Moore’s law [4] predicted in 1965 is still correct after 40 years, that is, the available complexity doubles every 16–18 months, the desired complexity follows Shannon’s law and is doubling even faster. While this was not so much of a problem in the past years since the available complexity was sufficient, it becomes a permanent problem nowadays, in particular when time-to-market aspects become crucial [5]. This so-called complexity gap is accompanied by another one called design gap or productivity gap. The studies in [1, 2] show that current improvements in productivity, that is, the designer’s capability to convert algorithms into silicon, is growing much slower than the available silicon complexity predicted by Moore’s law. The reason for such a lack in productivity lies on the poor tool support. After the definition of VHDL in the mideighties, industry took on tool support for design engineers and came up with many useful products. At these times, only very few universities were involved. Until the midnineties, the quality of the products improved only very little. While many tasks could be made automatically, the remaining tasks are rather difficult to solve and many research teams at universities are currently working on solutions. In particular, there are very few tools to support float-to-fixed conversion, that is, the conversion from floating point descriptions of high-level languages like Matlab or ANSI-C to a corresponding fixed-point representation. While graphical tools from Synopsys, CoWare, and Mathworks increase visibility, they support only manual partitioning in HW (hardware) and SW (software) modules. However, for complex algorithms, it may be much better to have an automatic tool support, taking possible hardware platforms and communication between the modules into account when deciding for partitioning. Often, specific HW platforms are already predefined and the task is to map an algorithm onto such a platform consisting of an embedded DSP with many HW accelerators, RAMs, ROMs, and busses. Also recently, embedded cores with power awareness were introduced [6], allowing idle algorithms to be switched off or to run on lowerclock rates to preserve energy. Tools for all of these important 34 Short Tutorials tasks are currently missing and manual solutions are time consuming, erroneous, and certainly suboptimal. This paper is organized as follows. In Section 2 the terminology in prototyping is defined distinguishing strictly between testbeds and prototypes. While Section 3 further ex- plains testbeds and provides many design examples utilizing a testbed; Section 4 provides the same details for prototypes. Here, in particular, the methods for rapid prototyping are focussed on. Examples on MIMO-WLAN and adaptive predistortion techniques are provided. Section 5 discusses further challenges in rapid prototyping, and Section 6 closes the paper with conclusions.

2. Prototyping Many years ago, prototyping was used to build a first (set of) working demonstrator(s) to prove that a new theory could really be applied and to learn how cumbersome and expensive it could once become turning it into a product. Real-time experiments with working product demonstrators were essential to understand all implications of a new technology and derisk future decisions before marketing. Due to tight time-to-market constraints and the permanently increasing complexity, prototyping was more and more abandoned in wireless industry. Prototyping would have required a larger and larger group of engineers, and the time to build a single prototype would have taken as long as designing the entire product. However, abandoning prototyping entirely and basing a design purely on simulations is very risky as some cases have proven(see, e.g., [7] for details). In order to bring prototyping back into the design chain, specific tools for prototyping must be available [7, 8], allowing for a rapid prototyping design that can be implemented much faster than the entire product with a small design team but certainly allowing for real-time experiments on a hardware very similar to the final product. Note that the terminology in this field is often used rather loosely. In [9] it has been distinguished between demonstrators, testbeds, and prototyping. We like to repeat this distinction for convenience of easier reading (see also Merriam-Webster [10] and Funk & Wagnalls Standard Dictionary of the English language [11]). (i) A demonstrator mainly serves as a sales vehicle and to show technology to customers. In general it will implement a new idea, concept, or standard that has already been established and has been finalized to some degree. Requirements on scalability are therefore less important than its functionality and often the required design time. (ii) A testbed on the other hand is generally used for research. It is a vehicle for further developments or for verification of algorithms or ideas under real-world or real-time conditions. This results in the requirement for scalability, modularity, and extendibility. (iii) A prototype is the initial realization of a research idea or a standard, either as a reference, a proof of concept, or as a vehicle for future developments and improvements. As opposed to a “simulation” it is not an “imitative representation” of the device. Instead it has significant similarities. In industry, a migration into a product is often intended. Testbeds and Rapid Prototyping in Wireless System Design 35

Table 1: Properties of testbeds and prototypes. Testbeds Prototypes Fast to program due to high-level language Considerable time required to build Data storage and offline processing possible Methods to speed up design are required Includes the wireless channel May include the wireless channel Only wireless transmissions in real-time Functionality in real-time DSP functions may be replaced by prototype Allows for design space exploration Fixed and/or floating point algorithms True fixed-point design

While demonstrators are most important for companies showing them to their potential customers, testbeds and prototypes are dominating the development process and are thus more of a research object. Both, testbeds and prototypes, allow for real-time experiments. However, testbeds allow only for a wireless transmission in real-time. The further processing of the received signals may be performed offline. While testbeds typically include the wireless channel, prototypes more often use channel emulators in order to provide an environment that allows for repeatable experiments. If testbeds are built properly, they may also allow for replacing the offline processing by a prototype or even the final product. While the processing of the algorithms is typically performed in a high-level language (i.e., in floating-point) when building a prototype, it is important to use a hardware platform close to the final product (i.e., processing is performed in fixed-point). The main properties and differences of testbeds and prototypes are summarized in Table 1.

3. Wireless testbeds The questions that can be solved with a testbed are essentially whether the channel model assumptions are viable and what impact the analog frontend may have. Although many MIMO channel models exist, most publications assume i.i.d. (independent identically distributed) flat Rayleigh fading channels. Measurements have proven that this is by far not true, and more or less complicated models have been proposed [12, 13]. Even the most complicated models still have imperfections and do not necessarily address all points of interest. Typically, models are optimized with respect to channel capacity. However, they may fit capacity and still give incorrect behavior for BER performance [14, 15]. Typically the analog frontend of a testbed is of very high quality, satisfying linearity for a very large dynamic range of the signal. AD and DA converters provide high resolution of 14–16 bits. By introducing imperfections (cutting off bits or misadjusting the oscillator, e.g.) one can analyze how sensitive the algorithms are on the analog part of the transmission chain. Also deriving an idea of the final product costs, based on linearity requirements for example, is possible. If it turns out that the high-precision analog frontend of the wireless testbed is not sufficient for the required performance, one knows for sure that the idea is not technically sound for a final product. Alternatively, if the required analog frontend is already available (reuse of a previous product, e.g.), it can be used in a testbed to clarify whether it is sufficient for the new algorithms. 36 Short Tutorials

3.1. Commercial tools Since companies do not use a consistent terminology and try to offer as many features as possible, it is not easy to categorize available products into testbeds and prototypes. Most can be used for both purposes, however some are suited better for one purpose than for the other. The selection here is thus somewhat subjective. Complete out-of-the box testbeds are available from Lyrtech (http://www.lyrtech.com)andSig- nalion (http://www.signalion.com). Components to build testbeds and prototypes are available from Sundance (http://www.sundance.com), Hunt-Engineering (http://www. hunteng.co.uk), and Pentek (http://www.pentek.com).1 Most products lack RF-frontends making it very hard to include the true physical channel. The development of RF frontends is costly and cumbersome. In particular, available frontends are missing flexibility in terms of supported carrier frequencies and bandwidths. Since specific design tools are typically missing in commercial testbeds and prototyping equipment, EDA (electronic design automation) tools used in the chip design process are also used for prototyping. For example, EDA tools are available from Synopsys and CoWare (and many smaller companies) but the licence costs are typically too high for prototyping. In particular, when heterogeneous systems including DSPs and FPGAs are required, no supporting design tools are available. More details about tools are provided further ahead in Section 5.1.

3.2. Vienna MIMO testbed Due to a lack of commercially available products, we decided to develop our own testbed. This testbed was designed by a team of researchers at the Institute of Communications and Radio Frequency Engineering of the Vienna University of Technology during the last three years. Throughout the development process, emphasis was placed on scalability, modularity, and extendibility, allowing for a multitude of very different experiments. The Vienna MIMO testbed [16, 17] primarily consists of (see Figure 1). (i) a transmit PC that automatically preprocesses and upconverts complex baseband data samples to a low IF (intermediate frequency) of 70 MHz, (ii) an analog transmitter frontend, performing linear frequency conversion of up to four low IF signal- to the radio frequency of 2.5 GHz, filtering, and amplification to the transmit power level of 30 dBm, (iii) a channel realized with channel emulators or a physical radio channel; two positioning tables are used to move the antennas in order to achieve the channel realizations needed for averaging the mean performance of a specific scenario, (iv) an analog receiver frontend that prefilters, amplifies, and downconverts the received signals of up to four antennas to a low IF of 70 MHz, (v) a receive PC that performs conversion to the digital baseband,

1Note that we only mention the most prominent ones. There is a multitude of smaller companies providing testbed and prototyping equipment. Testbeds and Rapid Prototyping in Wireless System Design 37

User PC (Matlab) TX Signal Optional feedback Receiver Synchro- RX ﬁlter generation algorithm nization ﬁlter

Transmit PC Receive PC Server software Server software Digital Analog Analog Digital baseband-to-IF IF-to-RF RF-to-IF IF-to-baseband upconversion DAC upconversion downconversion ADC downconversion TX antennas RX antennas Physical channel

XY positioning table XY positioning table 4xbaseband 4xIF @ 70 MHz 4xRF @ 2.5GHz 4xIF @ 70 MHz 4xbaseband

Figure 1: Block diagram of the Vienna MIMO testbed.

(vi) user PCs from where algorithm designers carry out various radio transmission experiments; using a ﬂexible Matlab Interface, complex baseband data samples of a 4 × 4 MIMO system can be easily transmitted and received.

The concept of the Vienna MIMO testbed allows the algorithm designer to focus on digital baseband data processing on his own PC. Next to being platform-independent, also multiple users can access the testbed directly out of Matlab from anywhere in the LAN. This not only makes it very convenient to integrate real-time air transmissions into existing Matlab simulations but also saves a lot of costs since very expensive2 hardware can be shared eﬃciently by several researchers. In the following, many examples are shown in which the testbed turned out to be useful.

3.3. Space-time codes In the last years, space-time codes were thoroughly investigated and optimized [18, 19, 20, 21, 22, 23]. However, hardly any results exist on how these space-time codes perform in a real system. Therefore, a MIMO testbed was used to measure the performance of an extended Alamouti (4 × 1 MISO) code [17, 24]. By using code selection with just two bits feedback (i.e., selection of one out of four codes) from the receiver to the transmitter, an extended Alamouti code transmission scheme achieves the full diversity order of four—at least in simulation. Surprisingly, the measurements with the testbed and channel emulators (which were configured to produce uncorrelated flat Rayleigh fading channels) showed an SNRlossofabout0.8dB(seeFigure 2) when using two bits feedback. It was found that this effect is caused by slightly asynchronous transmitter outputs. A mean delay of just 20 ns (approx. 10% of the symbol time) between the output samples of the transmitter chains caused the measured loss of 0.8 dB. For the implementation of a final product, exactly syn- chronous transmitter outputs are thus of utmost importance.

2The required equipment, including channel emulators and noise sources, costs approximately 500 000 C. 38 Short Tutorials

100

10 1 No diversity 10 2 2 x diversity 10 3 BER 4x diversity No 10 4 feedback

5 10 2bit feedback 10 6 2 0 2 4 6 8 10 12 14 16

Eb/N0 (dB) Simulation (0/2 bit feedback) Measurement (0/2 bit feedback)

Figure 2: BER performance of the extended Alamouti code (4.17 MSymbols/s) [17].

3.4. UMTS HSDPA equalizers The HSDPA (high speed downlink packet access) channels of UMTS provide high data rates to the user. These channels use a small spreading factor of 16 and are thus very sensitive to MAI (multiple access interference). MAI is caused by the multipath propagation channel which destroys the orthogonality of the spreading codes and therefore degrades the performance. The Vienna MIMO testbed, in conjunction with the channel emulators, was used to investigate the performance of diﬀerent equalizer structures for HSDPA [25]. Here, the channel emulators have a major advantage over Matlab channel models. The channel emulators allow for emulating paths with nearly arbitrary delays (0.5 ns stepsize). A Matlab channel model would need very high interpolation factors to oﬀer resolutions in the same dimension. Our measurement results showed that adaptive equalizer structures for HSDPA achieve the performance of the MMSE equalizer and clearly outperform the conventional RAKE receiver (Figure 3).

3.5. MIMO antenna design A general belief is that antennas for MIMO transmissions need to be at least λ/4spaced apart, resulting in large and visible antenna constructs. A wireless testbed can also be used to determine the performance of compact and realistic MIMO antennas. One example for such an antenna design is given in the following. A compact antenna with low constraints on its feasibility, designed for mobile communication equipment, is the so-called inverted-F antenna [26, 27]. It consists of a metal plate Testbeds and Rapid Prototyping in Wireless System Design 39

100

10 1

10 2 BER

10 3

10 4 2 0 2 4 6 8 101214

Eb/N0 (dB) 1code:RAKEreceiver 1 code: chip-rate adaptive equalizer 1 code: MMSE equalizer 8codes:RAKEreceiver 8 codes: chip-rate adaptive equalizer 8 codes: MMSE equalizer

Figure 3: Comparison of different receiver structures for HSDPA [17]. that is aligned in parallel to one side of a conducting box. One end of the metal plate is bent towards the box and connected to it (Figure 4). The feed consists of a wire that extends from the box and connects to the metal plate. Matching is achieved by choosing the position of the feeding wire and the width of the metal plate. The center frequency of the antenna is determined by the length of the metal plate (approx. λ/4). Bandwidth can be tuned by changing the distance between the metal plate and the box. Radiation of the inverted-F antenna is mainly caused by the currents in the feed wire and the part of the metal that is parallel to the feed wire. Furthermore, currents in the resonating metal plate and on the box surface cause radiation with additional arbitrary polarization states. To evaluate the performance that can be achieved by using MIMO techniques in a cell phone, a quad inverted-F antenna was built [28]. It consists of a metal box (28 × 28 × 8mm3 for the carrier frequency of 2.5 GHz) which is equipped with four inverted-F antenna elements (Figure 4). Due to the perpendicular or opposite alignment, the antennas are well decoupled from each other (S-parameter: |Sm,n|≤−15 dB for m = n). Furthermore, the perpendicularly aligned antenna elements emit most of the power into separate directions and employ different polarizations. All these effects enhance the MIMO performance of the quad inverted-F antenna because very different combinations of incident waves that 40 Short Tutorials

30 mm

Figure 4: Photograph of the quad inverted-F antenna. The four coaxial signal connectors seen on the top serve for interfacing to the RF frontend. reach the antenna in a multipath propagation scenario are presented at the four outputs. The overall size of 34 × 34 × 8mm3 of the quad inverted-F antenna allows for integration into cell phones. Furthermore, the RF frontend of the cell phone can be placed within the metal box of the antenna. The performance of such an antenna was evaluated with MIMO HSDPA throughput experiments reported in the following section.

3.6. MIMO UMTS HSDPA throughput The above-mentioned quad inverted-F MIMO antenna was used to directly investigate the impact of the antenna conﬁguration on the throughput of a MIMO UMTS system [29]. The system was implemented according to the DSTTD-SGRC (double space-time transmit diversity with subgroup rate control) proposal of Mitsubishi [30, 31]. The performance of the quad inverted-F antenna was compared to a linear antenna array consisting of four λ/4-monopole ground-plane antennas with an element spacing of λ/2. Figure 5 shows the results of throughput measurements plotted over Ec/Ior (energy of the transmitted chip-stream over total available transmitter energy). Surprisingly, the rather small quad inverted-F antenna shows approximately the same performance as the linear antenna array. The fourth antenna element of the quad inverted-F antenna does not add a signiﬁcant performance gain (approx. 0.4 dB). Nevertheless, the measurement results show that small and compact user equipments capable of MIMO transmission are feasible.

4. Rapid prototyping As mentioned in Section 2, conventional prototyping becomes more and more abandoned because of tight time-to-market constraints. Nevertheless, prototyping is an important Testbeds and Rapid Prototyping in Wireless System Design 41

106 3

2.5 4 = R 3 N = 2 R N 2 = R 1.5 N

1 Throughput (bits/s)

0.5

0 26 24 22 20 18 16 14 12 10 8 6

Ec/Ior (dB)

Linear array Quad inverted F

Figure 5: Throughput measurement for varying number of receive antenna elements NR [28]. means to derisk a new technology and should be applied somehow. A possibility is to overcome the slow prototyping development by including a particular rapid prototyping design methodology and thus speeding up the development considerably. Since very likely a future product will not only consist of a single DSP, it is of utmost importance to support DSPs as well as FPGAs, thus oﬀering HW/SW partitioning. Communication between connected blocks needs to be taken into account since it may be rather complicated to move functions from DSPs to FPGAs while it is usually relatively simple to stay on an FPGA or on a DSP when moving from block to block (as long as resources are available). An important feature to speed up development time is to use high-level languages wherever possible. For example, many DSPs from Texas Instruments (C6x) and Analog Devices (SHARK) can be programmed fully in C including driver functions and DMA transfer. Also many tools exist that convert C into VHDL [32] or SystemC, both allowing to generate hardware. Most im- portantly, these tools allow to satisfy the one-code paradigm [7], claiming that most errors can be avoided when sticking to a single code for description in high-level design and converting such code to lower design levels by automatic tools only. In this section, we begin by summarizing available commercial design tools and discussing their properties. Then we introduce the Vienna prototyping environment (VPE) and ﬁnally show two examples where it has been applied successfully. 42 Short Tutorials

4.1. Commercial tools Commercial tools are typically based entirely on DSPs or entirely on FPGAs. Texas Instru- ments C6x processors (and some others like Analog Devices’ SHARK) can fully and efficiently be programmed in ANSI-C. Matlab’s Simulink supports several platforms with C6x DSPs. Thus, if it would simply be a task of mapping to a multi-DSP board, this could efficiently be achieved by such tools. On the other hand, tools that map Simulink-code onto Xilinx or Altera FPGAs also exist, including a fixed-point library. Here the algorithm needs to be described with elements of this fixed-point library (can also be utilized in floating point). Once the algorithm runs sufficiently in simulation, it can be mapped on to the desired FPGA. However, typically such algorithms need to be of feed-forward structure. Once feedback elements are included, the simulation times become rather long due to inefficient scheduling. Also, the algorithms should be data-flow driven and not control-flow oriented. For control-flow oriented algorithms such programming is very tedious and even small algorithmic changes can require a long time. Lyrtech supports both conversion tools for DSPs and FPGAs on their boards including combinations of DSPs and FPGAs. The communication between a DSP and an FPGA is described by a special block under Simulink. While this is not extremely efficient, it provides a means to map algorithms directly onto FPGAs and DSPs provided one has sufficient resources available. Other companies like Sundance, Hunt Engineering, Signalion, or Pentek offer various modules with FPGAs and DSPs and driver software supporting the communication links between the various modules. However, complete prototyping tools that would allow to map from high level design to the hardware platform are typically not provided. Note that of the many providers, only Pentek and Signalion offer RF frontends (as of today), although most other companies have at least concepts for hardware they are planning to offer in the near future.

4.2. Vienna prototyping environment The origin of the current Vienna prototyping environment (VPE) goes back to a development at Bell Labs that supported prototyping of local wireless loops [33]andearlyUMTS MIMO receivers [34]. The VPE follows the one-code paradigm [7], that is, only the high- level description is refined in every design step, but no manual recoding is required. The design flow starts with COSSAP-C module descriptions, that is, ANSI C code that is en- riched with port information defining the input and output ports as well as the rates of the data streams and corresponding data types. This information can be used to convert the module to a Simulink S-function and run simulations. Once it has been decided which blocks are mapped onto DSPs and which onto FPGAs, the refinement of the C code follows different rules. For the DSP code, the variables are converted to 16- bit short types and intrinsic commands are included. With corresponding mapping algorithms the code can still run under Simulink and be tested. For FPGAs, a similar technique is used to refine the C code so that it can be mapped automatically to an FPGA using Adelantetech’s Builder- tool. Currently, many tools like this exist [32], often of academic origin and freely available. Testbeds and Rapid Prototyping in Wireless System Design 43

An important fact is that the algorithm in its refined form, although now suitable for a transformation to a hardware description language, can still be run under Simulink. Very significant is the possibility to coverify the algorithms once they are converted to hardware. Many people validate their algorithms with Modelsim [35] that can be used for cosimula- tion with Simulink. However, Modelsim models on a fine granular level and is thus very time consuming. We developed a method that allows not only to map the high-level description automatically onto DSPs and FPGAs but also to run the algorithm on the HW platform while the remaining parts are still running under Simulink. Originally developed at Rice University [36]andadoptedatBellLabs[7], the method was made more flexible to convert it easily onto other hardware platforms and it was extended to FPGAs as well [37]. Many details of the technique can be found in [7]. 4.3. WLAN prototyping The VPE was used to set up an 802.11n MIMO wireless LAN system utilizing the prototyping platform Smartsim by Seibersdorf Research [38, 39]. Since during the standardization of 802.11n several transmission schemes were discussed, the undefined parts of the system had to be implemented in a high-level language to quickly adopt for changes after the release of the final standard. The WLAN receiver is split into two parts. The first one consists of a receive filter, cyclic prefix removal, and a 64-point FFT. This part of the receiver was already used in the 802.11a standard and was not subject to changes to ensure downward compatibility of the MIMO extension to existing WLAN systems. Therefore, this part was implemented on a Xilinx Virtex 2 FPGA using Xilinx Core Generator which comprises highly optimized filters and FFT blocks. The functional chain, including receive filters and FFT, is implemented for every receive antenna. The second part of the receiver is the channel estimation and the space- time receiver. This part of the receiver was implemented in Generic C. Using a Generic C code, an automatic code conversion tool was used to generate a Simulink model and assembler code for a TI 6416 DSP.This method allows for coverification of the implemented Generic C receiver with a Simulink implementation. Furthermore, the method also allows to verify the code after refinement steps for improving the performance of the real-time implementation. 4.4. Adaptive predistortion Power amplifiers in wireless systems introduce nonlinear distortions and thus undesired spurious emissions in neighboring bands. In order to reduce such emissions, adaptive predistortion techniques are useful. The nonlinear behavior of the power amplifier is described by a parametric model. In a first step, its parameters have to be estimated. In the following, the inverse system is computed and the input of the power amplifier is predistorted by such an inverse system causing the output of the amplifier to be linear again. Narrow band systems (e.g., up to 1 MHz at 2 GHz) can be sufficiently described by memoryless map- pings, and inverse systems can be implemented by look-up tables [40]. For systems with higher bandwidth, memory plays a crucial role. Classical approaches use nonlinear models like Hammerstein or Wiener models. However, such models typically lead to a minimization problem that is nonlinear with respect to its parameters. Thus, simple gradient-type 44 Short Tutorials algorithms tend to run into local minima, not providing better results than memoryless approaches. A recently developed new method [41, 42] computes the predistorted signal of the amplifier directly, rather than computing the inverse explicitly. The corresponding method is an iterative method that approaches the desired quality in typically three steps. In order to check the method, a prototype based on Sundance modules was built applying the VPE. Figure 6 depicts the results showing strong improvement and at the same time a very good agreement of simulation and real-time measurement results.

5. Future challenges Modern wireless systems, with their fast exponential increase in complexity, are very challenging when it comes to converting algorithms to run on real-time hardware. 5.1. Methods Despite many efforts, several problems remain unsolved. Including hardware-in-the-loop (HIL) techniques for DSPs [36] and FPGAs [7, 37] certainly eases verification a lot. Some companies now offer such techniques [43]andaimatoffering techniques that allow to map from Simulink to heterogeneous DSP/FPGA boards. The main problems noted in the introduction, however, float-to-fix conversion, automatic partitioning, and platform-based design, remain open. Recently, some work on these topics has started [44, 45]butitwill take some time until all of these aspects are included into a smooth design flow. One major problem is the conversion of block based processing, as it is given in Simulink and many other graphical tools, to HW platforms not supporting such processing with relatively large storage buffers between the blocks. While large buffers are needed for high-speed simulation (we already mentioned the scheduling slow down in feedback systems in Section 4.1), small buffers are easier to implement. A solution could be utilizing variable buffer sizes when specifying the design, independent of the algorithm. The buffer size can then be set to a large value supporting fast simulation as well as to a small value supporting efficient implementation. 5.2. RF-frontends In modern communication systems with powerful digital signal processing, tasks like modulation and pulse shaping are more and more shifted from the analog to the digital domain. ADCs and DACs are the interfacing components of these two domains. Such converters exchange either analog baseband signals or low intermediate frequency signals with the analog frontend. The task of the RF frontend is to perform a linear frequency conversion to a carrier frequency suitable for wireless transmission. The carrier frequency conversion can be carried out in one single step (zero IF) or multiple steps (heterodyne). While highly integrated consumer products use the zero IF principle, in research often the heterodyne conversion is preferred. Due to market pressure, highly integrated consumer RF frontend chips are designed to just meet the minimum requirements of a specific final product. This makes single chip RF frontend solutions useless for wireless testbeds that should support amultitudeofdifferent present and future standards. Usually, high design flexibility and fast availability of an RF frontend can only be achieved by a heterodyne design which is Testbeds and Rapid Prototyping in Wireless System Design 45

RWB = 100 kHz VBW = 100 kHz ATT = 10 dB 30 35 40 45 50 No DPD, IBO 1 dB 55 DPD, real-time 60 65

(dBm/100 kHz) 70

P 75 80 85 90 DPD, non real-time (Matlab) 95 2.44 2.445 2.45 2.455 2.46 f (GHz)

Figure 6: Improvement by predistortion of a power amplifier (DPD = digital predistortion, IBO = input back-off). inherently much more complex. Thus, the RF frontend of a wireless testbed has to be built employing the heterodyne principle from available state-of-the-art components with lower integration level (e.g., mixers, amplifiers, filters) but much better performance. The main challenge when designing a wireless testbed is therefore to build a high-performance RF frontend supporting future standards with today’s technology. Note that the availability of components is crucial in the design process of a testbed or prototype since these are often built years before a standard is defined or first products are launched [46].

6. Conclusions No doubt, rapid prototyping has gained importance. In 2003 the proposal for the NoE NE- SAT was rejected by the European commission due to its lack of testbeds.3 Decisions on future wireless systems require a lot of ﬁnancial investment, making them crucially dependent on the utilized technology. Rapid prototyping can derisk such decisions long before the investment is made. In this tutorial, we presented means of prototyping. In particular, we distinguished strictly between wireless testbeds and prototypes. We showed, by many examples, how testbeds and prototypes help to overcome technological hurdles without requiring a too high-initial investment of time and money. Current challenges are in ﬂexible RF frontends, as well as in the development of automatic design tools.

3Comment of the reviewers: “A real-time testbed is a critical tool, allowing more impact on standardization bodies, however partnership is weak in this area.” 46 Short Tutorials

Acknowledgments The authors would like to thank many students who have taken part in the development and utilization of our Vienna MIMO testbed and the Vienna prototyping environment. In particular, the authors would like to mention Ernst Aschbacher, Werner Keim, Biljana Badic, Stefan Geirhofer, Georg Brandmayr, and Peter Brunmayr. This work has been funded by the Christian Doppler Laboratory for Design Methodology of Signal Processing Algorithms, http://www.nt.tuwien.ac.at/cdlab.

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Markus Rupp was born in 1963 in Volklingen,¨ Germany. He received his Dipl. Ing. degree in 1988 from the University of Saarbrucken,¨ Germany, and his Dr. Ing. degree in 1993 from the Technische Universitat¨ Darmstadt, Germany, where he worked with Eberhardt Hansler¨ on designing new algorithms for acoustical and electrical echo compensation. From November 1993 until July 1995, he had a postdoctoral positionat the University of Santa Barbara, California, with Sanjit Mitra where Testbeds and Rapid Prototyping in Wireless System Design 49 he worked with Ali H. Sayed on a robustness description of adaptive filters with impacts on neural networks and active noise control. From October 1995 until August 2001, he was a member of the technical staff in the Wireless Technology Research Department, Bell Labs, where he was working on various topics related to adaptive equalization and rapid implementation for IS-136, 802.11, and UMTS. Since October 2001 he has been a Full Professor of digital signal processing in mobile communications at the Vienna University of Technology and since July 2002 he has been Founder and Director of the Christian Doppler Laboratory for Design Methodology of Signal Processing Al- gorithms. He served as Associate Editor of IEEE Transactions on Signal Processing from 2002–2005, and is currently an Associated Editor of EURASIP Journal on Applied Signal Processing of EURASIP Journal on Embedded Systems and is an elected AdCom Member of EURASIP.He authored and coau- thored more than 200 papers and patents on adaptive filtering, wireless communications, and rapid prototyping.

Christian Mehlfuhrer¨ was born in 1979 in Vienna, Austria. In 2004 he received his Dipl. Ing. degree in electrical engineering from the Vienna University of Technology. Besides his diploma studies he worked part time at Siemens AG where he performed integration tests of GSM carrier units. After ﬁnishing his diploma thesis on implementation and real-time testing of space-time block codes at the Institute of Communications and Radio Frequency Engineering, Vienna University of Technol- ogy, for which he received the Vodafone “Forderpreis¨ 2006” (together with Sebastian Caban), he is now a member of the Christian Doppler Laboratory for Design Methodology of Signal Processing Algorithms and works towards his doctoral thesis at the same institute. His research interests include rapid prototyping, experimental investigation of MIMO systems, the upcoming UMTS MIMO HS- DPA mode, the MIMO extension for WLAN (802.11n), and WiMAX (802.16).

Sebastian Caban was born in 1980 in Vienna, Austria. Studying electrical engineering at the Vienna University of Technology, he received several scholarships as the best student of the college. After a year at the University of Illinois at Urbana-Champaign, USA, he is currently working as a Teaching Assistant on his Ph.D. thesis at the university’s Institute of Communications and Radio Frequency En- gineering. On June 23, 2006, he was awarded the “Forderpreis¨ 2006” of the German Vodafone Stiftung fur¨ Forschung for his research which focuses on rapid prototyping in wireless communications and multiple antenna systems.

Robert Langwieser was born in 1972 in Linz, Austria. He studied telecommunications at the Vienna University of Technology and earned his Master’s degree in April 2004 with distinction. At present he is employed as a Research Assistant and is Member of the Christian Doppler Laboratory for Design Methodology of Signal Processing Algorithms. He currently works on the development of a modular radio frontend for MIMO measurements.

Lukas W. Mayer was born in 1980 in Vienna, Austria. He ﬁnished his studies at the Vienna Univer- sity of Technology in 2005. In his diploma thesis he worked on the radio frequency frontend of the institute’s “Vienna MIMO testbed.” His research interests focus on rapid prototyping in wireless communications, antennas, and MIMO antenna systems. The task of his Ph.D. thesis is to develop small antennas that will be used for a novel RFID system. 50 Short Tutorials

Arpad L. Scholtz was born in 1947 in Kecskemet, Hungary. In 1956 he moved to Austria and received the Austrian citizenship in 1960. He studied telecommunications at the Vienna University of Technol- ogy, where he earned his Master’s degree in 1972 and the Ph.D. degree in 1976, both with distinction. From 1972 to 1982, he worked as an Assistant Professor for radio-frequency engineering at the In- stitute of Communications and Radio Frequency Engineering. Since 1982, he additionally has been serving at the same institute as a Lecturer, and in 1992 he was given the title of Associate Professor. He is author or coauthor of more than a hundred scientiﬁc publications. He teaches radio-frequency engineering with emphasis on electronic circuit design, antennas, and point-to-point communications. EURASIP (CO-)SPONSORED EVENTS

Report on the 48th International Symposium ELMAR-2006 focused on Multimedia Signal Processing and Communications, 07–09 June 2006, Zadar, Croatia

Plenary talks Symposium dinner with klapa “Galija” (Dalmatian folk singing)

Symposium participants on Skradinski Buk Symposium organizers in National park “Krka”

The 48th International Symposium ELMAR-2006 focused on Multimedia Signal Processing and Communications was organised by the Croatian Society Electronics in Marine, Zadar, Croatia, together with Department of Wireless Communications, Faculty of Electrical En- gineering and Computing, University of Zagreb, Croatia, during 7th, 8th and 9th of June 2006. ELMAR-2006 symposium was organized in cooperation with The European Associa- tion for Signal, Speech and Image Processing - EURASIP. Symposium took place under the technical co-sponsorship of IEEE Signal Processing Society, IEEE Region 8 and IEEE Croa- tia Section. As every year, ELMAR-2006 symposium was supported by Tankerska plovidba Zadar. 52 EURASIP (Co-)Sponsored Events

Symposium participants were very active on all sessions

Mr. Mihail Nicolaidis: “Satellite Communications in Marine-EUTELSAT Approach”

The ELMAR-2006 symposium programme consist of one keynote talk, two plenary talks, and 13 sessions, where 89 papers written by 190 authors were presented. The authors of the papers presented in ELMAR-2006 symposium are prominent researchers from 22 different countries. Overall, more then 64% papers are from abroad. The keynote talk was given by Professor Branka Zovko-Cihlar from the Faculty of Elec- trical Engineering and Computing of the University of Zagreb. She gave a talk: “Tesla’s In- ventions in the Field of Radiocommunications”, on the occasion of “2006 - The Year of Nikola Tesla” and 150th Anniversary of his birth. First plenary talk was given by Prof. Dr.- Ing. Thomas Sikora from the Technische Universitat¨ Berlin, Germany: “Squeezing Seman- tics from Media”, and the second plenary talk was given by Dr. Mubarak Shah from the University of Central Florida, USA: “Human Action Recognition”. This year, a Special Session on Optoelectronic Structures & High Bit Rates Transmission over Fiber was organized by Mrs. Sonja Zentner Pilinsky, Ph.D. 9 scientific papers from 6 European countries were presented in this interesting and special session that covered one very important scientific filed. Report on the 48th International Symposium ELMAR-2006 53

As every year, social events were organized to oﬀer a further opportunity to discuss both technical and non-technical subjects between attendees. First event was guided Zadar city sightseeing and visit to St. Mary’s church (The Permanent Ecclesiastical Art Exhibition: “The Gold and Silver of Zadar”), and the second event was excursion to National park “Krka.”

Mislav Grgic, ELMAR-2006 Program Chair [email protected] EURASIP (CO-)SPONSORED EVENTS

Report on XI International Conference SPECOM ’2006

25–29 June 2006 the 11th International Conference “Speech and Computer” was held in St. Petersburg, Russia. The International Conference SPECOM was established in 1996 by St. Petersburg Institute for Informatics and Automation of the Russian Academy of Sciences (SPIIRAS) together with the State Pedagogical University of Russia. Since then SPECOM was held in Moscow by Moscow State Linguistic University (MSLU), in Cluj-Napoca (Ro- mania) by Cluj-Napoca subsidiary of Research Institute for Computer Technique and 10th jubilee SPECOM ’2005 was hosted by the University of Patras in Greece. SPECOM ’2006 was organized by SPIIRAS at the financial support of the Administra- tion of St. Petersburg, SIMILAR NoE, INTAS International Association, Russian Founda- tion for Basic Research, ICA International Commission for Acoustics, U.S. Army Research Laboratory European Research Office; informational supporters: ISCA, EURASIP, ELSNET as well as service agency Monomax Ltd. The final technical program included 109 papers from 262 authors of 33 countries presented at 11 oral, poster and demonstration sessions. All papers have been subjected to a thorough review by members of the Scientific Committee consisted of 32 distinguished reviewers. Paper acceptance ratio was 74,2 % for regular sessions. Also in the framework of the conference several special events were organized: • Tutorials Day (Prof. Lawrence Rabiner: “Challenges in Speech Recognition and Nat- ural Language Understanding”; Prof. Thierry Dutoit: “Corpus-based speech synthesis”) Report on Report on XI International Conference SPECOM ’2006 55

• SIMILAR NoE Multimodal Interfaces Day; • SIMILAR NoE brainstorming on multimodal semantic fusion (Prof. Niels Ole Bernsen and Prof. Benoit Macq). • Keynote lectures (Prof. Christian Wellekens: “Impact of variabilities on speech recognition”; Prof. Ruediger Hoﬀmann: “Speech synthesis on the way to embedded systems”; Prof. Christoph Draxler: “Web-based speech data collection and annotation”).

Currently the development of natural means of human-computer interaction becomes one of the main directions in informatics. Permanent advancing possibilities of computers and network technologies are not used in full measure yet owing to “unnatural” form of a human-computer dialog. Existing insufficiency in this problem solving suppresses the development of various applied systems in telecommunications, medicine, edutainment and everyday life, since all modern techniques and network services use automated devices for information control and processing. Each year the SPECOM conference aims at providing all professionals dealing with human computer interaction for the best of science and the best of education. This forum is a unique opportunity to meet colleagues from all over the world and to exchange knowledge related to various aspects of Human-Computer Interac- tion. The conference venue and dates were selected so that the participants can possibly be exposed to St. Petersburg unique and wonderful phenomenon known as the White Nights, for our city is the world’s only metropolis where such a phenomenon occurs every summer. More details (members of the Scientific Committee, Scientific program, photos, etc.) can be found at the conference website http://www.specom.nw.ru.

Dr. Andrey Ronzhin, Speech Informatics Group of SPIIRAS [email protected] EURASIP (CO-)SPONSORED EVENTS

Calendar of Events

EURASIP Chairperson/Information Year Date Event Location Involvement 2006 September International Conference on Artiﬁcial Athens, Cooperation Stefanos Kollias 10–14 Neural Networks (ICANN ’06) Greece http://www.icann2006.org/ September Multimedia Content Representation, Istanbul, Cooperation Bilge Gunsel, Anil Jain 11–13 Classiﬁcation and Security (MRCS) Turkey http://www.ehb.itu.edu.tr/∼mrcs/ September Numerical Linear Algebra in Signals and Monopoli, Cooperation Nicola Mastronardi 11–15 Systems Italy http://www.cs.kuleuven.be/∼raf/ bari2006/ September International Workshop on Acoustic Echo Paris, France Cooperation Yves Grenier 12–14 and Noise Control (IWAENC 2006) http://www.iwaenc06.enst.fr/ September Advanced Concepts for Intelligent Vision Antwerp, Cooperation Jacques Blanc-Talon 18–21 Systems (ACIVS 2006) Belgium http://acivs.org/acivs2006 September The 13th International Conference on Budapest, Cooperation Kalman Fazekas, Jan Turan 21–23 Systems, Signals and Image Processing Hungary http://cyberspace.mht.bme.hu/ (IWSSIP 2006) iwssip06/ December 1st International Conference on Semantic Athens, Cooperation Yannis Avrithis 6–8 and Digital Media Technologies (SAMT) Greece http://www.samt2006.org/ 2007 February Signal Processing, Pattern Recognition, Insbruck, Cooperation Robert Sablatnig 14–16 and Applications (SPPRA 2007) Austria http://www.iasted.org/conferences/ home-554.html February International Symposium on Signal Pro- Sharjah, Cooperation B.Boashash 12–15 cessing and its Applications, ISSPA 2007 U.A.E http://www.isspa.info/ February International ITG / IEEE Workshop on Vienna, Cooperation C.Mecklenbrauker,¨ M.Rupp 26–27 Smart Antennas (WSA 2007) Austria http://www.ftw.at May Nonlinear Speech processing (Nolisp ’07) Paris, France Cooperation Mohamed Chetouani 20–25 http://www.congres.upmc.fr/ nolisp2007/ June Waveform Diversity & Design Pisa, Cooperation Vinny Amuso, Maria Greco 4–8 Conference Italy http://www.waveformdiversity.org/ June 7th International Workshop on Image Santorini, Cooperation Yiannis Kompatsiaris, 6–8 Analysis for Multimedia Interactive Greece Yannis Avrithis Services (WIAMIS 2007) http://mkg.iti.gr/wiamis2007 June 6th EURASIP Conference on Speech and Maribor, Cooperation Zarkoˇ F. Cuˇ cejˇ 26–30 Image Processing, Multimedia Commu- Slovenia http://ec2007.feri.uni-mb.si/ nications and Services (ECSIPM 2007) members.htm September 15th European Signal Processing Poznan, Sponsor Marek Domanski 3–7 Conference (EUSIPCO 2007) Poland http://www.eusipco2007.org September 26th Picture Coding Symposium (PCS Lisboa, Cooperation Fernando Pereira 7–9 2007) Portugal http://www.pcs2007.org/ 2008 August 16th European Signal Processing Lausanne, Sponsor Jean Philip Thiran 26–29 Conference (EUSIPCO 2008) Switzerland http://www.eusipco2008.org/

Markus Rupp; Workshops/Confs Coordinator EURASIP The Fourth IASTED International Conference on SIGNAL PROCESSING, PATTERN RECOGNITION, AND APPLICATIONS February 14 – 16, 2007 Innsbruck, Austria

ORGANIZING SPONSOR CALL FOR PAPERS The International Association of Science and Technology for Development (IASTED) Signal Processing, Pattern Recognition, and Application (SPPRA 2007) will be an • international forum for researchers and practitioners interested in the advances in, Technical Committee on and applications of, signal processing and pattern recognition. It is an opportunity to Signal Processing • present and observe the latest research, results, and ideas in these areas. All papers Technical Committee on submitted to SPPRA 2007 will be peer reviewed by at least two different reviewers. Pattern Recognition Acceptance will be based primarily on originality and contribution.

IN COOPERATION WITH SCOPE The European Association for Signal Image Processing (EURASIP) SIGNAL PROCESSING APPLICATIONS in: x Signal Analysis and Processing x Telecommunications x Detection and Estimation x Medicine x Audio and Video x Radar x Filters x Robotics CONFERENCE CHAIR x Algorithms and Techniques x Manufacturing Robert Sablatnig x Engineering Vienna University of Technology, PATTERN RECOGNITION Austria x Seismic x Image Analysis x Economics x Image Recognition, Coding, and INTERNATIONAL PROGRAM x Remote Sensing COMMITTEE Compression x Ocean Engineering (Preliminary List) x Watermarking Techniques E.-W. Bai, USA x Computer Vision D. Bull, UK D. Casasent, USA W.-K. Cham, PRC SUBMISSION A. la Cour-Harbo, Denmark F. Cruz-Roldán, Spain E.R. Davies, UK Guidelines for submitting papers, special sessions, and or tutorials can be found on G. Deng, Australia Y. Gao, Australia the conference website: http://www.iasted.org/conferences/cfp-554.html. All initial L. Giubbolini, USA paper submissions should be in Adobe Acrobat (.pdf) format and adhere to the M. das Gracas de Almeida, Brazil E.R. Hancock, UK formatting instructions at: http://www.iasted.org/formatting-initial.htm. B. Herbst, South Africa J. Hornegger, Germany C.-T. Hsieh, Taiwan IMPORTANT DEADLINES P. Jancovic, UK E. Krajnik, Czech Republic A. Kuba, Hungary Submissions Due September 15, 2006 S.T.W. Kwong, PRC S.-W. Lee, Korea C.C. Li, USA Notification of Acceptance November 1, 2006 J. Ma, UK M. Muraszkiewicz, Poland Camera-ready Manuscript Due November 22, 2006 H. Niemann, Germany A. Pai, UK Registration Deadlines December 8, 2006 V. Parsa, Canada S. Sanei, UK J. Sanubari, Indonesia V. Sebesta, Czech Republic M. Sliskovic, Germany CONTACT T. Stathaki, UK F. Stentiford, UK IASTED Secretariat - SPPRA 2007 M. Tahernezhadi, USA J. Tuckova, Czech Republic E-mail: [email protected] W.L. Woo, UK Web Site: www.iasted.org B. Zagar, Austria S. Zahir, Canada Z.-H. Zhou, PRC http://www.iasted.org/conferences/cfp-554.html ADVANCED CONCEPTS FOR INTELLIGENT VISION SYSTEMS September 18-21, 2006 University of Antwerp, Antwerp, Belgium http://acivs.org/acivs2006

STEERING COMMITTEE th Jacques BLANC-TALON is the 8 of a series of conferences focusing on techniques for building adaptive, MRIS, France intelligent, safe and secure imaging systems. will consist of 4 days of lecture Wilfried PHILIPS Ghent University, Belgium sessions, both regular (25 mn) and invited presentations, and poster sessions. Dan POPESCU will also feature a conference dinner and other social activities. CSIRO, Australia Paul SCHEUNDERS The conference fee includes the social program (conference dinner, coffee breaks, snacks and Atwerp University, Belgium cultural activities) and a hard copy of the LNCS proceedings. Students, IEEE and EURASIP members can register at a reduced fee.

PROGRAM COMMITTEE TOPICS INCLUDE (BUT ARE NOT LIMITED TO): Fritz ALBREGTSEN, University of Oslo, Norway Attila BASKURT, LIRIS, UCB Lyon 1, France − Image and Video Processing (linear/non-linear filtering and enhancement, restoration, Laure BLANC-FERAUD, INRIA/ARIANA, France segmentation, wavelets & multiresolution, Markovian techniques, color processing, modeling, Philippe BOLON, ESIA, France analysis, interpolation and spatial transforms, fractals-multifractals, structure from motion) Nikolaos BOURBAKIS, Wright State University, − Pattern Analysis (shape analysis, data and image fusion, pattern matching, neural nets, learning, USA grammatical techniques) and Content-Based Image Retrieval Salah BOURENNANE, EGIM, France − Remote Sensing (techniques for filtering, enhancing, compressing, displaying and analyzing Patrick BOUTHEMY, IRISA/INRIA, France Jocelyn CHANUSSOT, INPG-ENSIEG, France optical, infrared, radar, multi-hyperspectral airborne and spaceborne images) David CLAUSI, University of Waterloo, Canada − Still Image and Video Transmission (still image/video coding, model-based coding, Pamela COSMAN, University of California, USA synthetic/natural hybrid coding, quality metrics, image watermarking, image and video databases, Jennifer DAVIDSON, Iowa State University, USA image search and sorting, video indexing, multimedia applications) Ricardo de QUEIROZ, Univ. de Brasilia, Brazil − System Architecture and Performance Evaluation (implementation of algorithms, benchmarking, Christine FERNANDEZ-MALOIGNE, SP2MI/SIC, evaluation criteria, algorithmic evaluation) France Jan FLUSSER, IITA, Czech Republic Both classical research papers and application papers are welcome. Don FRASER, ADFA, Australia Sidharta GAUTAMA, Ghent University, Belgium VENUE Jérôme GILLES, CEP, Arcueil, France Georgy GIMEL'FARB, Auckland University, New Zealand will take place in the Middelheim campus of the university of Antwerp which can be easily Daniele GIUSTO, University of Cagliari, Italy reached from Antwerp airport or by train from Brussels main airport. John ILLINGWORTH, University of Surrey, UK Pieter JONKER, Delft University of Technology, PAPER SUBMISSION AND REVIEW PROCESS The Netherlands Prospective authors should prepare a full paper and submit it electronically. The paper should consist of 8- Frédéric JURIE, INRIA/GRAVIR, France Andrzej KASINSKI, Poznan University of 12 pages in A4 format and should conform to the style guidelines outlines on the Technology, Poland website. LaTex style sheets, MSWord templates and more detailed information on the submission process Ashraf KASSIM, National University of Singapore can be found on the conference website: http://acivs.org/acivs2006 Richard KLEIHORST, Philips Research, The Netherlands All submissions will be reviewed by at least 2 members of the PC; extra reviewers may be consulted if Murat KUNT, EPFL, Switzerland needed. Submissions should provide sufficient background information and should clearly indicate the Hideo KURODA, Nagasaki University, Japan original contribution; they should state and discuss the main results and provide adequate references. Kenneth LAM, The Hong Kong Polytechnic Paper submission implies that one of the authors will present the paper if it is accepted. University, China Bruce LITOW, James Cook University, Australia ONFERENCE PROCEEDINGS Jesus MALO, Universitat de Valencia, Spain C Gérard MEDIONI, USC/IRIS, USA Fabrice MÉRIAUDEAU, Le2i Le Creusot, France Ali MOHAMMAD-DJAFARI, SUPELEC/LSS, France The proceedings of Acivs 2006 will be published by Springer-Verlag in the Lecture Rafael MOLINA, Universidad de Granada, Spain Vittorio MURINO, Universita` degli Studi di Notes in Computer Science series. LNCS is published, in parallel to the printed books, in full-text Verona, Italy electronic form via Springer-Verlag’s internet platform http://www.springerlink.com Stanley OSHER, UCLA, Los Angeles, USA Marcin PAPRZYCKI, SWPS, Warsaw, Poland IMPORTANT DATES Jussi PARKKINEN, Lappeenranta University, Finland Full paper submission April 9, 2006 Early registration July 14, 2006 Fernando PEREIRA, IST, Lisboa, Portugal Notification of acceptance May 22, 2006 Late registration August 21, 2006 Béatrice PESQUET-POPESCU, ENST Paris, France Final papers + author registration June 16, 2006 Sep 18-21, 2006 Matti PIETIKÄINEN, Oulu University, Finland Aleksandra PIZURICA, Ghent University, Belgium Gianni RAMPONI, Trieste University, Italy Luis Salgado A. de SOTOMAYOR, Universidad Politécnica de Madrid, Spain Frederic TRUCHETET, Le2i Le Creusot, France Benelux Signal Dimitri VAN DE VILLE, EPFL, Switzerland Processing Chapter Peter VEELAERT, Hogeschool Ghent, Belgium 6th EURASIP Conference EC-SIPMCS 2007 June 27 – 30, Maribor, Slovenia Call for Papers

http://ec2007.feri.uni-mb.si This conference is initiated by the European Association for Speech, Signal and Image Processing (EURASIP) in order to start a new tradition of conferences, each devoted to a specific area of discipline. It is focused on Speech and Image Processing, Multimedia Communications and Services (EC-SIPMCS). The goal of EC-SIPMCS is to promote the interface researchers involved in the development and applications of methods and techniques within the framework of speech/image processing, multimedia communications and services. The 6th EC-SIPMCS will be held in Maribor, Slovenia from June 27 – June 30, 2007. Topics of Interest The program includes keynote and invited lectures by eminent international experts, peer reviewed contributed papers, posters, invited sessions on the same or related topics, industrial presentations and exhibitions around but not limited to the following topics:

• Image and Video Processing • Digital Video Broadcasting (DVB) • Image and Video Coding • Video Streaming and Videoconferencing • Image Scanning, Display and Printing Multimedia Signal Processing • Image and Video Indexing and Retrieval • Multimedia Databases • Speech and Audio Processing • Multimedia and DTV Technologies • Watermarking and Encryption • Multimedia Communications and Networking • Digital Signal Processing (DSP) • Multimedia Human-Machine Interface and • Standards and Related Issues Interaction • ICT in e-learning/consulting • Multimedia Services and Applications Publications All accepted papers will be published in CD Proceedings that will be available at the Conference. Abstracts of accepted papers will be printed and included in the INSPEC database. Selected papers will be considered for possible publication in scholarly journals. Tutorial and Special Sessions Those willing to prepare a tutorial course during EC-SIPMCS 2007 Conference and those willing to organize special session EC-SIPMCS 2007 Conference should contact dr. Zarko Cucej at [email protected]. Submission Guidelines Regular Papers Papers must be submitted electronically by March 18, 2007. Each paper will be evaluated by at least two independent reviewers, and will be accepted based on its originality, significance and clarity. Papers must not exceed 6 pages single spaced 11-point font. Important Dates Paper and Poster Submissions: March 18, 2007 Notification of acceptance: April 20, 2007 Camera ready copy due: May 6, 2007 Author Registration: May 6, 2007

General Chair Program Chair dr. Zarko Cucej, dr. Peter Planinsic, University of Maribor, University of Maribor, Faculty of Electrical Engineering and Computer Faculty of Electrical Engineering and Computer Science, Science, Maribor, Slovenia Maribor, Slovenia For further information pleas visit: http://ec2007.feri.uni-mb.si samt 1st International Conference on Semantic 2006 and Digital Media Technologies December 6-8, 2006 Athens, Greece

C a l l F o r P a p e r s

Technical Program Committee The first international conference on Semantics And digital Media Technology (SAMT) targets to narrow the large disparity between the lowlevel descriptors that can be computed automatically Bruno Bachimont, INA, France from multimedia content and the richness and subjectivity of semantics in user queries and human Wolf-Tilo Balke, Univ Hannover, Germany interpretations of audiovisual media - The Semantic Gap. Jenny Benois-Pineau, University of Bordeaux, France Jesus Bescos, GTI-UAM, Spain Nozha Boujemaa, INRIA, France SAMT started out as two workshops, EWIMT 2004 and EWIMT 2005, that quickly achieved enormous Patrick Bouthemy, IRISA, France success in attracting high-quality papers and over 100 participants from across Europe and beyond. Pablo Castells, NETS-UAM, Spain This year EWIMT has turned into the full-fledged conference SAMT,addressing integrative research Andrea Cavallaro, Queen Mary University of London on new knowledge-based forms of digital media systems. It brings together those forums, projects, Stavros Christodoulakis, Technical Univ. of Crete, Greece institutions and individuals investigating the integration of knowledge, semantics and low-level Fabio Ciravegna, University of Sheffield, UK multimedia processing, and links them with industrial research and development engineers who Thierry Declerck, DFKI, Germany exploit the underlying emerging technology. Anastasios Delopoulos, Aristotle Univ. of Thessaloniki, Greece Edward Delp, Purdue University, US In cooperation with the European Commission DG Information Society, the third day of the Martin Dzbor, Knowledge Media Institute, UK conference is dedicated to featuring the launch of the 7th Framework ICT Research programme Touradj Ebrahimi, Swiss Federal Institute of Technology, CH and keynote talks from EC representatives. Jerome Euzenat, INRIA, FR Borko Furht, Florida Atlantic University, US Moncef Gabbouj, Tampere University of Technology, Finland Prospective contributors are invited to submit their papers on-line according Christophe Garcia, France Telecom, France William I. Grosky, University of Michigan, US to the guidelines at http://www.samt2006.org Werner Haas, Joanneum Research, Austria Christian Halaschek-Wiener, University of Maryland, US Siegfried Handschuh, DERI Galway, Ireland Topicsof interest include, but are not limited to: Alan Hanjalic, TU Delft, The Netherlands Lynda Hardman, CWI, The Netherlands l Andreas Hotho, Univ Kassel, Germany Integration of multimedia processing and Semantic Web technologies l Jane Hunter, University of Queensland, Australia Multimedia ontologies and infrastructures l Antoine Isaac Vrije, Universiteit, The Netherlands Knowledge assisted multimedia data mining Ebroul Izquierdo, Queen Mary University of London, UK lKnowledge based inference for semi-automatic semantic media annotation Franciska de Jong, University of Twente, The Netherlands lIntegration of content-based image/video analysis with natural language and speech processing Joemon Jose, University of Glasgow, UK lMultimodal techniques, high dimensionality reduction and low-level feature fusion Aggelos K. Katsaggelos, Northwestern University, US lRelevance feedback for finding semantics Hyoung Joong Kim, Kangwon National University, Korea lSemantic-driven multimedia indexing and retrieval Joachim Kohler, Fraunhofer IMK, Germany lMetadata management for multimedia Stefanos Kollias, National Technical Univ. of Athens, Greece lBrowsing large multimedia archives Paul Lewis, University of Southampton, UK lScalable, semantic-driven multimedia content adaptation and summarization Petros Maragos, National Technical Univ. Athens, Greece lInterfaces and personalization for interaction with large multimedia repositories Ferran Marques, Technical University of Catalonia, Spain lSemantics-driven multimedia presentation generation Jose Martinez, GTI-UAM, Spain lContent, user,network and semantics-aware media engineering Adrian Matellanes, Motorola Labs, UK lStandards bridging the multimedia and knowledge domains Bernard Merialdo, EURECOM, France Frank Nack, CWI, The Netherlands Jacco van Ossenbruggen, CWI, The Netherlands Jeff Pan, University of Manchester, UK 15 May 2006 Special Sessions Pappas Thrasyvoulos, Northwestern University, US Dietrich Paulus, Uni Koblenz, Germany 22 May 2006 Tutorials & Workshops Eric Pauwels, CWI, The Netherlands 16 June 2006 Regular Paper Submission Ioannis Pittas, Aristotle University of Thessaloniki, Greece 11 August 2006 Notification of Acceptance Dennis Quan, IBM 11 September 2006 Submission of camera-ready papers Keith van Rijsbergen, University of Glasgow, UK Andrew Salway, University of Surrey, UK Mark Sandler, Queen Mary, University of London Simone Santini, Univ. of California, San Diego, US General Chairs Guus Schreiber, Free Univ. Amsterdam, NL YannisAvrithis, National TechnicalUniversity of Athens, Greece Timothy Shih, Tam Kang University, Taiwan Sergej Sizov, Uni Koblenz, Germany Yiannis Kompatsiaris, Informatics and TelematicsInstitute, Greece Alan Smeaton, Dublin City University, Ireland John R. Smith, IBM Research, US Technical Program Chairs Giorgos Stamou, National Technical Univ. of Athens, Greece Noel O’ Connor, Dublin City University,Ireland Fred S. Stentiford, University College London, UK Steffen Staab, University of Koblenz-Landau, Germany Michael Strintzis, ITI, Greece Rudi Studer, Univ Karlsruhe, Germany Special Sessions Vojtech Svatek, UEP, Czech Republic Jose Martinez, Universidad Autónoma de Madrid, Spain Murat Tekalp, Univ. Rochester, US Raphael Troncy, CWI, NL Tutorials & Workshops Paulo Villegas, Telefonica I+D, Spain Raphael Troncy, CWI, The Netherlands Gerhard Widmer, Univ. Linz, Austria Vassilis Tzouvaras, National TechnicalUniversity of Athens, Greece Li-Qun Xu, British Telecom, UK EU Liaison Paola Hobson, Motorola Labs, UK

IVML Information Society

ISSPA 2007 International Symposium on Signal Processing and its Applications in conjunction with the International Conference on Information Sciences, Signal Processing and their Applications 12 – 15 February 2007, Sharjah, U.A.E.

ISSPA General Chair & Steering Committee Chair B. Boashash University of Sharjah, UAE The University of Queensland, Australia

Conference Chair M. Bettayeb University of Sharjah, UAE First Call For Papers Conference Vice-Chair S. Al-Araji ISSPA 2007 marks the 20th anniversary of launching the first ISSPA in 1987 in Brisbane, Etisalat University College, UAE Australia. Since its inception, ISSPA has provided, through a series of 8 symposia, Technical Program a high quality forum for engineers and scientists engaged in research and development of K. Assaleh, Chair Signal and Image Processing theory and applications. Effective 2007, ISSPA will extend its American University of Sharjah, UAE M. IbnKahla, Co-Chair scope to add the new track of information sciences. Hence, the intention that the previous Queens University, Canada full name of ISSPA is replaced after 2007 by the following new full name: I. Tabus, Co - C ha i r Tampere University of Technology, Finland International Conference on Information Sciences, Signal Processing and their Applications. ISSPA is an IEEE indexed conference. Plenary Sessions S. Mitra University of California, Santa Barbara, USA ISSPA 2007 will be organized by three prominent institutions located in Sharjah in the United Arab Emirates: University of Sharjah, American University of Sharjah, and Etisalat University College. Special Sessions M. Barkat American University of Sharjah, UAE The regular technical program will run for three days along with an exhibition of signal M. Cheriet University of Quebec, Canada processing and information sciences products. In addition, tutorial sessions will be held on the L. Karam first day of the symposium. Papers are invited in, but not limited to, the following topics: Arizona State University, USA

Tutorials 1.Filter Design Theory and Methods 11. Multimedia Signal Processing 21.Signal Processing for Bioinformatics M. El-Tarhuni 2. Multirate Filtering & Wavelets 12. Nonlinear signal processing 22. Signal Processing for Geoinformatics American University of Sharjah, UAE 3.Adaptive Signal Processing 13.Biomedical Signal and Image 23.Biometric Systems and Security Processing Publications 4.Time-Frequency/Time-Scale Analysis 14.Image and Video Processing 24.Machine Vision M. Al-Qutayri 5.Statistical Signal & Array Processing 15.Image Segmentation and Scene 25.Data visualization Etisalat University College, UAE Analysis 6.Radar & Sonar Processing 16. VLSI for Signal and Image 26. Data mining Publicity Processing M. Al-Mualla 7.Speech Processing & Recognition 17.Cryptology, Steganography, and 27. Sensor Networks and Sensor Fusion Etisalat University College, UAE Digital Watermarking 8.Fractals and Chaos Signal Processing 18. Image indexing & retrieval 28.Signal Processing and Information Sponsorship & Exhibits Sciences Education K. Al-Midfa Etisalat University College, UAE 9.Signal Processing in 19.Soft Computing & Pattern 29.Others Communications Recognition Student Sessions 10.Signal processing in Networking 20. Natural Language Processing A. Elwakil University of Sharjah, UAE Perspective authors are invited to submit full length (four pages) papers for presentation in any of Finance & Registration the areas listed above (indicate area in your submission). We also encourage the submission of C. B. Yahya proposal for student session, tutorial and sessions on special topics. All articles submitted to University of Sharjah, UAE ISSPA 2007 will be peer-reviewed using a blind review process. Local Arrangements I. Kamel For more details see University of Sharjah, UAE Social Events w w w . i s s p a . i n f o / A. Al-Ali American University of Sharjah, UAE

Web and IT B. Soudan Important Deadlines: Conference Secretary University of Sharjah, UAE Full Paper Submission: A-K. Hamid June 30, 2006 University of Sharjah, UAE IEEE Liaison Tutorials/Special Sessions Proposals: (TBA) May 30,2006 Tel : +971 6 5050932 Industry Liaison Notification of Paper Acce ptance : Fax :+971 6 5050872 (TBA) October 15, 2006 E-mail: [email protected] Final Accepted Paper Submission: International Liaisons November, 15, 2006 S. Anderson, Australia DSTO, Australia T. Fukuda, Asia and Pacific Nagoya University, Japan M . Gabbouj, Europe Tampere University of Technology, Finland M. Jaidane, Africa ENIT, Tunisia Y. Zhang, America Villanova University, USA ORGANIZING COMMITTEE

General Co-Chair Fazekas K., Hungary Turán J., Slovak Republic Publicity Chair Panos Liatsis, UK IWSSIP'06

Tutorials Chair th Zovko-Cihlar B., Croatia 13 International Conference on Systems, Signals &Image Processing September 21-23, 2006 - Budapest, Hungary Special Sessions Chair Mislav Grgic, Croatia CALL FOR PAPERS The 13th International Conference on Systems, Signals and Image Processing, IWSSIP'06 is co-organised by the BUTE Organizing Committee Enyedi B., Hungary of Budapest, Hungary and TUKE of Košice, Slovak Republic, in co-operation with IEEE, IEE, IEEE Region 8, IEEE Konyha L., Hungary Hungarian Section and EURASIP. IWSSIP'06 is an International Conference on the theoretical, experimental and Szombathy Cs., Hungary applied signal and image processing techniques and systems which brings together researchers and developers from Tran S. M., Hungary both academia and industry to report on the latest scientific and theoretical advances, to discuss and debate major issues Ovseník L., Slovak Republic and to demonstrate state of-the-art systems. The IWSSIP'06 program will also include a variety of special sessions and tutorials devoted to recent and important developments in the field. INTERNATIONAL PROGRAM COMMITTEE SCOPE Ansorge M., Switzerland 1. Signal Processing: General techniques and algorithms. Adaptive DSP algorithms; Filter Bank Theory; Spectrum Bojkovic Z., Serbia Estimation and Processing; Non-linear Systems; Digital Transforms; Multidimensional Signal Processing; Pattern Chariglione L., Italy Recognition. Cristea P., Romania 2. Technologies: Neural Networks; Fuzzy Systems; Expert Systems; Genetic Algorithms; Data Fusion. Cucej Z., Slovenia 3. Multimedia Content Processing: Speech Processing and Recognition, Audio Enhancement, Restoration and Domanski M., Poland Analysis, Image Representation and Modelling, Image Restoration and Enhancement; Colour Vision, 3D Vision, Image Doshi B., USA and Video Analysis; Pattern Recognition; Watermarking, New Media. Ebrahimi T., Switzerland 4. Multimedia Data Compression: Speech and Audio Compression, Image and Video Coding, Scalable Techniques, Fazekas K., Hungary Standards. Grgic M., Croatia 5. Multimedia Systems: Human Factors, Multimodal Interfaces, Networked Multimedia, Seamless Audiovisual Grgic S., Croatia Networks, Multimedia Services; Multimedia Servers; Multimedia Streaming, Wireless and Mobile Multimedia, Heute U., Germany Universal Multimedia Access, Right Protection and Management. Izguierdo E., UK 6. Content Description: Metadata and Media Abstracts, Audiovisual Databases. Katsaggelos A. K., USA 7. Implementations: Analog and Digital Circuits and Systems for Audio, Image and Video Processing; Architectures Kazakos D., USA and VLSI Hardware; Programmable Signal Processors; Real-time Software. Kocur D., Slovak Republic 8. Applications: Bioinformatics; Broadcasting; Control; Communications; Digital Production; Medical; Opto- Levický,D., Slovak Republic mechatronics; Remote Sensing; Robotics; Speech; Television; Telepresence; e-learning; Virtual Reality; Remote Liatsis P., UK Sensing. Lukáþ R., Canada Marchevský S., Slovak Republic SUBMISSION PROCEDURE Mertzios B., Greece Prospective authors are invited to submit original research papers in any of the technical areas listed above. They should Pereira F., Portugal submit their full paper in English, including the summary of the accomplishments and the significance of the Pitas I., Greece contribution. Submission should include in a separate sheet the author(s) name(s) and affiliation(s); the contact author Planinsic P., Slovenia should be identified by providing his/her mail and e-mail address, telephone and fax numbers. All papers will be fully Podhradský P., Slovak Republic peer reviewed. Hardcopy and electronic submission, along with www uploading are available. For more information Rao K.R., USA please visit the official IWSSIP’06 Site: http://cyberspace.mht.bme.hu/iwssip06/ Accepted papers will be published Rozinaj G., Slovak Republic in the IWSSIP’06 Proceedings. At least one author of each accepted paper must register for the conference and present Schaefer R., Germany the contribution. After further peer reviewing, selected papers will be proposed for publication in special issues of Sonka M., USA scientific journals. Šimák B., Czech Republic Stamatis V., GREECE SCHEDULE Stasinski R., Poland Deadline for submission of full papers May 22, 2006 Tasic J., Slovenia Notification of acceptance mailed out by June 30, 2006 Tekalp M., USA Early Registration deadline July 10, 2006 Turán J., Slovak Republic Deadline for submission of camera-ready papers July 14, 2006 Vlaicu A., Romania Registration deadline August 14, 2006 Wajda K., Poland Conference September 21, 2006 Zahariadis T., GREECE Zervos N., Greece CONFERENCE SITE Zorzi M., Italy IWSSIP’06 will be held at Flamenco Hotel, Tasvezér utca 7., Budapest, Hungary Zovko-Cihlar B., Croatia Information: http://cyberspace.mht.bme.hu/iwssip06 and [email protected] (will be available after December 1, 2005.)

Budapest University of Technology and Economics General Chairs

Bilge G unsel Istanbul Technical Univ.,Turkey

Anil K. Jain Michigan State University, USA CALL FO R PAPERS Technical Program Chair Workshop on M ultimedia C ontent R epresentation, Classification and Security Murat Tekalp (MRCS) is being organized w ith the objective of bringing together researchers, Koc University, Turkey developers and practitioners from academ ia and industry w orking in related areas of multimedia system s. MRCS aims to serve as a forum for the dissem ination of Publicity Chair state-of-the-art research, developm ent, and implem entations of multim edia content extraction and classification, multim edia security technologies, and innovative Kivanc Mihcak Microsoft R esearch, USA applications. IMRCS is co-sponsored by IAPR, EURASIP, TUBITAK, ITU, and IEEE Turkey C hapter. Financial Chair

G ozde Bozdagi-Akar The areas of interest include but are not limited to: Middle East Tech. Univ.,Turkey - Feature extraction, multim edia content representation and classification techniques Local A rrangem ents - Multimedia signal processing Sima Etaner-Uyar - Authentication, content protection and digital rights m anagem ent Istanbul Technical Univ.,Turkey - Audio/Video/Im age Watermarking/Fingerprinting - Information hiding, steganography, steganalysis Program Comm ittee - Audio/Video/Im age hashing and clustering techniques - Evolutionary algorithm s in content based multim edia data representation, A. Akansu, NJIT, USA indexing and retrieval A. Alatan, METU, Turkey - Transform dom ain representations M. Barni, Univ. of Sienna, Italy P. Bouthem y, IRISA, France - Multimedia mining R.Civanlar,Koc Univ., Turkey - Benchm arking and com parative studies E. Delp, Purdeu University, USA - Multimedia applications (broadcasting, medical, biom etrics, content aw are J. Dittm ann, Otto-von-G uericke networks, CBIR.) Univ., Germ any C. Dorai, IBM T.J. Watson, USA Prospective authors are invited to subm it extended summaries of not more than six A. Ercil, Sabanci Univ., Turkey (6) pages including results, figures and references. Subm itted papers w ill be A. Fred, IST Lisbon, Portuqal review ed by at least two members of the program committee. Conference M.G okm en,ITU, Turkey A. H anjalic, Tech. Univ. Delft, Proceedings w ill be available on site. Please check the w ebsite Netherlands http://www.ehb.itu.edu.tr/~m rcs for further information. H. Ip, City University, H ong Kong D. Kundur, Texas A&M , USA I. Lagendijk, Tech. Univ. Delft, Im portant Dates Netherlands K.J.R. Liu, Univ. of Maryland,USA Special S essions (contact the technical program chair): March 10, 2006 J. Luo, Eastm an Kodak, USA B. Macq, UCL, Belgium Subm ission of Extended Summ ary: April 10, 2006 B. M anjunath, Univ. of CA, USA Notificatin of Acceptance: June 10, 2006 V. M onga, Xerox Labs, USA Camera-ready Paper Submission D ue: July 10, 2006 P. M oulin, Univ. of Illinois, USA L. Onural, BilkentU niv., Turkey Contact F. G onzalez, Univ. of Vigo, Spain B. Sankur, Bogazici Univ., Turkey J. Sm ith, IBM T.J. Watson, USA Istanbul Technical University, Electrical-Electronics Eng. Faculty, 34469 Maslak S. Voloshynovskiy, Univ. of Istanbul, Turkey. Attn: Dr. Sima Etaner-Uyar G eneva, Switzerland Phone: (90) 212 285 6471 Fax: (90) 212 285 3679 R. Venkatesan, Microsoft E-mail: mrcs@ ehb.itu.edu.tr W eb:http://www.ehb.itu.edu.tr/~m rcs R esearch, USA H-J. Zhang, Microsoft China

FIRST CALL February 2006 NOLISP’07 AN ISCA TUTORIAL AND RESEARCH WORKSHOP ON NON-LINEAR SPEECH PROCESSING

After the success of NOLISP’03 held in Le Croisic, and NOLISP’05 held in Barcelona, we are pleased to present NOLISP’07

22-25 May 2007, Paris, France

Local Organizers : University Pierre and Marie Curie (UPMC)

AIMS OF THE WORKSHOP SCIENTIFIC COMMITTEE Many specifics of the speech signal are not well Frédéric BIMBOT, IRISA, Rennes (France) addressed by the conventional models currently used in Mohamed CHETOUANI, UPMC, Paris (France) the field of speech processing. The purpose of the Gérard CHOLLET, ENST, Paris (France) workshop is to present and discuss novel ideas, work Tariq DURRANI, University of Strathclyde, Glasgow (UK) and results related to alternative techniques for speech Marcos FAÚNDEZ-ZANUY, EUPMt, Barcelona (Spain) processing, which depart from mainstream approaches. Bruno GAS, UPMC, Paris (France) Hynek HERMANSKY, OGI, Portland (USA) FOCUS OF THE WORKSHOP Amir HUSSAIN, University of Stirling, Scotland (UK) Eric KELLER, University of Lausanne (Switzerland) Contributions are expected in the following domains Bastiaan KLEIJN, KTH, Stockholm (Sweden) (non-limited list): Gernot KUBIN, TUG, Graz (Austria) I. Non-Linear Approximation and Estimation Petros MARAGOS, Nat. Tech. Univ. of Athens (Greece) II. Non-Linear Oscillators and Predictors Stephen Mc LAUGHLIN, University of Edimburgh (UK) III. Higher-Order Statistics Maurice MILGRAM, UPMC, Paris (France) IV. Independent Component Analysis Kuldip PALIWAL, University of Brisbane (Australia) V. Nearest Neighbours Bojan PETEK, University of Ljubljana (Slovenia) VI. Neural Networks Jean ROUAT, University of Sherbrooke (Canada). VII. Decision Trees Jean SCHOENTGEN, Univ. Libre Bruxelles (Belgium) VIII. Non-Parametric Models Isabel TRANCOSO, INESC (Portugal) IX. Dynamics of Non-Linear Systems X. Fractal Methods SUBMISSION XI. Chaos Modeling Prospective authors are invited to submit a 3 to 4-page XII. Non-Linear Differential Equations paper proposal in English, which will be evaluated by the XIII. Others Scientific Committee. Final papers will be due 1 month after the workshop, for inclusion in the CD-ROM All fields of speech processing are targeted by the proceedings. workshop, namely : 1. Speech Production KEY DATES 2. Speech Analysis and Modeling Submission (full paper): 15 January 2007 3. Speech Coding Notification of acceptance: 23 February 2007 4. Speech Synthesis Workshop: 22-25 May 2007 5. Speech Recognition Final (revised) paper: 25 June 2007 6. Speaker Identification/ Verification 7. Speech Enhancement / Separation 8. Speech Perception 9. Others

ORGANISING COMMITTEE : Mohamed CHETOUANI (UPMC), Bruno GAS (UPMC), Amir HUSSAIN (Stirling), Maurice MILGRAM (UPMC), Jean-Luc ZARADER (UPMC).

CONTACT : [email protected] WEB SITE : http://www.congres.upmc.fr/nolisp2007/ PCS 2007 26th Picture Coding Symposium (www.pcs2007.org)

November 7-9, 2007, Lisbon, Portugal Call for Papers

Picture Coding Symposium (PCS) is an international forum devoted specifically to advancements in visual data coding. Since 1969, PCS has provided the meeting place for the visual coding community: industry, research, academia and users. The 26th PCS will be held in Lisbon, Portugal, on 7-9 November 2007, just before the 16th International Packet Video Workshop that will be held in Lausanne, Switzerland, on 12-13 November 2007.

Topics Organization Topics of interest include, but are not limited to: General Chair Coding of still and moving pictures x Fernando Pereira x Content-based and object-based coding IST-IT, Lisbon, Portugal x Scalable video coding Program Chair x Coding of multiview video and 3D graphics x Modeling and synthetic coding Paulo Lobato Correia x Virtual/augmented reality and telepresence IST-IT, Lisbon, Portugal x Coding for mobile, IP and sensor networks Special Sessions Chair x High fidelity visual data processing and coding Luis Ducla Soares x Analysis for coding and adaptation ISCTE-IT, Lisbon, Portugal x Transcoding and transmoding x Joint audio and visual processing and coding Deadlines x Subjective and objective quality assessment metrics and methods Submission of special sessions x Error robustness, resilience and concealment June 1, 2007 x Coding and indexing for database applications Submission of extended summaries x Protection and integrity of visual data x Persistent association of information to visual data June 10, 2007 x Joint source and channel coding Notification of acceptance x Implementation architectures and VLSI x New applications and techniques for visual data coding September 3, 2007 x Standards for visual data coding Submission of camera-ready papers September 21, 2007

Paper Submission Prospective authors are invited to submit extended summaries of no more than four (4) pages, in English, with font size 11, including results, figures and references. Submissions will be accepted only in PDF format and should be made using the on-line submission system available at www.pcs2007.org. Special Sessions Submission Proposals for special sessions must include a title, rationale, session outline, session chair, a list of authors who have agree to present a paper, and a tentative title and abstract of each paper. Proposals should be sent to the Special Session Chair at [email protected] before June 1, 2007.

Accepted papers will be published in the Workshop CD Proceedings. 2007 International Waveform Diversity & Design Conference

June 4 - 8, 2007 Pisa, Italy

Call for Papers Organizing Committee General Chairs Recent advances in hardware technology are enabling a much Prof. Vinny Amuso, USA Prof. Maria Greco, Italy wider range of design freedoms to be explored for sensor and communication systems. As a result, there are emerging and Technical Chairs compelling changes in system requirements such as more Dr. Eric Mokole, USA Prof. Chris Baker, UK efficient spectrum usage, higher sensitivities, transmitter/receiver agility, greater information content, improved robustness to errors, Publicity Chair Prof. Shannon Blunt, USA etc. The combination of these is fueling a worldwide interest in the subject of waveform design and the use of waveform diversity Tutorial Chair techniques. This third conference in the on-going series will Maj. Todd Hale, USA continue to build on the success of the previous two conferences Student Involvement Chair by bringing together researchers from numerous diverse Prof. Eli Saber, USA backgrounds and specialties to facilitate the exchange and cross- Awards Chair fertilization of ideas and research. Prof. Hugh Griffiths, UK

The WDD organizing committee invites original contributions to Waveform Diversity and Design in the general areas of Communications, Radar, Sonar, etc. Specifically, topics to be included are:

Radar Systems Multi-function Operation Modulation Schemes Sonar Systems 3G/4G Impulsive Systems Multiple-access Schemes Communication Systems Tomography Multi-user Operation Laser Systems Ultra-wideband Operation Bandwidth-on-Demand Interference Suppression Target Detection Synchronization RF Compatibility Tracking RF Imaging Space-Time Adaptive Processing Interferometry Hardware Efficiency Channel Estimation/Equalization SAR/ISAR Bi-static/Multi-static Operation Software Agile Radio/Radar MIMO Communications Sensor Fusion Passive Sensing Operation RF Hitchhiking Polarimetry Target-adaptive Matched Filtering Error Correction Coding EM Phenomenology

Abstracts of 1,000-1,500 words are solicited which should include examples of data and illustrations. Send abstracts to the Conference organizer at [email protected] in Word 97 or later, or (preferably) PDF format before 8 December 2006. Receipt of abstracts will be acknowledged by e-mail. Conference organizer contact: Patricia Woodard, (315) 330-2215. Additional information is available at http://www.waveformdiversity.org. Authors of accepted papers will be notified by 9 February 2007 and will receive instructions for publication at that time. Complete papers of a maximum of five pages (including text and illustrations) will be required by 30 March 2007. Dates to Remember: Abstracts Due 8 Dec 2006

Notification of Acceptance of Papers 9 Feb 2007

Final Papers Due 30 Mar 2007 WIAMIS 2007 8th International Workshop on Image Analysis for Multimedia Interactive Services http://mkg.iti.gr/wiamis2007 June 6-8, 2007,Petros Nomikos Conference Center, Santorini Island, Greece

General Chairs Yiannis Kompatsiaris First Call for Papers Informatics and Telematics Institute, Greece The International Workshop on Image Analysis for Multimedia Interactive Services Yannis Avrithis (WIAMIS) is one of the main international fora for the presentation and discussion National Technical University of Athens, of the latest technological advances in interactive multimedia services. The Greece objective of the workshop is to bring together researchers and developers from academia and industry working in all areas of image, video and audio applications, Special Sessions Chair with a special focus on analysis. After Louvain (1997), Berlin (1999), Tampere Murat Tekalp (2001), London (2003), Lisboa (2004), Montreux (2005), Incheon (2006), College of Engineering, Koc University, WIAMIS2007 is held in Santorini, Greece. Turkey

Technical Program Committee (tentative) Prof. Touradj Ebrahimi EPFL, Switzerland Prof. Moncef Gabbouj Topics of interest include, but are not limited to: Tampere University of Tech., Finland Multimedia content analysis and understanding Dr. Paola Hobson Content-based browsing, indexing and retrieval of images, video and audio Motorola Labs, UK 2D/3D feature extraction Prof. Aggelos K. Katsaggelos Advanced descriptors and similarity metrics for audio and video Northwestern University, USA Relevance feedback and learning systems Prof. Dong Yoon Kim Segmentation of objects in 2D/3D image sequences Ajou University, Korea Identification and tracking of regions in scenes Stefanos Kollias Voice/audio assisted video segmentation National Technical University of Athens, Analysis for coding efficiency and increased error resilience Greece Analysis and understanding tools for content adaptation Prof. Benoit Macq Multimedia content adaptation tools, transcoding and transmoding Universite Catholique de Louvain, Belgium Content summarization and personalization strategies Dr. Jan Nesvadba Data hiding and copyright protection of multimedia content Philips Research Laboratories Eindhoven, Semantic mapping and ontologies The Netherlands Multimedia analysis for advanced applications Prof. Fernando Pereira Multimedia analysis hardware and middleware IST, Portugal Prof. Thomas Sikora Technical University Berlin, Germany Paper Submission Prof. Michael G. Strinztis Prospective contributors are invited to submit extended summaries electronically Informatics and Telematics Institute, using the on-line submission interface, following the instructions available at Greece http://mkg.iti.gr/wiamis2007. Extended summaries should be in Adobe PDF format, written in English, with no more than four pages including figures, using a font size of 11. Important Dates 26 January 2007 Paper Submission 9 March 2007 Notification of acceptance 16 April 2007 Submission of camera-ready papers 2006 INTERNATIONAL WORKSHOP ON ACOUSTIC ECHO AND NOISE CONTROL IWAENC2006

Télécom Paris (ENST), Paris, France September 12 (Tue) - 14 (Thu), 2006 http://www.iwaenc06.enst.fr/

CALL FOR PAPERS The 10th International Workshop on Acoustic Echo and Noise Control will be held in Paris, France, September 12-14, 2006. Starting form 2006, this workshop will be held on even years (since the creation in 1989, the nine previous editions had been held on odd years). The workshop will focus on the topics of signal processing for acoustic echo and noise control. Applications are, among others, telephony, conferencing systems and voice control systems. The three day Organisation committee program includes poster presentations of recent work and latest results, keynote talks and demonstrations. Yves Grenier, Télécom Paris, France TECHNICAL SCOPE Gaël Richard, The technical scope of the workshop includes signal processing for speech enhancement like Télécom Paris, France Catherine Vazza, through the control of acoustic echo and the reduction of noise. These areas of signal processing Télécom Paris, France have been of wide academic interest for many years and have, more recently, seen a substantial Dominique Asselineau, growth in industrial applications such as hands-free mobile telephones and video-conferencing Télécom Paris, France systems. The relevant topics for the workshop include: Technical Committee - Adaptive filtering algorithms and - Systems for stereophonic echo and noise Jacob Benesty, structures for echo and noise control control Univ. Quebec, Canada - Noise reduction techniques - Microphone arrays and array signal Alberto Carini, - Active noise control, sound reproduction processing Univ. Urbino, Italy and hearing aids - Sound enhancement and sound separation Tomas Gaensler, - Transducers and acoustic front-ends - Temporal segmentation of signals Agere, USA - Hardware and real-time issues - Voice activity detection and double-talk Sharon Gannot - Speech-databases and software tools detection Bar-Llan Univ., Israel - Multirate filter banks and subband systems - Noise and acoustic environments and Andre Gilloire, characteristics FT R&D, France Steven Grant, Univ. Missouri-Rolla, USA Yves Grenier, DEMONSTRATIONS Télécom Paris, France Eberhard Haensler, You are strongly encouraged to give a relevant demonstration of your work Darmstadt Univ. of Tech., PROCEDURES Germany Yoichi Haneda, Authors are invited to propose papers in any of the technical areas relevant to the workshop. NTT SP Labs, Japan The technical committee will select papers for poster-presentation. A copy of each paper will be Kees Janse published on line at the time of the workshop. Philips Research, Eindhoven, To submit a proposal: The Netherlands Prepare 2 page paper summary or 4 page paper. The submission should be made through on-line. Walter Kellermann, If a demonstration of the work is planned, include a brief description of the demonstration in the Univ. of Erlangen, Germany paper summary. Please use our LaTeX style file and have a look at a sample paper in both tex and Shoji Makino, pdf for the design of your paper. NTT CS Labs, Japan Rainer Martin, SCHEDULE Rurh Univ., Germany Submission of 2 page paper summary or 4 page paper by: April 21, 2006 Marc Moonen, Notification of acceptance by: June 20, 2006 Katholieke Universiteit, Submission of 4 page final paper (as pdf-file only, at max. 10 MB !) by: July 31, 2006 Leuven, Belgium Patrick Naylor, BEST PAPER AWARD Imperial College, London, UK Piet Sommen, The IWAENC Technical Committee will award a prize for the best student paper. The selection of Eindhoven Univ., The the best paper will be based on originality, scientific merit of the research, and the quality of Netherlands the submitted paper. Peter Vary CONTACT ADDRESS FOR SUBMISSIONS AND MORE INFORMATION RWTH Aachen, Germany [email protected] General -Chair Stefanos Kollias, Preliminary Call for Papers National Technical University of Athens (NTUA), Greece Co-Chair International Conference on Artificial Neural Andreas Stafylopatis, Network (ICANN 06) NTUA, Greece Program Chair Wlodzislaw Duch, Torum, PL & Singapore; ENNS President-elect Erkki Oja, Helsinki, Fi; ENNS President Honorary Chair John G. Taylor, Kings College, London, 10-14 September 2006 UK; ENNS Past President Holiday Inn Hotel, Athens, Greece International Program Committee x Peter Andreas, U. Newcastle, UK Conference Framework x Panos Antsaklis, U. N. Dame, USA The 16th International Conference on Artificial Neural Networks, ICANN 2006, will be held x Nikolaos Bourbakis, Wright State from September 10 to September 14, 2006, at the Holiday Inn Hotel, Athens Greece. ICANN Univ., USA is an annual conference organized by the European Neural Network Society in cooperation x Peter Erdi, Univ. Budapest, HU & with the International Neural Network Society, Japanese Neural Network Society, and the Kalamazoo IEEE Computational Intelligence Society, and is a premier European event in all topics related x Georg Dorffner, Univ. Wien, AT to neural networks. x Erol Gelenbe, Imperial College ICANN 2006 (www.icann2006.org) welcomes contributions on the theory, algorithms, London, UK applications and implementations in the following broad areas: x Stan Gielen, Univ. Nijmegen, NL x Computational neuroscience; x Nikola Kasabov, Kedri, AUT, NZ x Connectionist cognitive science; x Janusz Kacprzyk, Warsaw, PL x Data analysis and pattern recognition; x Chris Koutsougeras, Tulane x Graphical networks models, Bayesian networks; University, USA x Hardware implementations and embedded systems; x Thomas Martinetz, Luebeck, DE x Neural and hybrid architectures and learning algorithms; x Evangelia Micheli-Tzanakou, x Neural control, reinforcement learning and robotics applications; Rutgers University, USA x Neuroinformatics; x Lars Niklasson, Skövde SE x Neural dynamics and complex systems; x Marios Polycarpou, Univ. of x Real world applications; Cyprus x Robotics, control, planning; x Demetris Psaltis, Caltech, USA x Signal and time series processing; x Olli Simula, Espoo, FI x Self-organization; x Alessandro Sperduti, U.Padova, IT x Vision and image processing; x Lefteris Tsoukalas, Purdue x Web semantics; University, USA x Intelligent Multimedia and the Semantic Web. , x Michel Verleysen Louvain-la- Ideas and nominations for interesting tutorials, special sessions, workshops and experts willing Neuve, BE to organize various session tracks are called for. Most active experts will be included in the x Alessandro Villa, U. Grenoble, FR scientific committee of the conference. Local Organizing Committee Proceedings of ICANN will be published in Springer's "Lecture Notes in Computer Science". Paper length is restricted to a maximum of 10 pages, including figures. x Yannis Avrithis, NTUA x Christos Douligeris, Piraeus Univ Deadlines and Conference dates x George Dounias, Aegean Univ x Kostas Karpouzis, ICCS-NTUA 06.01 Submission page opens x Aris Likas, Univ. of Ioannina 14.03 End of submission of papers (abstract+full paper) to regular sessions x Konstantinos Margaritis, Univ. of 30.03 End of submission of papers to special sessions Macedonia 30.04 Acceptance/rejection notification x Stavros Perantonis, NCSR 15.06 Deadline for camera ready papers Demokritos, Athens 01.07 Deadline for early registration x Yannis Pitas, AUTH, Salonica 10.09 Tutorials - first day of the conference x Kostas Pattichis, Univ. of Cyprus 11-13.09 Main part of the conference 14.09 Workshops x Apostolos Paul Refenes, Athens University Economics & Business For further information and/or contacts, send inquiries to x Christos Schizas, Univ. of Cyprus Prof. Stefanos Kollias ([email protected]) x Giorgos Stamou, ICCS-NTUA Prof Andreas Stafylopatis ([email protected]) x Sergios Theodoridis, UoA School of Electrical & Computer Engineering x Spyros Tzafestas, NTUA National Technical University of Athens x Mihalis Zervakis, TUC, Crete 9, Heroon Polytechniou str., 157 80 Zografou, Athens, Greece. The International ITG / IEEE Workshop on Smart Antennas WSA 2007 February 26-27, 2007 Vienna Call for Papers The International ITG / IEEE Workshop on Smart Antennas WSA 2007 provides a forum for presentation of the most recent research on smart antennas. The objective is to continue, accelerate, and broaden the momentum already gained with a series of ITG Workshops held since 1996: Munich and Zurich’96, Vienna and Kaiserslautern’97, Karlsruhe’ 98, Stuttgart’99, Ilmenau’01, Munich’04, Duisburg’05, and Ulm’06. This call for papers intends to solicit contributions on latest research of this key technology for wireless communication systems. . Workshop topics include, but are not limited to: - Antennas for beamforming and diversity - Multicarrier MIMO - Channel measurements - Multiuser MIMO - Spatial channel modeling - Cooperative and sensor networks - Beamforming - Crosslayer optimisation - Diversity concepts - Radio resource management - Space-time processing - Cellular systems - Space-time codes - Link, system and network level simulations - MIMO Systems - Hard- and software implementation issues There will be oral as well as poster presentations. The workshop will be jointly organized by the Institute of Communications and Radio Frequency at Vienna University of Technology and the ftw. Telecommunications Research Center Vienna in cooperation with the VDE, ÖVE, and the IEEE on February 26-27, 2007 in Vienna, Austria Organizers and Workshop Chairs Markus Rupp, E-Mail: [email protected] Christoph Mecklenbräuker, E-Mail: [email protected] Information about the workshop can soon be found at: http://www.ftw.at/ Technical program committee Jørgen Bach-Andersen David Gesbert Michael Meurer Werner Teich Sergio Barbarossa Martin Haardt Werner Mohr Reiner Thomä Ezio Biglieri Dirk Heberling Ralf Müller Wolfgang Utschick Holger Boche Ari Hottinen Josef A. Nossek Alle-Jan van der Veen Helmut Bölcskei Thomas Kaiser Björn Ottersten Mats Viberg Ernst Bonek Anja Klein Steffen Paul Emanuele Viterbo Andreas Czylwik Miguel Lagunas Ana I. Pérez-Neira Tobias Weber Armin Dekorsy Geert Leus Arpad L. Scholtz Joachim Wehinger Gerhard Fettweis Jürgen Lindner Nikos Sidiropoulos Werner Wiesbeck Bernard H. Fleury Gerald Matz Klaus Solbach Thomas Zemen Javier Fonollosa Utz Martin Michael Tangemann Abdelhak Zoubir Alex Gershman Tadashi Matsumoto Giorgio Taricco EURASIP JOURNALS

EURASIP JOURNAL ON APPLIED SIGNAL PROCESSING

Scope The overall aim of EURASIP Journal on Applied Signal Processing (EURASIP JASP) is to bring science and applications together with emphasis on practical aspects of signal processing in new and emerging technologies. It is directed as much at the practicing engineers as at the academic researchers. EURASIP JASP will highlight the diverse applications of signal processing and encourage a cross fertilization of techniques. All papers should attempt to bring theory to life with practical simulations and examples. Tutorial articles on topics of interest are also welcomed. EURASIP JASP employs paperless, electronic review process to foster fast and speedy turnaround in review process. There are two diﬀerent issues: regular issues and special issues. The regular issues publish collections of papers without special solicitation. The special issues have speciﬁcally aimed and targeted topics of interest contributed by authors responding to a particular Call-for- Papers or by invitation, edited by invited guest editor(s). Regular papers can be submitted at any time, while special issue papers can be submitted only based on planned schedules and submission guidelines of the Call-for-Papers. Proposals for special issues can be submitted directly to the Editor-in-Chief.

Subjects Subject areas include, but are by no means limited to: • Signal processing theory, algorithm, architecture, design, and implementation • Speech processing, coding, compression, and recognition • Audio signal processing, coding, and compression • Image/video processing, coding, compression, restoration, analysis and understanding, and communications • Multimedia signal processing and technology • Signal processing for communications and networking • Statistical and adaptive signal processing • Nonlinear signal processing techniques • Signal processing design tools • Signal processing for security, authentication, and cryptography • Analog signal processing • Signal processing for smart sensor and systems EURASIP Journal on Applied Signal Processing 73

Application areas include, but not limited to: communications; networking; sensors and actuators; radar and sonar; medical imaging; biomedical applications; remote sensing; consumer electronics; computer vision; pattern recognition; robotics; ﬁber optic sensing/transducers; industrial automation; transportation; stock market and ﬁnancial analysis; seismography; avionics.

Indexed/Abstracted In The articles of the EURASIP JASP are reviewed/indexed in Acoustics Abstracts; Computer and Communications Security Abstracts (CCSA); CompuMath Citation Index; Current Contents: Engineering, Computing & Technology; CSA Engineering Research Database; CSA High Technology Research Database with Aerospace; CSA High Technology Research Database with Metadex; Engineering Information databases including (Compendex and Paperchem); INIST-CNRS (Pascal Database); INSPEC; JournalSeek Database; Mathemat- ical Reviews; Science Citation Index Expanded; Scopus; Technology and Management (TEMA); and Zentralblatt fur¨ Mathematik.

Editor-in-Chief Ali H. Sayed, Electrical Engineering Dept., University of California, Los Angeles, CA 90095, USA EURASIP JOURNALS

EURASIP JOURNAL ON AUDIO, SPEECH, AND MUSIC PROCESSING

Scope The aim of “EURASIP Journal on Audio, Speech, and Music Processing” (EURASIP JASMP) is to bring together researchers and engineers working on the theory and applications of EURASIP Journal on Audio, Speech, and Music Processing. EURASIP JASMP will be an interdisciplinary journal for the dissemination of all basic and applied aspects of speech communication and audio processes. Its primary objectives are: • To publish papers on the advancement of both human speech communication science and automatic speech and audio systems • To allow rapid and wide diﬀusion of excellent contributions in these areas • To provide world-wide, barrier-free access to the full text of research articles • To conduct a rapid but thorough review process in order to assure high quality papers • To provide immediate web access once a paper is editorially approved The journal will be dedicated to having original research work, but will also allow tutorial and review articles. Articles will deal with both theoretical and practical aspects of EURASIP Journal on Audio, Speech, and Music Processing.

Subjects Subject areas include (but are not limited to): • Speech and audio technology, as well as related science and engineering methods • Speech analysis, synthesis, coding, recognition, speaker veriﬁcation, language modeling and recognition, human speech production and perception, speech enhancement • Room acoustics, human audition, analysis, synthesis, and coding of music and other audio, transducers, active sound and noise control • Speech and audio separation, computational auditory scene analysis and independent component analysis

Editor-in-Chief Douglas O’Shaughnessy, INRS-EMT (Place Bonaventure), 800 de la Gauchetiere west, suite 6900, Montreal, Quebec, Canada H5A 1K6 EURASIP JOURNALS

EURASIP JOURNAL ON BIOINFORMATICS AND SYSTEMS BIOLOGY

Scope The overall aim of “EURASIP Journal on Bioinformatics and Systems Biology” (EURASIP JBSB) is to publish research results related to signal processing and bioinformatics theories and techniques relevant to a wide area of applications into the core new disciplines of genomics, proteomics, and systems biology. The journal is intended to offer a common platform for scientists from several areas including signal processing, bioinformatics, statistics, biology and medicine, who are interested in the development of algorithmic, mathematical, statistical, modeling, simulation, data mining, and computational techniques, as demanded by various applications in genomics, proteomics, system biology, and more general in health and medicine. Papers should emphasize original results related to the theoretical and algorithmic aspects of signal processing and bioinformatics, in close connection with the applications to genomics, proteomics, systems biology and medicine. Tutorial papers, especially those emphasizing strong components of signal processing or bioinformatics in multidisciplinary views of genomics, proteomics and systems biology are also welcome. The journal will em- brace a wide range of topics, and will accommodate different exposition styles, to help scientists with various backgrounds, e.g., engineering, bioinformatics, or biology, to interact effortlessly and to facilitate the exchange of information across the multidisciplinary areas involved. EURASIP JBSB employs a paperless, electronic submission and evaluation system to promote a rapid turnaround in the peer review process. The journal publishes two types of issues: regular issues and special issues. Regular issues publish collections of papers without special solicitation. The special issues feature specifically aimed and targeted topics of interest contributed by authors responding to a particular Call-for-Papers or by invitation, edited by invited guest editor(s). Regular papers can be submitted at any time, while special issue papers can be submitted only based on planned schedules and submission guidelines of the Call-for-Papers. Proposals for special issues can be submitted directly to the Editor-in-Chief.

Subjects Subject areas include (but are by no means limited to): • Reverse engineering of biological circuits • Data mining methods for genomics and proteomics • Signal Processing theory and techniques for systems biology • Modeling and simulation of biological networks • Nanotechnology in genomics and proteomics • Signal processing methods in sequence analysis 76 EURASIP Journals

• Information theoretic approaches to genomics and proteomics • Microarray image and data analysis • Noise models in high-throughput technologies • Integration of heterogeneous data

Editor-in-Chief Ioan Tabus, Room TF414 (Tietotalo), Institute of Signal Processing Tampere University of Techology, Korkeakoulunkatu 1, P.O. Box 553, FIN-33101, Tampere, Finland EURASIP JOURNALS

EURASIP JOURNAL ON EMBEDDED SYSTEMS

Scope “EURASIP Journal on Embedded Systems” is an international journal that serves the large community of researchers and professional engineers who deal with the theory and practice of embedded systems, particularly encompassing all practical aspects of theory and methods used in designing homogeneous as well as heterogeneous embedded systems that combine data-driven and control-driven behaviors. There are two diﬀerent issues: regular issues and special issues. The regular issues publish collections of papers without special solicitation. The special issues have speciﬁcally aimed and targeted topics of interest contributed by authors responding to a particular Call-for- Papers or by invitation, edited by invited guest editor(s). Regular papers can submitted at any time, while special issue papers can be submitted only based on planned schedules and submission guidelines of the Call-for-Papers. Proposals for special issues can be submitted directly to Editor-in-Chief.

Subjects Original full and short papers, correspondence and reviews on design and development of embedded systems, methodologies applied for their specification, modeling and design, and adaptation of algorithms for real-time execution are encouraged for submission. The coverage includes complex homogeneous and heterogeneous embedded systems, specification languages and tools for embedded systems, modeling and verification techniques, hardware/software trade-offs and co-design, new design flows, design methodologies and synthesis methods, platform-based design, component-based design, adaptation of signal processing algorithms to limited implementation resources, rapid prototyping, computing structures and architectures for complex embedded systems, real-time operating systems, methods and techniques for the design of low-power systems, interfacing with the real world, novel application case studies and experiences, and does not exclude other interesting related and emerging topics like software defined radio. Example applications include wireless and data communication systems, speech processing, image and video-processing, digital signal processing applications as well as control and instrumentation.

Editor-in-Chief Zoran Salcic, University of Auckland, Department of Electrical and Computer Engineering, Science Centre (Building 303, level 2, room 242), Private Bag 92019, 38 Princess Street, Auckland, New Zealand EURASIP JOURNALS

EURASIP JOURNAL ON WIRELESS COMMUNICATIONS AND NETWORKING

Scope The overall aim of EURASIP Journal on Wireless Communications and Networking (EURASIP JWCN) is to bring science and applications together on wireless communications and networking technologies with emphasis on signal processing techniques and tools. It is directed at both practicing engineers and academic researchers. EURASIP JWCN highlights the continued growth and new challenges in wireless technology, both for application development and basic research. Papers should emphasize original results relating to the theory and/or applications of wireless communications and networking. Tutorial papers, especially those emphasizing multidisciplinary views of communications and networking, are also welcomed. EURASIP JWCN employs a paperless, electronic submission and evaluation system to promote a rapid turnaround in the peer review process. The journal publishes two types of issues: regular issues and special issues. Regular issues publish collections of papers without special solicitation. The special issues feature speciﬁ- cally aimed and targeted topics of interest contributed by authors responding to a particular Call-for-Papers or by invitation, edited by invited guest editor(s). Regular papers can be submitted at any time, while special issue papers can be submitted only based on planned schedules and submission guidelines of the Call-for-Papers. Proposals for special issues can be submitted directly to the Editor-in-Chief.

Subjects Subject areas include, but are by no means limited to: Ad hoc networks; Channel modeling and propagation; Detection, estimation, and synchronization; Diversity and space- time techniques; End-to-end design techniques; Error control coding; Iterative techniques for joint optimization; Modulation techniques (CDMA, OFDM, multicarrier, spread- spectrum, etc.); Multiuser, MIMO channels, and multiple access schemes; Network performance, reliability, and quality of service; Resource allocation over wireless networks; Security, authentication, and cryptography; Signal processing techniques and tools; Ultra wideband systems; Wireless network services and medium access control.

Editor-in-Chief Phillip Regalia, Institut National des Tel´ ecommunications,´ 9 rue Charles Fourier, F-91011 Evry Cedex, France EURASIP JOURNALS

SIGNAL PROCESSING

Editorial Policy Signal Processing is an interdisciplinary journal presenting the theory and practice of signal processing. Its primary objectives are the following: • dissemination of research results and of engineering developments to all signal processing groups and individuals; • presentation of practical solutions to current signal processing problems in engineering and science. The editorial policy and the technical content of the journal are the responsibility of the Editor-in-Chief and the Editorial Board. The journal is self-supporting from the subscription income and contains a minimum amount of advertisements. Advertisements are subject to the prior approval of the Editor-in-Chief. The journal welcomes contributions from every country in the world.

Scope Signal Processing incorporates all aspects of the theory and practice of signal processing (analogue and digital). It features original research work, tutorial and review articles, and accounts of practical developments. It is intended for a rapid dissemination of knowledge and experience to engineers and scientists working on signal processing research, development, or practical application.

Subjects Subject areas covered by the journal include: Signal Theory; Stochastic Processes; Detection and Estimation; Spectral Analysis; Filtering; Communication Signal Processing; Biomedical Signal Processing; Geophysical and Astrophysical Signal Processing; Earth Resources Signal Processing; Acoustic and Vibration Signal Processing; Signal Processing Systems; Software Developments; Image Processing; Pattern Recognition; Optical Signal Processing; Multidi- mensional Signal Processing; Data Processing; Remote Sensing; Signal Processing Technol- ogy; Speech Processing; Radar Signal Processing; Sonar Signal Processing; Special Signal Processing; Industrial Applications; New Applications.

Editor-in-Chief B. Ottersten, Royal Institute of Technology, Stockholm, Sweden EURASIP JOURNALS

SIGNAL PROCESSING: IMAGE COMMUNICATION

Editorial Policy Signal Processing: Image Communication is an international journal for the development of the theory and practice of image communication. Its primary objectives are the following: • to present a forum for the advancement of the theory and practice of image communication; • to simulate cross fertilization between areas similar in nature which have traditionally been separated, for example, various aspects of visual communications and information systems; • to contribute to a rapid information exchange between the industrial and academic environments. The editorial policy and the technical content of the journal are the responsibility of the Editor-in-Chief and the Editorial Board. The journal is self-supporting from the subscription income and contains a minimum amount of advertisements. Advertisements are subject to the prior approval of the Editor-in-Chief. The journal welcomes contributions from every country in the world.

Scope Signal Processing: Image Communication publishes articles relating to aspects of design, implementation, and use of image communication systems. Signal Processing: Image Com- munication features original research work, tutorial and review articles, and accounts of practical developments.

Subjects Subject areas covered by the journal include: TV, HDTV, and 3DTV systems; Visual Sci- ence; Image; TV and Advanced TV; Broadcasting; Image Storage and Retrieval; Graphic Arts; Electronic Printing; Image Transmission; Interactive Image Coding Communication; Imaging Technology; Display Technology; VLSI Processors for Image Communications.

Editor-in-Chief M. Tekalp, Koc¸ University, College of Engineering, Rumelifeneri Yolu, 34450 Sariyer, Istan- bul, Turkey EURASIP JOURNALS

SPEECH COMMUNICATION

Editorial Policy The journal’s primary objectives are the following: • to present a forum for the advancement of human and human-machine speech communication science; • to stimulate cross fertilization between different fields of this domain; • to contribute towards the rapid and wide diffusion of scientifically sound contributions in this domain. Speech Communication is an interdisciplinary journal whose primary objective is to fulfill the need for the rapid dissemination and thorough discussion of basic and applied research results. In order to establish frameworks of inter-relate results from the various areas of the field, emphasis will be placed on viewpoints and topics of a transdisciplinary nature. The editorial policy and the technical content of the journal are the responsibility of the Editors and the Institutional Representatives. The Institutional Representatives assist the Editors in the definition and the control of editorial policy as well as in maintaining connections with scientific associations, international congresses, and regional events. The Editorial Board contributes towards the gathering of material for publication and assists the Editors in the editorial process.

Scope Speech Communication is an interdisciplinary journal for the development and dissemination of all basic and applied aspects of speech communication processes. Speech Com- munication features original research work, tutorial and review articles dealing with the theoretical, empirical, and practical aspects of this scientiﬁc ﬁeld.

Subject Coverage Subject areas covered in this journal include: • Basics of oral communication and dialogue: modelling of production and perception processes; phonetics and phonology; syntax; semantics of speech communication; cognitive aspects. • Models and tools for language learning: functional organisation and developmen- tal models of human language capabilities; acquisition and rehabilitation of spoken language; speech and hearing defects and aids. • Speech signal processing: analysis; coding; transmission; enhancement, robustness to noise. • Models for automatic speech communication: speech recognition; language identiﬁ- cation; speaker recognition; speech synthesis; oral dialogue. 82 EURASIP Journals

• Development and evaluation tools: monolingual and multilingual databases; assessment methodologies; specialised hardware and software packages; ﬁeld experiments; market development. • Multimodal human-computer interface: using speech I/O in combination with modalities, for example, gesture and handwriting.

Editors-in-Chief J.-L. Gauvain, Universite´ Paris-Sud, LIMSI-CNRS, Orsay Cedex, France J. Hirschberg, Columbia University, Department of Computer Science, 1214 Amsterdam Ave., M/C 0401, 450 CS Building, New York, NY 10027, USA K. Paliwal, Griﬃth University, School of Microelectronic Engineering, Brisbane, QLD 4111, Australia EURASIP Journal on Applied Signal Processing, Volume 2006 © 2006 Hindawi Publishing Corporation

Volume 2006 Contents and Abstracts

FPGA Prototyping of RNN Decoder for Convolutional Codes Zoran Salcic, Stevan Berber, and Paul Secker DOI: 10.1155/ASP/2006/15640 This paper presents prototyping of a recurrent type neural network (RNN) convolutional decoder using system-level design specification and design flow that enables easy mapping to the target FPGA architecture. Implementation and the performance measurement results have shown that an RNN decoder for hard-decision decoding coupled with a simple hard-limiting neuron activation function results in a very low complexity, which easily fits into standard Altera FPGA. Moreover, the design methodology allowed modeling of complete testbed for prototyping RNN decoders in simulation and real-time environment (same FPGA), thus enabling evaluation of BER performance characteristics of the decoder for various conditions of communication channel in real time.

A Human Body Analysis System Vincent Girondel, Laurent Bonnaud, and Alice Caplier DOI: 10.1155/ASP/2006/61927 This paper describes a system for human body analysis (segmentation, tracking, face/hands localisation, posture recognition) from a single view that is fast and completely automatic. The system first extracts low-level data and uses part of the data for high-level interpretation. It can detect and track several persons even if they merge or are completely occluded by another person from the camera’s point of view. For the high-level interpretation step, static posture recognition is performed using a belief theory-based classifier. The belief theory is considered here as a new approach for performing posture recognition and classification using imprecise and/or conflicting data. Four different static postures are considered: standing, sitting, squatting, and lying. The aim of this paper is to give a global view and an evaluation of the performances of the entire system and to describe in detail each of its processing steps, whereas our previous publications focused on a single part of the system. The efficiency and the limits of the system have been highlighted on a database of more than fifty video sequences where a dozen different individuals appear. This system allows real- time processing and aims at monitoring elderly people in video surveillance applications or at the mixing of real and virtual worlds in ambient intelligence systems. 84 Regular Issue, Vol. 2006

An FIR Notch Filter for Adaptive Filtering of a Sinusoid in Correlated Noise Osman Kukrer and Aykut Hocanin DOI: 10.1155/ASP/2006/38190 A novel adaptive FIR filter for the estimation of a single-tone sinusoid corrupted by additive noise is described. The filter is based on an offline optimization procedure which, for a given notch frequency, computes the filter coefficients such that the frequency response is unity at that frequency and a weighted noise gain is minimized. A set of such coefficients is obtained for notch frequencies chosen at regular intervals in a given range. The filter coefficients corresponding to any frequency in the range are computed using an interpolation scheme. An adaptation algorithm is developed so that the filter tracks the sinusoid of unknown frequency. The algorithm first estimates the frequency of the sinusoid and then updates the filter coefficients using this estimate. An application of the algorithm to beamforming is included for angle-of-arrival estimation. Simulation results are presented for a sinusoid in correlated noise, and compared with those for the adaptive IIR notch filter.

Efficient Realization of Sigma-Delta (Σ-Δ) Kalman Lowpass Filter in Hardware Using FPGA Saman S. Abeysekera and Charayaphan Charoensak DOI: 10.1155/ASP/2006/52736 Sigma-delta (Σ-Δ) modulation techniques have moved into mainstream applications in signal processing and have found many practical uses in areas such as high-resolution A/D, D/A conversions, voice communication, and software radio. Σ-Δ modulators produce a single, or few bits output resulting in hardware saving and thus making them suitable for implementation in very large scale integration (VLSI) circuits. To reduce quantization noise produced, higher-order modulators such as multiloop and multistage architectures are commonly used. The quantization noise behavior of higher-order Σ-Δ modulators is well understood. Based on these quantization noise characteristics, various demodulator architectures, such as sinc filter, optimal FIR filter, and Laguerre filter are reported in literature. In this paper, theory and design of an efficient Kalman recursive demodulator filter is shown. Hardware implementation of Kalman lowpass filter, using field programmable gate array (FPGA), is explained. The FPGA synthesis results from Kalman filter design are compared with previous designs for sinc filter, optimum FIR filter, and Laguerre filter.

An Exact FFT Recovery Theory: A Nonsubtractive Dither Quantization Approach with Applications L. Cheded and S. Akhtar DOI: 10.1155/ASP/2006/34838 Fourier transform is undoubtedly one of the cornerstones of digital signal processing (DSP). The introduction of the now famous FFT algorithm has not only breathed a new lease of life into an otherwise latent classical DFT algorithm, but also led to an explosion in applications Regular Issue, Vol. 2006 85 that have now far transcended the confines of the DSP field. For a good accuracy, the digital implementation of the FFT requires that the input and/or the 2 basis functions be finely quantized. This paper exploits the use of coarse quantization of the FFT signals with a view to further improving the FFT computational efficiency while preserving its computational accuracy and simplifying its architecture. In order to resolve this apparent conflict between preserving an excellent computational accuracy while using a quantization scheme as coarse as can be desired, this paper advances new theoretical results which form the basis for two new and practically attractive FFT estimators that rely on the principle of 1 bit nonsubtractive dithered quantization (NSDQ). The proposed theory is very well substantiated by the extensive simulation work carried out in both noise-free and noisy environments. This makes the prospect of implementing the 2 proposed 1 bit FFT estimators on a chip both practically attractive and rewarding and certainly worthy of a further pursuit.

Shape-from-Shading for Oblique Lighting with Accuracy Enhancement by Light Direction Optimization Osamu Ikeda DOI: 10.1155/ASP/2006/92456 We present a shape-from-shading approach for oblique lighting with accuracy enhancement by light direction optimization. Based on an application of the Jacobi iterative method to the consistency between the reflectance map and image, four surface normal approxima- tions are introduced and the resulting four iterative relations are combined as constraints to get an iterative relation. The matrix that converts the shading information to the depth is modified so as to be uniform over the whole image region, making the iteration stable and, as a result, the resulting shape more accurate. Then, to enhance the accuracy, the light direction is optimized for slant angle using two criteria based on the initial boundary value and the rank of the converting matrix. The method is examined using synthetic and real images to show that it is superior to the current state-of-the-art methods and that it is effective for oblique light direction whose slant angle ranges from 55 to 75 degrees.

Time-Frequency Signal Synthesis and Its Application in Multimedia Watermark Detection Lam Le and Sridhar Krishnan DOI: 10.1155/ASP/2006/86712 We propose a novel approach to detect the watermark message embedded in images under the form of a linear frequency modulated chirp. Localization of several time-frequency distributions (TFDs) is studied for diﬀerent frequency modulated signals under various noise conditions. Smoothed pseudo-Wigner-Ville distribution (SPWVD) is chosen and applied to detect and recover the corrupted image watermark bits at the receiver. The synthesized watermark message is compared with the referenced one at the transmitter as a detection evaluation scheme. The correlation coeﬃcient between the synthesized and the referenced chirps reaches 0.9oraboveforamaximumbiterrorrateof15%underintentionaland 86 Regular Issue, Vol. 2006 nonintentional attacks. The method provides satisfactory result for detection of image watermark messages modulated as chirp signal and could be a potential tool in multimedia security applications.

Performance Analysis of the Blind Minimum Output Variance Estimator for Carrier Frequency Offset in OFDM Systems Feng Yang, Kwok H. Li, and Kah C. Teh DOI: 10.1155/ASP/2006/49257 Carrier frequency offset (CFO) is a serious drawback in orthogonal frequency division multiplexing (OFDM) systems. It must be estimated and compensated before demodu- lation to guarantee the system performance. In this paper, we examine the performance of a blind minimum output variance (MOV) estimator. Based on the derived probability density function (PDF) of the output magnitude, its mean and variance are obtained and it is observed that the variance reaches the minimum when there is no frequency offset. This observation motivates the development of the proposed MOV estimator. The theoretical mean-square error (MSE) of the MOV estimator over an AWGN channel is obtained. The analytical results are in good agreement with the simulation results. The performance evaluation of the MOV estimator is extended to a frequency-selective fading channel and the maximal-ratio combining (MRC) technique is applied to enhance the MOV estimator’s performance. Simulation results show that the MRC technique significantly improves the accuracy of the MOV estimator.

Contour Estimation by Array Processing Methods Salah Bourennane and Julien Marot DOI: 10.1155/ASP/2006/95634 This work is devoted to the estimation of rectilinear and distorted contours in images by high-resolution methods. In the case of rectilinear contours, it has been shown that it is possible to transpose this image processing problem to an array processing problem. The existing straight line characterization method called subspace-based line detection (SLIDE) leads to models with orientations and offsets of straight lines as the desired parameters. Firstly, a high-resolution method of array processing leads to the orientation of the lines. Secondly, their offset can be estimated by either the well-known method of extension of the Hough transform or another method, namely, the variable speed propagation scheme, that belongs to the array processing applications field. We associate it with the method called “modified forward-backward linear prediction” (MFBLP). The signal generation process devoted to straight lines retrieval is retained for the case of distorted contours estimation. This issue is handled for the first time thanks to an inverse problem formulation and a phase model determination. The proposed method is initialized by means of the SLIDE algorithm. Regular Issue, Vol. 2006 87

Global Motion Model for Stereovision-Based Motion Analysis Jia Wang, Zhencheng Hu, Keiichi Uchimura, and Hanqing Lu DOI: 10.1155/ASP/2006/53691 An advantage of stereovision-based motion analysis is that the depth information is available, thus motion can be estimated more precisely in 2.5D stereo coordinate system (SCS) constructed by the depth and the image coordinates. In this paper, stereo global motion in SCS, which is induced by 3D camera motion in real-world coordinate system (WCS), is parameterized by a five-parameter global motion model (GMM). Based on such model, global motion can be estimated and identified directly in SCS without knowing the physical parameters about camera motion and camera setup in WCS. The reconstructed global motion field accords with the spatial structure of the scene much better. Experiments on both synthetic data and real-world images illustrate its promising performance.

A Novel Efficient Cluster-Based MLSE Equalizer for Satellite Communication Channels with M-QAM Signaling Eleftherios Kofidis, Vassilis Dalakas, Yannis Kopsinis, and Sergios Theodoridis DOI: 10.1155/ASP/2006/34343 In satellites, nonlinear amplifiers used near saturation severely distort the transmitted signal and cause difficulties in its reception. Nevertheless, the nonlinearities introduced by memoryless bandpass amplifiers preserve the symmetries of the M-ary quadrature amplitude modulation (M-QAM) constellation. In this paper, a cluster-based sequence equalizer (CBSE) that takes advantage of these symmetries is presented. The proposed equalizer exhibits enhanced performance compared to other techniques, including the conventional linear transversal equalizer, Volterra equalizers, and RBF network equalizers. Moreover, this gain in performance is obtained at a substantially lower computational cost.

Generalized Sampling Theorem for Bandpass Signals Ales Prokes DOI: 10.1155/ASP/2006/59587 The reconstruction of an unknown continuously defined function f (t) from the samples of the responses of m linear time-invariant (LTI) systems sampled by the 1/mth Nyquist rate is the aim of the generalized sampling. Papoulis (1977) provided an elegant solution for the case where f (t) is a band-limited function with finite energy and the sampling rate is equal to 2/m times cutoff frequency. In this paper, the scope of the Papoulis theory is extended to the case of bandpass signals. In the first part, a generalized sampling theorem (GST) for bandpass signals is presented. The second part deals with utilizing this theorem 88 Regular Issue, Vol. 2006 for signal recovery from nonuniform samples, and an efficient way of computing images of reconstructing functions for signal recovery is discussed.

Analysis of Effort Constraint Algorithm in Active Noise Control Systems F. Taringoo, J. Poshtan, and M. H. Kahaei DOI: 10.1155/ASP/2006/54649 In ANC systems, in case of loudspeakers saturation, the adaptive algorithm may diverge due to nonlinearity. The most common algorithm used in ANC systems is the FXLMS which is especially used for feed-forward ANC systems. According to its mathematical representation, its cost function is conventionally chosen independent of control signal magnitude, and hence the control signal may increase unlimitedly. In this paper, a modified cost function is proposed that takes into account the control signal power. Choosing an appropriate weight can prevent the system from becoming nonlinear. A region for this weight is obtained and the mean weight behavior of the algorithm using this cost function is achieved. In addition to the previous paper results, the linear range for the effort coefficient variation is obtained. Simulation and experimental results follow for confirmation.

Source Depth Estimation Using a Horizontal Array by Matched-Mode Processing in the Frequency-Wavenumber Domain Barbara Nicolas, Jer´ omeˆ I. Mars, and Jean-Louis Lacoume DOI: 10.1155/ASP/2006/65901 In shallow water environments, matched-field processing (MFP) and matched-mode processing (MMP) are proven techniques for doing source localization. In these environments, the acoustic field propagates at long range as depth-dependent modes. Given a knowledge of the modes, it is possible to estimate source depth. In MMP, the pressure field is typically sampled over depth with a vertical line array (VLA) in order to extract the mode amplitudes. In this paper, we focus on horizontal line arrays (HLA) as they are generally more practical for at sea applications. Considering an impulsive low-frequency source (1– 100 Hz) in a shallow water environment (100–400 m), we propose an efficient method to estimate source depth by modal decomposition of the pressure field recorded on an HLA of sensors. Mode amplitudes are estimated using the frequency-wavenumber transform, which is the 2D Fourier transform of a time-distance section. We first study the robustness of the presented method against noise and against environmental mismatches on simulated data. Then, the method is applied both to at sea and laboratory data. We also show that the source depth estimation is drastically improved by incorporating the sign of the mode amplitudes. Regular Issue, Vol. 2006 89

A Secure Watermarking Scheme for Buyer-Seller Identification and Copyright Protection Fawad Ahmed, Farook Sattar, Mohammed Yakoob Siyal, and Dan Yu DOI: 10.1155/ASP/2006/56904 We propose a secure watermarking scheme that integrates watermarking with cryptography for addressing some important issues in copyright protection. We address three copyright protection issues—buyer-seller identification, copyright infringement, and ownership verification. By buyer-seller identification, we mean that a successful watermark extraction at the buyer’s end will reveal the identities of the buyer and seller of the watermarked image. For copyright infringement, our proposed scheme enables the seller to identify the specific buyer from whom an illegal copy of the watermarked image has originated, and further prove this fact to a third party. For multiple ownership claims, our scheme enables a legal seller to claim his/her ownership in the court of law. We will show that the combination of cryptography with watermarking not only increases the security of the overall scheme, but it also enables to associate identities of buyer/seller with their respective watermarked images.

On the Channel Capacity of Multiantenna Systems Open Access with Nakagami Fading Feng Zheng and Thomas Kaiser DOI: 10.1155/ASP/2006/39436 We discuss the channel capacity of multiantenna systems with the Nakagami fading channel. Analytic expressions for the ergodic channel capacity or its lower bound are given for SISO, SIMO, and MISO cases. Formulae for the outage probability of the capacity are presented. It is shown that the channel capacity could be increased logarithmically with the number of receive antennas for SIMO case; while employing 3–5 transmit antennas (irrespective of all other parameters considered herein) can approach the best advantage of the multiple transmit antenna systems as far as channel capacity is concerned for MISO case. We have shown that for a given SNR, the outage probability decreases considerably with the number of receive antennas for SIMO case, while for MISO case, the upper bound of the outage probability decreases with the number of transmit antennas when the transmission rate is lower than some value, but increases instead when the transmission rate is higher than another value. A critical transmission rate is identiﬁed.

Partial Equalization of Non-Minimum-Phase Impulse Responses Ahfir Maamar, Izzet Kale, Artur Krukowski, and Berkani Daoud DOI: 10.1155/ASP/2006/67467 We propose a modified version of the standard homomorphic method to design a minimum-phase inverse filter for non-minimum-phase impulse responses equalization. In 90 Regular Issue, Vol. 2006 the proposed approach some of the dominant poles of the filter transfer function are replaced by new ones before carrying out the inverse DFT. This method is useful when partial magnitude equalization is intended. Results for an impulse response measured in the car interior show that by using the modified version we can control the sound quality more precisely than when using the standard method.

Fast Registration of Remotely Sensed Images for Earthquake Damage Estimation Arash Abadpour, Shohreh Kasaei, and S. Mohsen Amiri DOI: 10.1155/ASP/2006/76462 Analysis of the multispectral remotely sensed images of the areas destroyed by an earthquake is proved to be a helpful tool for destruction assessments. The performance of such methods is highly dependant on the preprocess that registers the two shots before and after an event. In this paper, we propose a new fast and reliable change detection method for remotely sensed images and analyze its performance. The experimental results show the eﬃciency of the proposed algorithm.

High Efﬁciency EBCOT with Parallel Coding Open Access Architecture for JPEG2000 Jen-Shiun Chiang, Chun-Hau Chang, Chang-Yo Hsieh, and Chih-Hsien Hsia DOI: 10.1155/ASP/2006/42568 This work presents a parallel context-modeling coding architecture and a matching arithmetic coder (MQ-coder) for the embedded block coding (EBCOT) unit of the JPEG2000 encoder. Tier-1 of the EBCOT consumes most of the computation time in a JPEG2000 encoding system. The proposed parallel architecture can increase the throughput rate of the context modeling. To match the high throughput rate of the parallel context-modeling architecture, an eﬃcient pipelined architecture for context-based adaptive arithmetic encoder is proposed. This encoder of JPEG2000 can work at 180 MHz to encode one symbol each cycle. Compared with the previous context-modeling architectures, our parallel architectures can improve the throughput rate up to 25%.

Frequency and 2D Angle Estimation Based on a Sparse Uniform Array of Electromagnetic Vector Sensors Fei Ji and Sam Kwong DOI: 10.1155/ASP/2006/80720 We present an ESPRIT-based algorithm that yields extended-aperture two-dimensional (2D) arrival angle and carrier frequency estimates with a sparse uniform array of electromagnetic vector sensors. The ESPRIT-based frequency estimates are first achieved by using the temporal invariance structure out of the two time-delayed sets of data collected from vector sensor array. Each incident source’s coarse direction of arrival (DOA) estimation is Regular Issue, Vol. 2006 91 then obtained through the Poynting vector estimates (using a vector cross-product estimator). The frequency and coarse angle estimate results are used jointly to disambiguate the cyclic phase ambiguities in ESPRIT’s eigenvalues when the intervector sensor spacing exceeds a half wavelength. Monte Carlo simulation results verified the effectiveness of the proposed method.

Erratum to “A New Class of Particle Filters for Random Dynamic Systems with Unknown Statistics” Joaqu´ın M´ıguez, Monica´ F. Bugallo, and Petar M. Djuric´ DOI: 10.1155/ASP/2006/78708 We have found an error in the proof of Lemma presented in our paper “A New Class of Particle Filters for Random Dynamic Systems with Unknown Statistics” (EURASIP Journal on Applied Signal Processing, 2004). In the sequel, we provide a restatement of the lemma and a corrected (and simpler) proof. We emphasize that the original result in the said paper still holds true. The only diﬀerence with the new statement is the relaxation of condition (3), which becomes less restrictive.

Aspects of Radar Imaging Using Frequency-Stepped Chirp Signals Qun Zhang and Ya-Qiu Jin DOI: 10.1155/ASP/2006/85823 The high-resolution, frequency-stepped chirp signal can be applied to radar systems employing narrow-bandwidth chirp pulses, in order to enhance the range resolution, and to implement SAR/ISAR imaging capabilities. This paper analyzes the effect of moving targets on the synthetic high-resolution range profile obtained using this signal waveform. Some constraints are presented for compensation of the radial motion from shift and amplitude depression of the synthetic range profile. By transmitting two chirp pulses with the same carrier frequency in a pulse-set, a method of ground clutter cancellation is designed with respect to this signal format. Finally, our simulation data demonstrate the effectiveness of the proposed method.

2D Four-Channel Perfect Reconstruction Filter Bank Realized with the 2D Lattice Filter Structure S. Sezen and A. Ertuz¨ un¨ DOI: 10.1155/ASP/2006/42672 A novel orthogonal 2D lattice structure is incorporated into the design of a nonseparable 2D four-channel perfect reconstruction filter bank. The proposed filter bank is obtained by using the polyphase decomposition technique which requires the design of an orthogonal 2D lattice filter. Due to constraint of perfect reconstruction, each stage of this lattice filter bank is simply parameterized by two coefficients. The perfect reconstruction property is satisfied regardless of the actual values of these parameters and of the number of the lattice stages. It is also shown that a separable 2D four-channel perfect reconstruction lattice filter 92 Regular Issue, Vol. 2006 bank can be constructed from the 1D lattice filter and that this is a special case of the proposed 2D lattice filter bank under certain conditions. The perfect reconstruction property of the proposed 2D lattice filter approach is verified by computer simulations.

Image Quality Assessment Using the Joint Spatial/Spatial-Frequency Representation Azeddine Beghdadi and Razvan˘ Iordache DOI: 10.1155/ASP/2006/80537 This paper demonstrates the usefulness of spatial/spatial-frequency representations in image quality assessment by introducing a new image dissimilarity measure based on 2D Wigner-Ville distribution (WVD). The properties of 2D WVD are shortly reviewed, and the important issue of choosing the analytic image is emphasized. The WVD-based measure is shown to be correlated with subjective human evaluation, which is the premise towards an image quality assessor developed on this principle.

Improved Mumford-Shah Functional for Coupled Edge-Preserving Regularization and Image Segmentation Zhang Hongmei and Wan Mingxi DOI: 10.1155/ASP/2006/37129 An improved Mumford-Shah functional for coupled edge-preserving regularization and image segmentation is presented. A nonlinear smooth constraint function is introduced that can induce edge-preserving regularization thus also facilitate the coupled image segmentation. The formulation of the functional is considered from the level set perspective, so that explicit boundary contours and edge-preserving regularization are both addressed naturally. To reduce computational cost, a modified additive operator splitting (AOS) algorithm is developed to address diffusion equations defined on irregular domains and multi- initial scheme is used to speed up the convergence rate. Experimental results by our approach are provided and compared with that of Mumford-Shah functional and other edge- preserving approach, and the results show the effectiveness of the proposed method.

Texture-Gradient-Based Contour Detection Nasser Chaji and Hassan Ghassemian DOI: 10.1155/ASP/2006/21709 In this paper, a new biologically motivated method is proposed to effectively detect per- ceptually homogenous region boundaries. This method integrates the measure of spatial variations in texture with the intensity gradients. In the first stage, texture representation is calculated using the nondecimated complex wavelet transform. In the second stage, gradient images are computed for each of the texture features, as well as for grey scale intensity. These gradients are efficiently estimated using a new proposed algorithm based on a hypothesis model of the human visual system. After that, combining these gradient images, a Regular Issue, Vol. 2006 93 region gradient which highlights the region boundaries is obtained. Nonmaximum suppression and then thresholding with hysteresis is used to detect contour map from the region gradients. Natural and textured images with associated ground truth contour maps are used to evaluate the operation of the proposed method. Experimental results demonstrate that the proposed contour detection method presents more effective performance than conventional approaches.

Adaptive DOA Estimation Using a Database of PARCOR Coefficients Eiji Mochida and Youji Iiguni DOI: 10.1155/ASP/2006/91567 An adaptive direction-of-arrival (DOA) tracking method based upon a linear predictive model is developed. This method estimates the DOA by using a database that stores PAR- COR coefficients as key attributes and the corresponding DOAs as non-key attributes. The k-dimensional digital search tree is used as the data structure to allow efficient multidimensional searching. The nearest neighbour to the current PARCOR coefficient is retrieved from the database, and the corresponding DOA is regarded as the estimate. The processing speed is very fast since the DOA estimation is obtained by the multidimensional searching. Simulations are performed to show the effectiveness of the proposed method. EURASIP Journal on Applied Signal Processing, Volume 2006 © 2006 Hindawi Publishing Corporation

Special Issue on Advances in Multimicrophone Speech Processing

Sharon Gannot School of Engineering, Bar-Ilan University, Ramat-Gan, 52900, Israel

Jacob Benesty INRS-EMT, University of Quebec, 800 de la Gauchetiere Ouest, Montreal, QC, Canada H5A 1K6

Jorg¨ Bitzer Institute of Audiology and Hearing Science, University of Applied Sciences, Oldenburg/Ostfriesland/ Wilhelmshaven Ofener Street 16, 26121 Oldenburg, Germany

Israel Cohen Department of Electrical Engineering, Technion — Israel Institute of Technology, Technion City, Haifa 32000, Israel

Simon Doclo Department of Electrical Engineering (ESAT-SCD), Katholieke Universiteit Leuven, Kasteelpark Arenberg 10, 3001 Leuven, Belgium

Rainer Martin Institute of Communication Acoustics, Ruhr-Universitaet Bochum, 44780 Bochum, Germany

Sven Nordholm Western Australian Telecommunications Research Institute, The University of Western Australia, 35 Stirling Hwy, Crawley, 6009, Australia

Speech quality may signiﬁcantly deteriorate in the presence of interference, especially when the speech signal is also subject to reverberation. Consequently, modern communication systems, such as cellular phones, employ some speech enhancement procedure at the preprocessing stage, prior to further processing (e.g., speech coding). Generally, the performance of single-microphone techniques is limited, since these techniques can utilize only spectral information. Especially for the dereverberation problem, no adequate single-microphone enhancement techniques are presently available. Special Issue on Advances in Multimicrophone Speech Processing 95

Hence, in many applications, such as hands-free mobile telephony, voice-controlled systems, teleconferencing, and hearing instruments, a growing tendency exists to move from single-microphone systems to multimicrophone systems. Although multimicrophone systems come at an increased cost, they exhibit the advantage of incorporating both spatial and spectral information. The use of multimicrophone systems raises many practical considerations such as tracking the desired speech source, and robustness to unknown microphone positions. Further- more, due to the increased computational load, real-time algorithms are more difficult to obtain and hence the efficiency of the algorithms becomes a major issue. The main focus of this special issue is on emerging methods for speech processing using multimicrophone arrays. In the following, the specific contributions are summarized and grouped according to their topic. It is interesting to note that none of the papers deal with the important and difficult problem of dereverberation. Speaker separation In the paper “Speaker separation and tracking system,” Anliker et al. propose a two-stage integrated speaker separation and tracking system. This is an important problem with several potential applications. The authors also propose quantitative criteria to measure the performance of such a system, and present experimental evaluation of their method. In the paper “Speech source separation in convolutive environments using space-time-frequency analysis” Dubnov et al. present a new method for blind separation of convolutive mixtures based on the assumption that the signals in the time-frequency (TF) domain are par- tially disjoint. The method involves detection of single-source TF cells using eigenvalue decomposition of the TF-cells correlation matrices, clustering of the detected cells with expectation-maximization (EM) algorithm based on Gaussian mixture model (GMM), and estimation of smoothed transfer functions between microphones and sources via extended Kalman filtering (EKF). Serviere and Pham propose in their paper “Permutation correction in the frequency-domain in blind separation of speech mixtures” a method for blind separation of convolutive mixtures of speech signals, based on the joint diagonalization of the time-varying spectral matrices of the observation records. This paper proposes a two-step method. First, the frequency continuity of the unmixing filters is used in the initialization of the diagonalization algorithm. Then, the continuity of the time variation of the source energy is exploited on a sliding frequency bandwidth to detect the remaining frequency permutation jumps. In their paper “Geometrical interpretation of the PCA subspace approach for overdetermined blind source separation” Winter et al. discuss approaches for blind source separation where the number of sources can exceed the number of users. Two methods are compared. The first is based on principal component analysis (PCA). The second is based on geometric considerations. Echo cancellation In their paper “Efficient fast stereo acoustic echo cancellation based on pairwise optimal weight realization technique,” Yukawa et al. propose a class of efficient fast acoustic echo cancellation algorithms with linear computational complexity. These algorithms are based on pairwise optimal weight realization power technique. Numerical examples demonstrate that the proposed schemes significantly improve the convergence behavior compared with conventional methods in terms of system mismatch as well as echo return loss enhancement (ERLE). 96 EURASIP Journal on Applied Signal Processing, Vol. 2006

Acoustic source localization Time-delay estimation is a first stage that feeds into subsequent processing blocks for identifying, localizing, and tracking radiating sources. The paper “Time-delay estimation in room acoustic environments: an overview” by Chen et al. presents a systematic overview of the state of the art of time-delay-estimation algorithms ranging from the simple cross- correlation method to the advanced blind channel identification based techniques. In their work “Kalman filters for time-delay of arrival-based source localization,” Klee et al. propose an algorithm for acoustic source localization based on time-delay-of-arrival (TDOA) estimation. In their approach, they use a Kalman filter to directly update the speaker position estimate based on the observed TDOAs. In their contribution, “Microphone array speaker localizers using spatial-temporal information,” Gannot and Dvorkind propose to exploit the speaker’s smooth trajectory for improving the position estimate. Based on TDOA read- ings, three localization schemes, which use the temporal information, are presented. The first is a recursive form of the Gauss method. The other two are extensions of the Kalman filter to the nonlinear problem at hand, namely, the extended Kalman filter and the unscented Kalman filter. In their paper, “Particle filter design using importance sampling for acoustic source localization and tracking in reverberant environments,” Lehmann and Williamson develop a new particle filter for acoustic source localization using importance sampling, and compare its tracking ability with that of a bootstrap algorithm proposed previously in the literature. A real-time implementation of the algorithm also shows that the proposed particle filter can reliably track a person talking in real reverberant rooms. Speech enhancement and speech detection The paper “Dual channel speech enhancement by superdirective beamforming” by Lotter and Vary presents a dual channel input-output speech enhancement system. The proposed algorithm is an adaptation of the well-known superdirective beamformer including postfiltering to the binaural application. In contrast to conventional beamformer processing, the proposed system outputs enhanced stereo signals while preserving the important interaural amplitude and phase differences of the original signal. In their paper “Sector-based detection for hands-free speech enhancement in cars” Lathoud et al. investigate an adaptation control of beamforming interference cancellation techniques for in-car speech acquisition. Two efficient adaptation control methods are proposed that avoid target cancellation. Ex- periments on real in-car data validate both methods, including a case with 100 km/h background road noise. In their paper “Using intermicrophone correlation to detect speech in spatially-separated noise,” Koul and Greenberg provide a theoretical analysis of a system for determining intervals of high and low signal-to-noise ratio when the desired signal and interfering noise arise from distinct spatial regions. The system uses the correlation coefficient between two microphone signals configured in a broadside array as the decision variable in a hypothesis test, and can, for example, be used as an adaptation control method for an adaptive beamformer. Sharon Gannot Jacob Benesty Jorg¨ Bitzer Israel Cohen Simon Doclo Rainer Martin Sven Nordholm Special Issue on Advances in Multimicrophone Speech Processing 97 Volume 2006, No. 12 Contents and Abstracts

Speaker Separation and Tracking System U.Anliker,J.F.Randall,andG.Troster¨ DOI: 10.1155/ASP/2006/29104 Replicating human hearing in electronics under the constraints of using only two microphones (even with more than two speakers) and the user carrying the device at all times (i.e., mobile device weighing less than 100 g) is nontrivial. Our novel contribution in this area is a two-microphone system that incorporates both blind source separation and speaker tracking. This system handles more than two speakers and overlapping speech in a mobile environment. The system also supports the case in which a feedback loop from the speaker tracking step to the blind source separation can improve performance. In order to develop and optimize this system, we have established a novel benchmark that we herewith present. Using the introduced complexity metrics, we present the tradeoﬀs between system performance and computational load. Our results prove that in our case, source separation was signiﬁcantly more dependent on frame duration than on sampling frequency.

Speech Source Separation in Convolutive Open Access Environments Using Space-Time-Frequency Analysis Shlomo Dubnov, Joseph Tabrikian, and Miki Arnon-Targan DOI: 10.1155/ASP/2006/38412 We propose a new method for speech source separation that is based on directionally- disjoint estimation of the transfer functions between microphones and sources at different frequencies and at multiple times. The spatial transfer functions are estimated from eigenvectors of the microphones’ correlation matrix. Smoothing and association of transfer function parameters across different frequencies are performed by simultaneous extended Kalman filtering of the amplitude and phase estimates. This approach allows transfer function estimation even if the number of sources is greater than the number of microphones, and it can operate for both wideband and narrowband sources. The performance of the proposed method was studied via simulations and the results show good performance.

Permutation Correction in the Frequency Domain in Blind Separation of Speech Mixtures Ch. Serviere` and D. T. Pham DOI: 10.1155/ASP/2006/75206 This paper presents a method for blind separation of convolutive mixtures of speech signals, based on the joint diagonalization of the time varying spectral matrices of the observation 98 EURASIP Journal on Applied Signal Processing, Vol. 2006 records. The main and still largely open problem in a frequency domain approach is permutation ambiguity. In an earlier paper of the authors, the continuity of the frequency response of the unmixing ﬁlters is exploited, but it leaves some frequency permutation jumps. This paper therefore proposes a new method based on two assumptions. The frequency continuity of the unmixing ﬁlters is still used in the initialization of the diagonalization algorithm. Then, the paper introduces a new method based on the time-frequency representations of the sources. They are assumed to vary smoothly with frequency. This hypothesis of the continuity of the time variation of the source energy is exploited on a sliding frequency bandwidth. It allows us to detect the remaining frequency permutation jumps. The method is compared with other approaches and results on real world recordings demonstrate superior performances of the proposed algorithm.

Geometrical Interpretation of the PCA Subspace Open Access Approach for Overdetermined Blind Source Separation S. Winter, H. Sawada, and S. Makino DOI: 10.1155/ASP/2006/71632 We discuss approaches for blind source separation where we can use more sensors than sources to obtain a better performance. The discussion focuses mainly on reducing the dimensions of mixed signals before applying independent component analysis. We compare two previously proposed methods. The ﬁrst is based on principal component analysis, where noise reduction is achieved. The second is based on geometric considerations and selects a subset of sensors in accordance with the fact that a low frequency prefers a wide spacing, and a high frequency prefers a narrow spacing. We found that the PCA-based method behaves similarly to the geometry-based method for low frequencies in the way that it emphasizes the outer sensors and yields superior results for high frequencies. These results provide a better understanding of the former method.

Efficient Fast Stereo Acoustic Echo Cancellation Open Access Based on Pairwise Optimal Weight Realization Technique Masahiro Yukawa, Noriaki Murakoshi, and Isao Yamada DOI: 10.1155/ASP/2006/84797 In stereophonic acoustic echo cancellation (SAEC) problem, fast and accurate tracking of echo path is strongly required for stable echo cancellation. In this paper, we propose a class of efficient fast SAEC schemes with linear computational complexity (with respect to filter length). The proposed schemes are based on pairwise optimal weight realization (POWER) technique, thus realizing a “best” strategy (in the sense of pairwise and worst-case optimization) to use multiple-state information obtained by preprocessing. Numerical examples demonstrate that the proposed schemes significantly improve the convergence behavior comparedwithconventionalmethodsintermsofsystemmismatchaswellasechoreturn loss enhancement (ERLE). Special Issue on Advances in Multimicrophone Speech Processing 99

Time Delay Estimation in Room Acoustic Environments: An Overview Jingdong Chen, Jacob Benesty, and Yiteng (Arden) Huang DOI: 10.1155/ASP/2006/26503 Time delay estimation has been a research topic of significant practical importance in many fields (radar, sonar, seismology, geophysics, ultrasonics, hands-free communications, etc.). It is a first stage that feeds into subsequent processing blocks for identifying, localizing, and tracking radiating sources. This area has made remarkable advances in the past few decades, and is continuing to progress, with an aim to create processors that are tolerant to both noise and reverberation. This paper presents a systematic overview of the state- of-the-art of time-delay-estimation algorithms ranging from the simple cross-correlation method to the advanced blind channel identification based techniques. We discuss the pros and cons of each individual algorithm, and outline their inherent relationships. We also provide experimental results to illustrate their performance differences in room acoustic environments where reverberation and noise are commonly encountered.

Kalman Filters for Time Delay of Arrival-Based Source Localization Ulrich Klee, Tobias Gehrig, and John McDonough DOI: 10.1155/ASP/2006/12378 In this work, we propose an algorithm for acoustic source localization based on time delay of arrival (TDOA) estimation. In earlier work by other authors, an initial closed-form approximation was first used to estimate the true position of the speaker followed by a Kalman filtering stage to smooth the time series of estimates. In the proposed algorithm, this closed- form approximation is eliminated by employing a Kalman filter to directly update the speaker’s position estimate based on the observed TDOAs. In particular, the TDOAs com- prise the observation associated with an extended Kalman filter whose state corresponds to the speaker’s position. We tested our algorithm on a data set consisting of seminars held by actual speakers. Our experiments revealed that the proposed algorithm provides source localization accuracy superior to the standard spherical and linear intersection techniques. Moreover, the proposed algorithm, although relying on an iterative optimization scheme, proved efficient enough for real-time operation.

Microphone Array Speaker Localizers Using Spatial-Temporal Information Sharon Gannot and Tsvi Gregory Dvorkind DOI: 10.1155/ASP/2006/59625 A dual-step approach for speaker localization based on a microphone array is addressed in this paper. In the first stage, which is not the main concern of this paper, the time difference between arrivals of the speech signal at each pair of microphones is estimated. These read- ings are combined in the second stage to obtain the source location. In this paper, we focus on the second stage of the localization task. In this contribution, we propose to exploit the 100 EURASIP Journal on Applied Signal Processing, Vol. 2006 speaker’s smooth trajectory for improving the current position estimate. Three localization schemes, which use the temporal information, are presented. The first is a recursive form of the Gauss method. The other two are extensions of the Kalman filter to the nonlinear problem at hand, namely, the extended Kalman filter and the unscented Kalman filter. These methods are compared with other algorithms, which do not make use of the temporal information. An extensive experimental study demonstrates the advantage of using the spatial- temporal methods. To gain some insight on the obtainable performance of the localization algorithm, an approximate analytical evaluation, verified by an experimental study, is con- ducted. This study shows that in common TDOA-based localization scenarios—where the microphone array has small interelement spread relative to the source position—the elevation and azimuth angles can be accurately estimated, whereas the Cartesian coordinates as well as the range are poorly estimated.

Particle Filter Design Using Importance Sampling for Acoustic Source Localisation and Tracking in Reverberant Environments Eric A. Lehmann and Robert C. Williamson DOI: 10.1155/ASP/2006/17021 Sequential Monte Carlo methods have been recently proposed to deal with the problem of acoustic source localisation and tracking using an array of microphones. Previous implementations make use of the basic bootstrap particle filter, whereas a more general approach involves the concept of importance sampling. In this paper, we develop a new particle filter for acoustic source localisation using importance sampling, and compare its tracking ability with that of a bootstrap algorithm proposed previously in the literature. Experimental results obtained with simulated reverberant samples and real audio recordings demonstrate that the new algorithm is more suitable for practical applications due to its reinitialisation capabilities, despite showing a slightly lower average tracking accuracy. A real-time implementation of the algorithm also shows that the proposed particle filter can reliably track a person talking in real reverberant rooms.

Dual-Channel Speech Enhancement by Open Access Superdirective Beamforming Thomas Lotter and Peter Vary DOI: 10.1155/ASP/2006/63297 In this contribution, a dual-channel input-output speech enhancement system is introduced. The proposed algorithm is an adaptation of the well-known superdirective beamformer including postfiltering to the binaural application. In contrast to conventional beamformer processing, the proposed system outputs enhanced stereo signals while preserving the important interaural amplitude and phase differences of the original signal. Instrumental performance evaluations in a real environment with multiple speech sources indicate that the proposed computational efficient spectral weighting system can achieve significant attenuation of speech interferers while maintaining a high speech quality of the target signal. Special Issue on Advances in Multimicrophone Speech Processing 101

Sector-Based Detection for Hands-Free Speech Enhancement in Cars Guillaume Lathoud, Julien Bourgeois, and Jurgen¨ Freudenberger DOI: 10.1155/ASP/2006/20683 Adaptation control of beamforming interference cancellation techniques is investigated for in-car speech acquisition. Two eﬃcient adaptation control methods are proposed that avoid target cancellation. The “implicit” method varies the step-size continuously, based on the ﬁltered output signal. The “explicit” method decides in a binary manner whether to adapt or not, based on a novel estimate of target and interference energies. It estimates the average delay-sum power within a volume of space, for the same cost as the classical delay-sum. Experiments on real in-car data validate both methods, including a case with 100 km/h background road noise.

Using Intermicrophone Correlation to Detect Open Access Speech in Spatially Separated Noise Ashish Koul and Julie E. Greenberg DOI: 10.1155/ASP/2006/93920 This paper describes a system for determining intervals of “high” and “low” signal-to-noise ratios when the desired signal and interfering noise arise from distinct spatial regions. The correlation coefficient between two microphone signals serves as the decision variable in a hypothesis test. The system has three parameters: center frequency and bandwidth of the bandpass filter that prefilters the microphone signals, and threshold for the decision variable. Conditional probability density functions of the intermicrophone correlation coefficient are derived for a simple signal scenario. This theoretical analysis provides insight into optimal selection of system parameters. Results of simulations using white Gaussian noise sources are in close agreement with the theoretical results. Results of more realistic simulations using speech sources follow the same general trends and illustrate the performance achievable in practical situations. The system is suitable for use with two microphones in mild-to-moderate reverberation as a component of noise-reduction algorithms that require detecting intervals when a desired signal is weak or absent. EURASIP Journal on Applied Signal Processing, Volume 2006 © 2006 Hindawi Publishing Corporation

Special Issue on Design Methods for DSP Systems

Markus Rupp Institute of Communications and Radio Frequency Engineering, Vienna University of Technology, Gusshausstrasse 25/389, 1040 Vienna, Austria Bernhard Wess Institute of Communications and Radio Frequency Engineering, Vienna University of Technology, Gusshausstrasse 25/389, 1040 Vienna, Austria Shuvra S. Bhattacharyya Department of Electrical & Computer Engineering, University of Maryland, College Park, MD 20742, USA

Industrial implementations of DSP systems today require extreme complexity. Examples are wireless systems satisfying standards like WLAN or 3GPP, video components, or multimedia players. At the same time, often harsh constraints like low-power requirements burden the designer even more. Conventional methods for ASIC design are not sufficient any more to guarantee a fast conversion from initial concept to final product. In industry, the problem has been addressed by the wording design crisis or design gap. While this design gap exists in a complexity gap, that is, a difference between existing, available, and demanded complexity, there is also a productivity gap, that is, the difference between available complexity and how much we are able to efficiently convert into gate-level representations. This special issue intends to present recent solutions to such gaps addressing algorithmic design methods, algorithms for floating-to-fixed-point conversion, automatic DSP coding strategies, architectural exploration methods, hardware/software partitioning, as well as virtual and rapid prototyping. We received 20 submissions from different fields and areas of expertise from which finally only 12 were accepted for publication. These 12 papers can be categorised into four groups: pure VLSI design methods, prototyping methods, experimental reports on FPGAs, and floating-to-fixed-point conversions. Most activities in design methods are related to the final product. VLSI design methods intend to deal with high complexity in a rather short time. In this special issue, we present five contributions allowing to design complex VLSI designs in substantially lower time pe- riods. In “Macrocell builder: IP-block-based design environment for high-throughput VLSI dedicated digital signal processing systems,” N.-E. Zergainoh et al. present a design tool, called DSP macrocell builder, that generates SystemC register transfer level architectures for VLSI Special Issue on Design Methods for DSP Systems 103 signal processing systems from high-level representations as interconnections of intellectual property (IP) blocks. The development emphasizes extensive parameterization and component reuse to improve productivity and flexibility. Careful generation of control structures is also performed to manage delays and coordinate parallel execution. Effectiveness of the tool is demonstrated on a number of high-throughput signal processing applications. In “Multiple-clock cycle architecture for the VLSI design of a system for time-frequency analysis,” Veselin N. Ivanovic´ et al. present a streamlined architecture for time-frequency signal analysis. The architecture enables real-time analysis of a number of important time- frequency distributions. By providing for multiple-clock-cycle operation and resource sharing across the design in an efficient manner, the architecture achieves these features with relatively low hardware complexity. Results are given based on implementation of the architecture on field-programmable gate arrays, and a thorough comparison is given against a single-cycle implementation architecture. In “3D-SoftChip: a novel architecture for next-generation adaptive computing systems,” C. Kim et al. present an architecture for real-time communication and signal processing through vertical integration of a configurable array processor subsystem and a switch subsystem. The proposed integration is achieved by means of an indium bump interconnection array to provide high interconnection bandwidth at relatively low levels of power dissipa- tion. The paper motivates and develops the design of the proposed system architecture, along with its 2D subsystems and hierarchical interconnection network. Details on hardware/software codesign aspects of the proposed system are also discussed. In “Highly flexible multimode digital signal processing systems using adaptable components and controllers,” V. V. Kumar and J. Lach present a design methodology for signal processing systems. The targeted class of applications involves those that can be decomposed naturally into multiple application modes, where the different modes operate during nonoverlap- ping time intervals. The approach developed in the paper emphasizes supporting flexible application of reconfigurability in multimode signal processing architectures, including reconfigurability in datapath components, controllers, and interconnect, as well as both intra- and inter-mode reconfigurability. The approach is demonstrated through synthesis of multimode applications that are composed of various DSP benchmark subsystems. In “Rapid VLIW processor customization for signal processing applications using combinational hardware functions,” R. R. Hoare et al. present a VLIW processor with multiple application-specific hardware functions for computationally intensive signal processing applications. The hardware functions share the register file with the processor to eliminate overhead by data movement. A design methodology including profiling, compiler transformations for combinational logic synthesis, and code restructuring is proposed to map algorithms written in C onto this architecture. Application speedups are reported for several signal processing benchmarks from the MediaBench suite. A large amount of activities can currently be found in rapid prototyping where it is important to find feasible solutions to a challenging system design in rather short time. Afinalproductmaylookdifferent than the prototype but the prototype is intended to deliver a first hands-on experience of whether a proposal architectural solution is feasible at all. The prototype thus provides the designers with decisions for a final product while still giving them a chance to further explore parts of the design. In “Rapid prototyping for heterogeneous multicomponent systems: an MPEG-4 stream over a UMTS communication link,” M. Raulet et al. present a rapid prototyping method using the SynDEx CAD tool, a half-automated method, to map algorithms that are typically specified 104 EURASIP Journal on Applied Signal Processing, Vol. 2006 in C onto various real-time platforms. Supported platforms are by Sundance and Pentek using a multitude of conventional DSPs and FPGAs. In order to support various platforms, means to describe hardware and software components as well as their communications links are provided in terms of SynDEx kernels. The communication kernel, for example, supports communication between the various functional units via shared RAMs. The efficiency of the proposed method is shown by a rather challenging example: an MPEG-4 stream is provided over a UMTS link. A second contribution in this field entitled “A fully automated environment for verification of virtual prototypes,” P. Belanovic et al. present a computer-aided design tool for automated derivation and verification support of virtual prototypes. The targeted virtual prototypes include definitions of the hardware/software interfaces in the given system, which enables parallel development and improved validation support across hardware and software. The developed tool operates in the context of algorithmic specifications developed through the COSSAP commercial design system for signal processing, and also in the context of target platforms based on the StarCore DSP. Retargetability to other algorithm development environments and target platforms is promising due to the general principles and modular architecture of the developed approach. Many clever ideas to build prototypes based on FPGA were submitted. The three most interesting ones will be presented in this special issue. In “FPGA-based reconfigurable measurement instruments with functionality defined by user,” G.-R. Tsai and M.-C. Lin develop an approach using FPGAs to provide a framework for configurable measurement instruments, where the features and functionality of the instruments can be customized flexibly by the user. A hardware kernel for the configurable instrument approach is presented along with associated implementation considerations. Several examples are developed based on the proposed framework to illustrate the utility of the approach. In “FPGA implementation of a MUD based on cascade filters for a WCDMA system,” Q.- T. Ho et al. present an FPGA-based implementation of a multiuser detector for WCDMA transmission systems. They exploit a serial interference structure in form of a cascade filter. Their design methodology strives for support of maximum number of users while reflecting limited FPGA resources and timing constraints. Elaborate resource utilisation studies for VIRTEX II and VIRTEX II Pro FPGAs from XILINX validate their results. In “A new pipelined systolic array-based architecture for matrix inversion in FPGAs with Kalman filter case study,” A. Bigdeli et al. propose an optimized systolic array-based matrix inversion for implementation in FPGAs. The main advantage of their structure is the small logic resource consumption compared to other systolic arrays in the literature. The hardware complexity is reduced from O(n2)toO(n) for inverting an nxn matrix. The new pipelined systolic array is used for rapid prototyping of a Kalman filter and compared with other implementations. Floating-to-fixed-point conversion is an ongoing topic in system design. Although many concepts have been proposed over the years, there is hardly any tool support in commercial EDA products. In “Floating-to-fixed-point conversion for digital signal processors,” D. Menard et al. follow a different path than researchers have done before. Rather than minimizing signal-to-quantization noise energy, they minimize code execution time on a DSP for a given accuracy constraint. This method includes taking into account the DSP architectural structure. To evaluate the fixed-point accuracy, an analytical approach is used to reduce the optimisation time compared to existing methods. Special Issue on Design Methods for DSP Systems 105

In “Optimum wordlength search using sensitivity information,” K. Han and B. L. Evans propose a fast algorithm for searching for an optimum wordlength by trading oﬀ hardware complexity for arithmetic precision at the system outputs. The optimization is based on the complexity-and-distortion measure that combines hardware complexity information with propagated quantized precision loss. Two case studies demonstrate that the proposed method can ﬁnd optimum wordlengths in less time compared to local search strategies.

Markus Rupp Bernhard Wess Shuvra S. Bhattacharyya 106 EURASIP Journal on Applied Signal Processing, Vol. 2006 Volume 2006, No. 14 Contents and Abstracts

Macrocell Builder: IP-Block-Based Design Environment for High-Throughput VLSI Dedicated Digital Signal Processing Systems Nacer-Eddine Zergainoh, Ludovic Tambour, Pascal Urard, and Ahmed Amine Jerraya DOI: 10.1155/ASP/2006/28636 We propose an efficient IP-block-based design environment for high-throughput VLSI systems. The flow generates SystemC register-transfer-level (RTL) architecture, starting from a Matlab functional model described as a netlist of functional IP. The refinement model inserts automatically control structures to manage delays induced by the use of RTL IPs. It also inserts a control structure to coordinate the execution of parallel clocked IP. The delays may be managed by registers or by counters included in the control structure. The flow has been used successfully in three real-world DSP systems. The experimentations show that the approach can produce efficient RTL architecture and allows to save huge amount of time.

Multiple-Clock-Cycle Architecture for the VLSI Design of a System for Time-Frequency Analysis Veselin N. Ivanovic´ , Radovan Stojanovic´ , and LJubivsaˇ Stankovic´ DOI: 10.1155/ASP/2006/60613 Multiple-clock-cycle implementation (MCI) of a flexible system for time-frequency (TF) signal analysis is presented. Some very important and frequently used time-frequency distributions (TFDs) can be realized by using the proposed architecture: (i) the spectrogram (SPEC) and the pseudo-Wigner distribution (WD), as the oldest and the most important tools used in TF signal analysis; (ii) the S-method (SM) with various convolution window widths, as intensively used reduced interference TFD. This architecture is based on the short-time Fourier transformation (STFT) realization in the first clock cycle. It allows the mentioned TFDs to take different numbers of clock cycles and to share functional units within their execution. These abilities represent the major advantages of multicycle design and they help reduce both hardware complexity and cost. The designed hardware is suitable for a wide range of applications, because it allows sharing in simultaneous realizations of the higher-order TFDs. Also, it can be accommodated for the implementation of the SM with signal-dependent convolution window width. In order to verify the results on real devices, proposed architecture has been implemented with a field programmable gate array (FPGA) chips. Also, at the implementation (silicon) level, it has been compared with the single-cycle implementation (SCI) architecture. Special Issue on Design Methods for DSP Systems 107

3D-SoftChip: A Novel Architecture for Next-Generation Adaptive Computing Systems Chul Kim, Alex Rassau, Stefan Lachowicz, Mike Myung-Ok Lee, and Kamran Eshraghian DOI: 10.1155/ASP/2006/75032 This paper introduces a novel architecture for next-generation adaptive computing systems, which we term 3D-SoftChip. The 3D-SoftChip is a 3-dimensional (3D) vertically integrated adaptive computing system combining state-of-the-art processing and 3D interconnection technology. It comprises the vertical integration of two chips (a configurable array processor and an intelligent configurable switch) through an indium bump interconnection array (IBIA). The configurable array processor (CAP) is an array of heterogeneous processing elements (PEs), while the intelligent configurable switch (ICS) comprises a switch block, 32- bit dedicated RISC processor for control, on-chip program/data memory, data frame buffer, along with a direct memory access (DMA) controller. This paper introduces the novel 3D- SoftChip architecture for real-time communication and multimedia signal processing as a next-generation computing system. The paper further describes the advanced HW/SW codesign and verification methodology, including high-level system modeling of the 3D- SoftChip using SystemC, being used to determine the optimum hardware specification in the early design stage.

Highly Flexible Multimode Digital Signal Processing Systems Using Adaptable Components and Controllers Vinu Vijay Kumar and John Lach DOI: 10.1155/ASP/2006/79595 Multimode systems have emerged as an area- and power-efficient platform for implementing multiple timewise mutually exclusive digital signal processing (DSP) applications in a single hardware space. This paper presents a design methodology for integrating flexible components and controllers into primarily fixed logic multimode DSP systems, thereby increasing their overall efficiency and implementation capabilities. The components are built using a technique called small-scale reconfigurability (SSR) that provides the necessary flexibility for both intermode and intramode reconfigurabilities, without the penal- ties associated with general-purpose reconfigurable logic. Using this methodology, area and power consumption are reduced beyond what is provided by current multimode systems, without sacrificing performance. The results show an average of 7% reduction in datapath component area, 26% reduction in register area, 36% reduction in interconnect MUX cost, and 68% reduction in the number of controller signals, with an average 38% increase in component utilization for a set of benchmark 32-bit DSP applications. 108 EURASIP Journal on Applied Signal Processing, Vol. 2006

Rapid VLIW Processor Customization for Signal Processing Applications Using Combinational Hardware Functions Raymond R. Hoare, Alex K. Jones, Dara Kusic, Joshua Fazekas, John Foster, Shenchih Tung, and Michael McCloud DOI: 10.1155/ASP/2006/46472 This paper presents an architecture that combines VLIW (very long instruction word) processing with the capability to introduce application-specific customized instructions and highly parallel combinational hardware functions for the acceleration of signal processing applications. To support this architecture, a compilation and design automation flow is described for algorithms written in C. The key contributions of this paper are as follows: (1) a 4-way VLIW processor implemented in an FPGA, (2) large speedups through hardware functions, (3) a hardware/software interface with zero overhead, (4) a design methodology for implementing signal processing applications on this architecture, (5) tractable design automation techniques for extracting and synthesizing hardware functions. Several design tradeoffs for the architecture were examined including the number of VLIW functional units and register file size. The architecture was implemented on an Altera Stratix II FPGA. The Stratix II device was selected because it offers a large number of high-speed DSP (digital signal processing) blocks that execute multiply-accumulate operations. Using the MediaBench benchmark suite, we tested our methodology and architecture to accelerate software. Our combined VLIW processor with hardware functions was compared to that of software executing on a RISC processor, specifically the soft core embedded NIOS II processor. For software kernels converted into hardware functions, we show a hardware performance multiplier of up to 230 times that of software with an average 63 times faster. For the entire application in which only a portion of the software is converted to hardware, the performance improvement is as much as 30X times faster than the nonaccelerated application, with a 12X improvement on average.

Rapid Prototyping for Heterogeneous Open Access Multicomponent Systems: An MPEG-4 Stream over a UMTS Communication Link M. Raulet, F. Urban, J.-F. Nezan, C. Moy, O. Deforges, and Y. Sorel DOI: 10.1155/ASP/2006/64369 Future generations of mobile phones, including advanced video and digital communication layers, represent a great challenge in terms of real-time embedded systems. Programmable multicomponent architectures can provide suitable target solutions combining ﬂexibility and computation power. The aim of our work is to develop a fast and automatic prototyping methodology dedicated to signal processing application implementation on parallel heterogeneous architectures, two major features required by future systems. This paper presents the whole methodology based on the SynDEx CAD tool that directly generates a distributed implementation onto various platforms from a high-level application description, taking real-time aspects into account. It illustrates the methodology in the context of real-time distributed executives for multilayer applications based on an MPEG-4 video codec and a UMTS telecommunication link. Special Issue on Design Methods for DSP Systems 109

A Fully Automated Environment for Verification of Open Access Virtual Prototypes P.Belanovic,´ B. Knerr, M. Holzer, and M. Rupp DOI: 10.1155/ASP/2006/32408 The extremely dynamic and competitive nature of the wireless communication systems market demands ever shorter times to market for new products. Virtual prototyping has emerged as one of the most promising techniques to offer the required time savings and resulting increases in design efficiency. A fully automated environment for development of virtual prototypes is presented here, offering maximal efficiency gains, and supporting both design and verification flows, from the algorithmic model to the virtual prototype. The environment employs automated verification pattern refinement to achieve increased reuse in the design process, as well as increased quality by reducing human coding errors.

FPGA-Based Reconfigurable Measurement Instruments with Functionality Defined by User Guo-Ruey Tsai and Min-Chuan Lin DOI: 10.1155/ASP/2006/84340 Using the field-programmable gate array (FPGA) with embedded software-core processor and/or digital signal processor cores, we are able to construct a hardware kernel for measurement instruments, which can fit common electronic measurement and test requirements. We call this approach the software-defined instrumentation (SDI). By properly configuring, we have used the hardware kernel to implement an n-channel arbitrary waveform generator with various add-on functions, a wideband and precise network analyzer, a high-speed signal digitizer, and a real-time sweep spectrum analyzer. With adaptively re- configuring the hardware kernel, SDI concept can easily respond to the rapidly changing user-application-specified needs in measurement and test markets.

FPGA Implementation of an MUD Based on Cascade Filters for a WCDMA System Quoc-Thai Ho, Daniel Massicotte, and Adel-Omar Dahmane DOI: 10.1155/ASP/2006/52919 The VLSI architecture targeted on FPGAs of a multiuser detector based on a cascade of adaptive filters for asynchronous WCDMA systems is presented. The algorithm is briefly described. This paper focuses mainly on real-time implementation. Also, it focuses on a design methodology exploiting the modern technology of programmable logic and overcoming the limitations of commercial tools. The dedicated architecture based on a regular structure of processors and a special structure of memory exploiting FPGA architecture maximizes the processing rate. The proposed architecture was validated using synthesized data in UMTS communication scenarios. The performance goal is to maximize the number 110 EURASIP Journal on Applied Signal Processing, Vol. 2006 of users of different WCDMA data traffics. This dedicated architecture can be used as an intellectual property (IP) core processing an MUD function in the system-on-programmable- chip (SOPC) of UMTS systems. The targeted FPGA components are Virtex-II and Virtex-II Pro families of Xilinx.

A New Pipelined Systolic Array-Based Architecture for Matrix Inversion in FPGAs with Kalman Filter Case Study Abbas Bigdeli, Morteza Biglari-Abhari, Zoran Salcic, and Yat Tin Lai DOI:10.1155/ASP/2006/89186 A new pipelined systolic array-based (PSA) architecture for matrix inversion is proposed. The pipelined systolic array (PSA) architecture is suitable for FPGA implementations as it efficiently uses available resources of an FPGA. It is scalable for different matrix size and as such allows employing parameterisation that makes it suitable for customisation for application-specific needs. This new architecture has an advantage of O(n) processing element complexity, compared to the O(n2) in other systolic array structures, where the size of the input matrix is given by n × n. The use of the PSA architecture for Kalman filter as an implementation example, which requires different structures for different number of states, is illustrated. The resulting precision error is analysed and shown to be negligible.

Floating-to-Fixed-Point Conversion for Open Access Digital Signal Processors

Daniel Menard, Daniel Chillet, and Olivier Sentieys DOI: 10.1155/ASP/2006/96421 Digital signal processing applications are specified with floating-point data types but they are usually implemented in embedded systems with fixed-point arithmetic to minimise cost and power consumption. Thus, methodologies which establish automatically the fixed- point specification are required to reduce the application time-to-market. In this paper, a new methodology for the floating-to-fixed point conversion is proposed for software implementations. The aim of our approach is to determine the fixed-point specification which minimises the code execution time for a given accuracy constraint. Compared to previous methodologies, our approach takes into account the DSP architecture to opti- mise the fixed-point formats and the floating-to-fixed-point conversion process is coupled with the code generation process. The fixed-point data types and the position of the scaling operations are optimised to reduce the code execution time. To evaluate the fixed- point computation accuracy, an analytical approach is used to reduce the optimisation time compared to the existing methods based on simulation. The methodology stages are described and several experiment results are presented to underline the efficiency of this approach. Special Issue on Design Methods for DSP Systems 111

Optimum Wordlength Search Using Open Access Sensitivity Information Kyungtae Han and Brian L. Evans DOI: 10.1155/ASP/2006/92849 Many digital signal processing algorithms are first developed in floating point and later converted into fixed point for digital hardware implementation. During this conversion, more than 50% of the design time may be spent for complex designs, and optimum wordlengths are searched by trading off hardware complexity for arithmetic precision at system outputs. We propose a fast algorithm for searching for an optimum wordlength. This algorithm uses sensitivity information of hardware complexity and system output error with respect to the signal wordlengths, while other approaches use only one of the two sensitivities. This paper presents various optimization methods, and compares sensitivity search methods. Wordlength design case studies for a wireless demodulator show that the proposed method can find an optimum solution in one fourth of the time that the local search method takes. In addition, the optimum wordlength searched by the proposed method yields 30% lower hardware implementation costs than the sequential search method in wireless demodula- tors. Case studies demonstrate the proposed method is robust for searching for the optimum wordlength in a nonconvex space EURASIP Journal on Applied Signal Processing, Volume 2006 © 2006 Hindawi Publishing Corporation

Special Issue on Performance Evaluation in Image Processing

Michael Wirth Department of Computing and Information Science, University of Guelph, Guelph, ON, Canada N1G 2W1 Matteo Fraschini Department of Computing and Information Science, University of Guelph, Guelph, ON, Canada N1G 2W1 Martin Masek Department of Computing and Information Science, University of Guelph, Guelph, ON, Canada N1G 2W1 Michel Bruynooghe Department of Computing and Information Science, University of Guelph, Guelph, ON, Canada N1G 2W1

The scanning and computerized processing of images had its birth in 1956 at the National Bureau of Standards (NBS, now National Institute of Standards and Technology (NIST)) [1]. Image enhancement algorithms were some of the first to be developed [2]. Half a cen- tury later, literally thousands of image processing algorithms have been published. Some of these have been specific to certain applications such as the enhancement of latent finger- prints, whilst others have been more generic in nature, applicable to all, yet master of none. The scope of these algorithms is fairly expansive, ranging from automatically extracting and delineating regions of interest such as in the case of segmentation, to improving the perceived quality of an image, by means of image enhancement. Since the early years of image processing, as in many subfields of software design, there has been a portion of the design process dedicated to algorithm testing. Testing is the process of determining whether or not a particular algorithm has satisfied its specifications relating to criteria such as accuracy and robustness. A major limitation in the design of image processing algorithms lies in the difficulty in demonstrating that algorithms work to an acceptable measure of performance. The purpose of algorithm testing is two-fold. Firstly it provides either a qualitative or a quantitative method of evaluating an algorithm. Secondly, it provides a comparative measure of the algorithm against similar algorithms, assuming similar criteria are used. One of the greatest caveats in designing algorithms incorporating image processing is how to conceive the criteria used to analyze the results. Do we design a criterion which measures sensitivity, robustness, or accuracy? Performance evaluation in the broadest sense refers to a measure Special Issue on Performance Evaluation in Image Processing 113 of some required behavior of an algorithm, whether it is achievable accuracy, robustness, or adaptability. It allows the intrinsic characteristics of an algorithm to be emphasized, as well as the evaluation of its benefits and limitations. More often than not though, such testing has been limited in its scope. Part of this is attributable to the actual lack of formal process used in performance evaluation of image processing algorithms, from the establishment of testing regimes, to the design of metrics. Selection of an appropriate evaluation methodology is dependent on the objective of the task. For example, in the context of image enhancement, requirements are essentially different for screen-based enhancement and enhancement which is embedded within a subalgorithm. Screen-based enhancement is usually assessed in a subjective manner, whereas when an algorithm is encapsulated within a larger system, subjective evaluation is not available, and the algorithm itself must determine the quality of a processed image. Very few approaches to the evaluation of image processing algorithms can be found in the literature, although the concept has been around for decades. A significant difficulty which arises in the evaluation of algorithms is finding suitable metrics which provide an objective measure of performance. A performance metric is a meaningful and computable measure used for quantitatively evaluating the performance of any algorithm. Consider the process of assessing image quality. There is no single quantitative metric which correlates well with image quality as perceived by the human visual system. The process of analyzing failure is intrinsi- cally coupled with the process of performance evaluation. In order to ascertain whether an algorithm fails or not, you have to define the characteristics of success. Failure analysis is the process of determining why an algorithm fails during testing. The knowledge generated is then fed back to the design process in order to engender refinements in the algorithm. This is a difficult process in applications such as image enhancement primarily because there is usually no reference image which can be used as an “ideal” image. The assessment of image quality plays an important role in applications such as consumer electronics. Met- rics could be used to monitor or optimize image quality in digital cameras, benchmark and evaluate image enhancement algorithms. There is no single metric that correlates well with image quality as perceived by the human visual system. Selection of an appropriate evaluation methodology is dependent on the objective of the task. In the context of image enhancement, requirements are essentially different for screen-based enhancement and enhancement that is embedded within an algorithm (as a subalgorithm). The purpose of evaluating an algorithm is to understand its behavior in dealing with different categories of images, and/or help in estimating the best parameters for different applications [3]. Ultimately this may involve some comparison with similar algorithms, in order to rank their performance and provide guidelines for choosing algorithms on the basis of application domain [3]. Assessing the performance of any algorithm in image processing is difficultbecauseperformancedependsonseveralfactors,asconcludedbyHeathetal.[4]: (1) the algorithm itself, (2) the nature of images used to measure the performance of the algorithm, (3) the algorithm parameters used in the evaluation, (4) the method used for evaluating the algorithm. The ease to which an algorithm can be evaluated is directly proportional to the number of parameters it requires. For example, a segmentation algorithm which has no parameters bar, the image to be processed will be easier to evaluate than one which has three parameters which need to be tailored in order to obtain optimal performance. The nature of the 114 EURASIP Journal on Applied Signal Processing, Vol. 2006 image itself also impacts performance. Evaluation with a set “easy” images may produce a higher accuracy than the use of more difficult images containing complex regions. There are no rigid guidelines as to exactly how the process of performance evaluation should be characterized, however there are a number of facets to be considered [5]: testing protocol; testing regime; performance indicators; performance metrics, and image databases. The first of these, testing protocol relates to the successive approach used to perform testing. There are three basic tenets: visual assessment, statistical evaluation,andground truth evaluation. The first stage of performance evaluation involves obtaining a qualitative im- pression of how well an algorithm has performed. For example, when design begins on a new algorithm, a few sample images may be used in a coarse analysis of the usefulness of existing algorithms by means of visual assessment. Visual assessment usually implies comparing the processed image with the original one. Algorithms judged useful at the first stage are investigated in the next stage as to their accuracy using quantitative performance metrics and ground truth data. The “final” stage of evaluation looks at aspects of performance such as robustness, adaptability, and reliability. This process may iterate through a number of cycles. Next is the testing regime which relates to the strategy used for testing the images. There are four basic testing categories. The first of these is exhaustive testing, which is a brute force approach to testing whereby an algorithm is presented with every possible image in a database to test. Such an approach can be overwhelming, and should be limited to the verification component of the design process. Next is boundary value testing, which evaluates a subset of images identified as being representative. The third regime relates to random testing in which images are indiscriminately selected. This relates to a more sta- tistically based process of evaluating an algorithm providing more realistic conditions. For instance, is it realistic to test a mass detection algorithm on a database of mammograms containing only malignant masses and assume it works accurately? What happens when the algorithm is faced with a normal mammogram: will it mark a feature as false-positive? The final testing regime concerns worst-case testing. What happens when an algorithm processes images containing rare or unusual features? Performance evaluation relies on the use of performance indicators. Such indicators convey the qualities of an algorithm. They are often loose characterizations used in the specification of an algorithm, and in themselves are difficult to measure. Typical performance indicators include [5] (1) accuracy: how well the algorithm has performed with respect to some reference; (2) robustness: an algorithm’s capacity for tolerating various conditions; (3) sensitivity: how responsive an algorithm is to small changes in features; (4) adaptability: how the algorithm deals with variability in images; (5) reliability: the degree to which an algorithm, when repeated using the same stable data, yields the same result; (6) efficiency: the practical viability of an algorithm (time and space). Finally there is the notion of the image database: which images should be selected to test an algorithm? This relates to the diversity and complexity of the selected images, how many databases are used in the selection process, and the significance of the images to the segmentation task. The goal of this special issue is to present an overview of current methodologies related to performance evaluation, performance metrics, and failure analysis of image processing algorithms. The first seven papers deal with aspects of performance evaluation in image segmentation, from metrics derived for video object relevance, to skew-tolerance evaluation of Special Issue on Performance Evaluation in Image Processing 115 page segmentation algorithms and evaluation of edge detection. The last five papers deal with diverse areas of performance evaluation. This includes a methodology for designing experiments for performance evaluation and parameter tuning, the verification and validation of fingerprint registration algorithms, and using performance measures in feedback. As both consumer and commercial electronics evolve, spanning applications as diverse as food processing, biometrics, medicine, digital photography, and home theatres, it is increasingly essential to provide software which is both accurate and robust. This requires a standard- ized methodology for testing image processing algorithms, and innovative means to tackle quantifying and automatically resolving issues relating to algorithm functioning. The assessment and characterization of image processing algorithms is an emerging field, which has been growing for the past three decades. We hope that this special issue will direct more energy to the problem of performance evaluation, and revitalize interest in this burgeoning field.

Michael Wirth Matteo Fraschini Martin Masek Michel Bruynooghe 116 EURASIP Journal on Applied Signal Processing, Vol. 2006 Volume 2006, No. 15 Contents and Abstracts

Video Object Relevance Metrics for Overall Segmentation Quality Evaluation Paulo Correia and Fernando Pereira DOI: 10.1155/ASP/2006/82195 Video object segmentation is a task that humans perform efficiently and effectively, but which is difficult for a computer to perform. Since video segmentation plays an important role for many emerging applications, as those enabled by the MPEG-4 and MPEG-7 standards, the ability to assess the segmentation quality in view of the application targets is a relevant task for which a standard, or even a consensual, solution is not available. This paper considers the evaluation of overall segmentation partitions quality, highlighting one of its major components: the contextual relevance of the segmented objects. Video object relevance metrics are presented taking into account the behaviour of the human visual system and the visual attention mechanisms. In particular, contextual relevance evaluation takes into account the context where an object is found, exploiting, for instance, the contrast to neighbours or the position in the image. Most of the relevance metrics proposed in this paper can also be used in contexts other than segmentation quality evaluation, such as object-based rate control algorithms, description creation, or image and video quality evaluation.

Evaluating Edge Detection through Boundary Detection Song Wang, Feng Ge, and Tiecheng Liu DOI: 10.1155/ASP/2006/76278 Edge detection has been widely used in computer vision and image processing. However, the performance evaluation of the edge-detection results is still a challenging problem. A major dilemma in edge-detection evaluation is the difficulty to balance the objectivity and generality: a general-purpose edge-detection evaluation independent of specific applications is usually not well defined, while an evaluation on a specific application has weak generality. Aiming at addressing this dilemma, this paper presents new evaluation methodology and a framework in which edge detection is evaluated through boundary detection, that is, the likelihood of retrieving the full object boundaries from this edge-detection output. Such a likelihood, we believe, reflects the performance of edge detection in many applications since boundary detection is the direct and natural goal of edge detection. In this framework, we use the newly developed ratio-contour algorithm to group the detected edges into closed boundaries. We also collect a large data set (1030) of real images with unambiguous ground-truth boundaries for evaluation. Five edge detectors (Sobel, LoG, Canny, Roth- well, and Edison) are evaluated in thispaper and we find that the current edge-detection Special Issue on Performance Evaluation in Image Processing 117 performance still has scope for improvement by choosing appropriate detectors and detector parameters.

Fast and Accurate Ground Truth Generation for Skew-Tolerance Evaluation of Page Segmentation Algorithms Oleg Okun and Matti Pietikainen¨ DOI: 10.1155/ASP/2006/12093 Many image segmentation algorithms are known, but often there is an inherent obstacle in the unbiased evaluation of segmentation quality: the absence or lack of a common objective representation for segmentation results. Such a representation, known as the ground truth, is a description of what one should obtain as the result of ideal segmentation, independently of the segmentation algorithm used. The creation of ground truth is a laborious process and therefore any degree of automation is always welcome. Document image analysis is one of the areas where ground truths are employed. In this paper, we describe an automated tool called GROTTO intended to generate ground truths for skewed document images, which can be used for the performance evaluation of page segmentation algorithms. Some of these algorithms are claimed to be insensitive to skew (tilt of text lines). However, this fact is usually supported only by a visual comparison of what one obtains and what one should obtain since ground truths are mostly available for upright images, that is, those without skew. As a result, the evaluation is both subjective; that is, prone to errors, and tedious. Our tool allows users to quickly and easily produce many suﬃciently accurate ground truths that can be employed in practice and therefore it facilitates automatic performance evaluation. The main idea is to utilize the ground truths available for upright images and the concept of the representative square [9] in order to produce the ground truths for skewed images. The usefulness of our tool is demonstrated through a number of experiments with real- document images of complex layout.

A Method for Single-Stimulus Quality Assessment of Segmented Video R. Piroddi and T. Vlachos DOI: 10.1155/ASP/2006/39482 We present a uniﬁed method for single-stimulus quality assessment of segmented video. This method takes into consideration colour and motion features of a moving sequence and monitors their changes across segment boundaries. Features are estimated using a local neighbourhood which preserves the topological integrity of segment boundaries. Further- more the proposed method addresses the problem of unreliable and/or unavailable feature estimates by applying normalized diﬀerential convolution (NDC). Our experimental results suggest that the proposed method outperforms competing methods in terms of sensitivity as well as noise immunity for a variety of standard test sequences. 118 EURASIP Journal on Applied Signal Processing, Vol. 2006

A Method for Assessment of Segmentation Success Open Access Considering Uncertainty in the Edge Positions Ruben´ Usamentiaga, Daniel F. Garc´ıa, Carlos Lopez,´ and Diego Gonzalez´ DOI: 10.1155/ASP/2006/21746 A method for segmentation assessment is proposed. The technique is based on a comparison of the segmentation produced by an algorithm with an ideal segmentation. The procedure to obtain the ideal segmentation is described in detail. Uncertainty regarding the edge positions is accounted for in the discrepancy calculation of each edge using fuzzy rea- soning. The uncertainty measurement consists of a generalization, using fuzzy membership functions, of the similarity metrics used by well-known assessment methods. Several alter- natives for the fuzzy membership functions, based on statistical properties of the possible positions of each edge, are deﬁned. The proposed uncertainty measurement can be easily applied to other well-known methods. Finally, the segmentation assessment method is used to determine the best segmentation algorithm for thermographic images, and also to tune the optimum parameters of each algorithm.

Unsupervised Performance Evaluation of Image Segmentation Sebastien Chabrier, Bruno Emile, Christophe Rosenberger, and Helene Laurent DOI: 10.1155/ASP/2006/96306 We present in this paper a study of unsupervised evaluation criteria that enable the quan- tification of the quality of an image segmentation result. These evaluation criteria compute some statistics for each region or class in a segmentation result. Such an evaluation criterion can be useful for different applications: the comparison of segmentation results, the automatic choice of the best fitted parameters of a segmentation method for a given image, or the definition of new segmentation methods by optimization. We first present the state of art of unsupervised evaluation, and then, we compare six unsupervised evaluation criteria. For this comparative study, we use a database composed of 8400 synthetic gray- level images segmented in four different ways. Vinet’s measure (correct classification rate) is used as an objective criterion to compare the behavior of the different criteria. Finally, we present the experimental results on the segmentation evaluation of a few gray-level natural images.

Design of Experiments for Performance Evaluation and Parameter Tuning of a Road Image Processing Chain Yves Lucas, Antonio Domingues, Driss Driouchi, and Sylvie Treuillet DOI: 10.1155/ASP/2006/48012 Tuning a complete image processing chain (IPC) is not a straightforward task. The first problem to overcome is the evaluation of the whole process. Until now researchers have focused on the evaluation of single algorithms based on a small number of test images and Special Issue on Performance Evaluation in Image Processing 119 ad hoc tuning independent of input data. In this paper, we explain how the design of experiments applied on a large image database enables statistical modeling for IPC significant parameter identification. The second problem is then considered: how can we find the relevant tuning and continuously adapt image processing to input data? After the tuning of the IPC on a typical subset of the image database using numerical optimization, we develop an adaptive IPC based on a neural network working on input image descriptors. By testing this approach on an IPC dedicated-to-road obstacle detection, we demonstrate that this experimental methodology and software architecture can ensure continuous efficiency. The reason is simple: the IPC is globally optimized, from a large number of real images and with adaptive processing of input data.

Verification and Validation of a Fingerprint Image Registration Software Dejan Desovski, Vijai Gandikota, Yan Liu, Yue Jiang, and Bojan Cukic DOI: 10.1155/ASP/2006/15940 The need for reliable identification and authentication is driving the increased use of biometric devices and systems. Verification and validation techniques applicable to these systems are rather immature and ad hoc, yet the consequences of the wide deployment of biometric systems could be significant. In this paper we discuss an approach towards validation and reliability estimation of a fingerprint registration software. Our validation approach includes the following three steps: (a) the validation of the source code with respect to the system requirements specification; (b) the validation of the optimization algorithm, which is in the core of the registration system; and (c) the automation of testing. Since the optimization algorithm is heuristic in nature, mathematical analysis and test results are used to estimate the reliability and perform failure analysis of the image registration module.

Performance Measure as Feedback Variable in Image Processing Danijela RisticandAxelGr´ aser¨ DOI: 10.1155/ASP/2006/27848 This paper extends the view of image processing performance measure presenting the use of this measure as an actual value in a feedback structure. The idea behind is that the control loop, which is built in that way, drives the actual feedback value to a given set point. Since the performance measure depends explicitly on the application, the inclusion of feedback structures and choice of appropriate feedback variables are presented on example of optical character recognition in industrial application. Metrics for quantiﬁcation of performance at diﬀerent image processing levels are discussed. The issues that those metrics should address from both image processing and control point of view are considered. The performance measures of individual processing algorithms that form a character recognition system are determined with respect to the overall system performance. 120 EURASIP Journal on Applied Signal Processing, Vol. 2006

A Comparison of Set Redundancy Compression Techniques Samy Ait-Aoudia and Abdelhalim Gabis DOI: 10.1155/ASP/2006/92734 Medical imaging applications produce large sets of similar images. Thus a compression technique is necessary to reduce space storage. Lossless compression methods are necessary in such critical applications. Set redundancy compression (SRC) methods exploit the interimage redundancy and achieve better results than individual image compression techniques when applied to sets of similar images. In this paper, we make a comparative study of SRC methods on sample datasets using various archivers. We also propose a new SRC method and compare it to existing SRC techniques.

Accuracy Evaluation for Region Centroid-Based Registration of Fluorescent CLSM Imagery Sang-Chul Lee, Peter Bajcsy, Amy Lin, and Robert Folberg DOI: 10.1155/ASP/2006/82480 We present an accuracy evaluation of a semiautomatic registration technique for 3D volume reconstruction from fluorescent confocal laser scanning microscope (CLSM) imagery. The presented semiautomatic method is designed based on our observations that (a) an accurate point selection is much harder than an accurate region (segment) selection for a human, (b) a centroid selection of any region is less accurate by a human than by a computer, and (c) registration based on structural shape of a region rather than based on intensity-defined point is more robust to noise and to morphological deformation of features across stacks. We applied the method to image mosaicking and image alignment registration steps and evaluated its performance with 20 human subjects on CLSM images with stained blood vessels. Our experimental evaluation showed significant benefits of automation for 3D volume reconstruction in terms of achieved accuracy, consistency of results, and performance time. In addition, the results indicate that the differences between registration accuracy obtained by experts and by novices disappear with the proposed semiautomatic registration technique while the absolute registration accuracy increases. EURASIP Journal on Applied Signal Processing, Volume 2006 © 2006 Hindawi Publishing Corporation

Special Issue on Advanced Signal Processing Techniques for Bioinformatics

Xue-Wen Chen Department of Electrical Engineering and Computer Science, University of Kansas, Lawrence, KS 66045, USA Sun Kim School of Informatics, Indiana University, Bloomington, IN 47408, USA

Vladimir Pavlovic´ Department of Computer Science, Rutgers University, Piscataway, NJ 08854, USA

David P.Casasent Department of Electrical and Computer Engineering, Carnegie Mellon University, Pittsburgh, PA 15213, USA

The success of bioinformatics in recent years has been driven, in part, by advanced signal processing techniques: estimation theory, classification, pattern recognition, information theory, networks, imaging, image processing, coding theory, and speech recognition. For example, Fourier analysis methods are used to elucidate the relationship between sequence structure and function; wavelet analysis methods have been applied in sequence comparison and classification; and various image processing methods have been developed to improve microarray image quality. The development of advanced high-throughput technologies such as genome sequenc- ing and whole genome expression analysis creates new opportunities and poses new challenges for the signal processing community. Analysis of data for life-science problems provides an interesting application domain for standard signal processing methods such as time series detection and prediction, casual modeling, and structure inference. At the same time, this increasingly important life-science domain draws the need for novel and computationally efficient analysis approaches. The goal of this special issue is to present the applications of cutting-edge signal processing methods to bioinformatics. Eleven papers accepted for this special issue cover a broad range of topics, from RNA sequence analysis and gene expression analysis to protein structure predictions. The authors developed a variety of signal processing algorithms, such as artificial neural networks, decision trees, biclustering, matrix factorization, and FPGA reconfiguration methods, to tackle these central bioinformatics problems. 122 EURASIP Journal on Applied Signal Processing, Vol. 2006

The issue starts with two papers on gene sequence analyses. Churkin and Barash developed a pattern recognition-based utility to perform mutational analysis and detect vulnerable spots within an RNA sequence that affect structures; Babu et al. presented image processing/computer vision methods for automatic recovery and visualization of the 3D chromosome structure from a sequence of 2D tomographic reconstruction images taken through the nucleus of a cell. The advent of microarray techniques that allow for measuring the expression levels of tens of thousands of genes simultaneously has drawn increased interest in signal processing community, covering a range of problems from microarray image processing and biomarker detection to genetic regulatory network reconstruction. Three papers in this special issue address microarray applications: Bajcsy provided an excellent overview on DNA microarray grid alignment and foreground separation approaches; Jin et al. proposed two automated methods for microarray image analysis; Tchagang and Tewfik described a novel biclustering algorithm for microarray data. As gene products, proteins play an essential role in nearly all cellular functions. The remaining papers deal with issues in proteomics. Two papers focused on the predictio n of protein-protein interactions (PPIs) based on domain information: Zhang et al. modeled the problem of interaction inference as a constraint satisfiability problem and solved it using linear programming; Chen and Liu developed neural network and decision tree-based approaches for predicting PPIs, and demonstrated that with decision trees, multiple domain interactions could be identified. The following three papers moved to protein structure related topics: Okun and Priisalu applied fast nonnegative matrix factorization methods to protein fold recognition; VanCourt et al. presented an FPGA reconfiguration method for alternative force laws with applications to molecular docking; Armano et al. developed a pattern recognition system for protein secondary structure prediction. Finally, Kolibal and Howard developed a stochastic Bernstein approximation method for obtaining the baseline shift removal of matrix-assisted laser desorption ionization time-off-light mass spectrometry. The guest editors would like to thank all the authors for their high quality work contributed to this special issue and all the reviewers for their hard work and expert comments in evaluating the manuscripts.

Xue-wen Chen Sun Kim Vladimir Pavlovi´c David P. Casasent Special Issue on Advanced Signal Processing Techniques for Bioinformatics 123 Volume 2006, No. 7 Contents and Abstracts

Structural Analysis of Single-Point Mutations Given an RNA Sequence: A Case Study with RNAMute Alexander Churkin and Danny Barash DOI: 10.1155/ASP/2006/56246 We introduce here for the ﬁrst time the RNAMute package, a pattern-recognition-based utility to perform mutational analysis and detect vulnerable spots within an RNA sequence that aﬀect structure. Mutations in these spots may lead to a structural change that directly relates to a change in functionality. Previously, the concept was tried on RNA genetic control elements called “riboswitches” and other known RNA switches, without an organized utility that analyzes all single-point mutations and can be further expanded. The RNAMute package allows a comprehensive categorization, given an RNA sequence that has functional relevance, by exploring the patterns of all single-point mutants. For illustration, we apply the RNAMute package on an RNA transcript for which individual point mutations were shown experimentally to inactivate spectinomycin resistance in Escherichia coli. Functional analysis of mutations on this case study was performed experimentally by creating a library of point mutations using PCR and screening to locate those mutations. With the availability of RNAMute, preanalysis can be performed computationally before conducting an experiment.

Recovery and Visualization of 3D Structure of Chromosomes from Tomographic Reconstruction Images Sabarish Babu, Pao-Chuan Liao, Min C. Shin, and Leonid V. Tsap DOI: 10.1155/ASP/2006/45684 The objectives of this work include automatic recovery and visualization of a 3D chromosome structure from a sequence of 2D tomographic reconstruction images taken through the nucleus of a cell. Structure is very important for biologists as it affects chromosome functions, behavior of the cell, and its state. Analysis of chromosome structure is significant in the detection of diseases, identification of chromosomal abnormalities, study of DNA structural conformation, in-depth study of chromosomal surface morphology, observation of in vivo behavior of the chromosomes over time, and in monitoring environmental gene mutations. The methodology incorporates thresholding based on a histogram analysis with a polyline splitting algorithm, contour extraction via active contours, and detection of the 3D chromosome structure by establishing corresponding regions throughout the slices. Vi- sualization using point cloud meshing generates a 3D surface. The 3D triangular mesh of the chromosomes provides surface detail and allows a user to interactively analyze chromosomes using visualization software. 124 EURASIP Journal on Applied Signal Processing, Vol. 2006

An Overview of DNA Microarray Grid Alignment and Foreground Separation Approaches Peter Bajcsy DOI: 10.1155/ASP/2006/80163 This paper overviews DNA microarray grid alignment and foreground separation approaches. Microarray grid alignment and foreground separation are the basic processing steps of DNA microarray images that affect the quality of gene expression information, and hence impact our confidence in any data-derived biological conclusions. Thus, understanding microarray data processing steps becomes critical for performing optimal microarray data analysis. In the past, the grid alignment and foreground separation steps have not been covered extensively in the survey literature. We present several classifications of existing algorithms, and describe the fundamental principles of these algorithms. Challenges related to automation and reliability of processed image data are outlined at the end of this overview paper.

Extended δ-Regular Sequence for Automated Open Access Analysis of Microarray Images Hee-Jeong Jin, Bong-Kyung Chun, and Hwan-Gue Cho DOI: 10.1155/ASP/2006/13623 Microarray study enables us to obtain hundreds of thousands of expressions of genes or genotypes at once, and it is an indispensable technology for genome research. The ﬁrst step is the analysis of scanned microarray images. This is the most important procedure for obtaining biologically reliable data. Currently most microarray image processing systems require burdensome manual block/spot indexing work. Since the amount of experimental data is increasing very quickly, automated microarray image analysis software becomes important. In this paper, we propose two automated methods for analyzing microarray images. First, we propose the extended δ-regular sequence to index blocks and spots, which enables a novel automatic gridding procedure. Second, we provide a methodology, hierarchical metagrid alignment, to allow reliable and eﬃcient batch processing for a set of microarray images. Experimental results show that the proposed methods are more reliable and convenient than the commercial tools.

DNA Microarray Data Analysis: A Novel Biclustering Algorithm Approach Alain B. Tchagang and Ahmed H. Tewfik DOI: 10.1155/ASP/2006/59809 Biclustering algorithms refer to a distinct class of clustering algorithms that perform simultaneous row-column clustering. Biclustering problems arise in DNA microarray data analysis, collaborative filtering, market research, information retrieval, text mining, elec- toral trends, exchange analysis, and so forth. When dealing with DNA microarray experimental data for example, the goal of biclustering algorithms is to find submatrices, that is, Special Issue on Advanced Signal Processing Techniques for Bioinformatics 125 subgroups of genes and subgroups of conditions, where the genes exhibit highly correlated activities for every condition. In this study, we develop novel biclustering algorithms using basic linear algebra and arithmetic tools. The proposed biclustering algorithms can be used to search for all biclusters with constant values, biclusters with constant values on rows, biclusters with constant values on columns, and biclusters with coherent values from a set of data in a timely manner and without solving any optimization problem. We also show how one of the proposed biclustering algorithms can be adapted to identify biclusters with coherent evolution. The algorithms developed in this study discover all valid biclusters of each type, while almost all previous biclustering approaches will miss some.

Towards Inferring Protein Interactions: Challenges and Solutions Ya Zhang, Hongyuan Zha, Chao-Hsien Chu, and Xiang Ji DOI: 10.1155/ASP/2006/37349 Discovering interacting proteins has been an essential part of functional genomics. How- ever, existing experimental techniques only uncover a small portion of any interactome. Furthermore, these data often have a very high false rate. By conceptualizing the interactions at domain level, we provide a more abstract representation of interactome, which also facilitates the discovery of unobserved protein-protein interactions. Although several domain-based approaches have been proposed to predict protein-protein interactions, they usually assume that domain interactions are independent on each other for the convenience of computational modeling. A new framework to predict protein interactions is proposed in this paper, where no assumption is made about domain interactions. Protein interactions may be the result of multiple domain interactions which are dependent on each other. A conjunctive norm form representation is used to capture the relationships between protein interactions and domain interactions. The problem of interaction inference is then modeled as a constraint satisﬁability problem and solved via linear programing. Experimental results on a combined yeast data set have demonstrated the robustness and the accuracy of the proposed algorithm. Moreover, we also map some predicted interacting domains to three-dimensional structures of protein complexes to show the validity of our predictions.

Domain-Based Predictive Models for Protein-Protein Interaction Prediction Xue-Wen Chen and Mei Liu DOI: 10.1155/ASP/2006/32767 Protein interactions are of biological interest because they orchestrate a number of cellular processes such as metabolic pathways and immunological recognition. Recently, methods for predicting protein interactions using domain information are proposed and preliminary results have demonstrated their feasibility. In this paper, we develop two domain-based statistical models (neural networks and decision trees) for protein interaction predictions. Un- like most of the existing methods which consider only domain pairs (one domain from one protein) and assume that domain-domain interactions are independent of each other, the proposed methods are capable of exploring all possible interactions between domains and make predictions based on all the domains. Compared to maximum-likelihood estimation 126 EURASIP Journal on Applied Signal Processing, Vol. 2006 methods, our experimental results show that the proposed schemes can predict protein- protein interactions with higher speciﬁcity and sensitivity, while requiring less computation time. Furthermore, the decision tree-based model can be used to infer the interactions not only between two domains, but among multiple domains as well.

Fast Nonnegative Matrix Factorization and Its Application for Protein Fold Recognition Oleg Okun and Helen Priisalu DOI: 10.1155/ASP/2006/71817 Linear and unsupervised dimensionality reduction via matrix factorization with nonnega- tivity constraints is studied. Because of these constraints, it stands apart from other linear dimensionality reduction methods. Here we explore nonnegative matrix factorization in combination with three nearest-neighbor classifiers for protein fold recognition. Since typically matrix factorization is iteratively done, convergence, can be slow. To speed up convergence, we perform feature scaling (normalization) prior to the beginning of iterations. This results in a significantly (more than 11 times) faster algorithm. Justification of why it happens is provided. Another modification of the standard nonnegative matrix factorization algorithm is concerned with combining two known techniques for mapping unseen data. This operation is typically necessary before classifying the data in low-dimensional space. Combining two mapping techniques can yield better accuracy than using either technique alone. The gains, however, depend on the state of the random number generator used for initialization of iterations, a classifier, and its parameters. In particular, when employing the best out of three classifiers and reducing the original dimensionality by around 30%, these gains can reach more than 4%, compared to the classification in the original, high- dimensional space.

Rigid Molecule Docking: FPGA Reconfiguration for Alternative Force Laws Petros T. Boufounos and Alan V. Oppenheim DOI: 10.1155/ASP/2006/97950 Molecular docking is one of the primary computational methods used by pharmaceutical companies to try to reduce the cost of drug discovery. A common docking technique, used for low-resolution screening or as an intermediate step, performs a three-dimensional correlation between two molecules to test for favorable interactions between them. We extend our previous work on FPGA-based docking accelerators, using reconfigurability for customization of the physical laws and geometric models that describe molecule interaction. Our approach, based on direct summation, allows straightforward combination of multiple forces and enables nonlinear force models; the latter, in particular, are incompatible with the transform-based techniques typically used. Our approach has the further advantage of supporting spatially oriented values in molecule models, as well as the detection of multiple positions representing favorable interactions. We report performance measurements on several different models of chemical behavior and show speedups of from 130 to 1100 over aPC. Special Issue on Advanced Signal Processing Techniques for Bioinformatics 127

MASSP3: A System for Predicting Protein Secondary Structure Giuliano Armano, Alessandro Orro, and Eloisa Vargiu DOI: 10.1155/ASP/2006/42961 A system that resorts to multiple experts for dealing with the problem of predicting secondary structures is described, whose performances are comparable to those obtained by other state-of-the-art predictors. The system performs an overall processing based on two main steps: ﬁrst, a “sequence-to-structure” prediction is performed, by resorting to a population of hybrid genetic-neural experts, and then a “structure-to-structure” prediction is performed, by resorting to a feedforward artiﬁcial neural networks. To investigate the performance of the proposed approach, the system has been tested on the RS126 set of proteins. Experimental results (about 76% of accuracy) point to the validity of the approach.

MALDI-TOF Baseline Drift Removal Using Stochastic Bernstein Approximation Joseph Kolibal and Daniel Howard DOI: 10.1155/ASP/2006/63582 Stochastic Bernstein (SB) approximation can tackle the problem of baseline drift correction of instrumentation data. This is demonstrated for spectral data: matrix-assisted laser desorption/ionization time-of-flight mass spectrometry (MALDI-TOF) data. Two SB schemes for removing the baseline drift are presented: iterative and direct. Following an explanation of the origin of the MALDI-TOF baseline drift that sheds light on the inherent difficulty of its removal by chemical means, SB baseline drift removal is illustrated for both proteomics and genomics MALDI-TOF data sets. SB is an elegant signal processing method to obtain a numerically straightforward baseline shift removal method as it includes a free parameter σ(x) that can be optimized for different baseline drift removal applications. Therefore, research that determines putative biomarkers from the spectral data might benefit from a sensitivity analysis to the underlying spectral measurement that is made possible by varying the SB free parameter. This can be manually tuned (for constant σ) or tuned with evolutionary computation (for σ(x)). EURASIP JOURNAL ON APPLIED SIGNAL PROCESSING

Special Issue on Super-resolution Enhancement of Digital Video

CALL FOR PAPERS When designing a system for image acquisition, there is generally a desire for high spatial resolution and a wide field-of-view. To achieve this, a camera system must typically employ small f-number optics. This produces an image with very high spatial-frequency bandwidth at the focal plane. To avoid aliasing caused by undersampling, the corresponding focal plane array (FPA) must be sufficiently dense. However, cost and fabrication com- plexities may make this impractical. More fundamentally, smaller detectors capture fewer photons, which can lead to potentially severe noise levels in the acquired imagery. Consid- ering these factors, one may choose to accept a certain level of undersampling or to sacrifice some optical resolution and/or field-of-view. In image super-resolution (SR), postprocessing is used to obtain images with resolutions that go beyond the conventional limits of the uncompensated imaging system. In some systems, the primary limiting factor is the optical resolution of the image in the focal plane as defined by the cut-off frequency of the optics. We use the term “optical SR” to refer to SR methods that aim to create an image with valid spatial-frequency content that goes beyond the cut-off frequency of the optics. Such techniques typically must rely on extensive a priori information. In other image acquisition systems, the limiting factor may be the density of the FPA, subsequent postprocessing requirements, or transmission bitrate constraints that require data compression. We refer to the process of overcoming the limitations of the FPA in order to obtain the full resolution afforded by the selected optics as “detector SR.” Note that some methods may seek to perform both optical and detector SR. Detector SR algorithms generally process a set of low-resolution aliased frames from a video sequence to produce a high-resolution frame. When subpixel relative motion is present between the objects in the scene and the detector array, a unique set of scene samples are acquired for each frame. This provides the mechanism for effectively increasing the spatial sampling rate of the imaging system without reducing the physical size of the detectors. With increasing interest in surveillance and the proliferation of digital imaging and video, SR has become a rapidly growing field. Recent advances in SR include innovative algorithms, generalized methods, real-time implementations, and novel applications. The purpose of this special issue is to present leading research and development in the area of super-resolution for digital video. Topics of interest for this special issue include but are not limited to: • Detector and optical SR algorithms for video • Real-time or near-real-time SR implementations • Innovative color SR processing • Novel SR applications such as improved object detection, recognition, and tracking • Super-resolution from compressed video • Subpixel image registration and optical flow

Authors should follow the EURASIP JASP manuscript format described at the journal site http://www.hindawi.com/journals/asp/. Prospective authors should submit an electronic copy of their complete manuscripts through the EURASIP JASP manuscript tracking system at http://www.hindawi.com/mts/, according to the following timetable:

Manuscript Due September 1, 2006 Acceptance Notiﬁcation February 1, 2007 Final Manuscript Due April 15, 2007 Publication Date 3rd Quarter, 2007

GUEST EDITORS: Russell C. Hardie, Department of Electrical and Computer Engineering, University of Dayton, 300 College Park, Dayton, OH 45469-0026, USA; [email protected] Richard R. Schultz, Department of Electrical Engineering, University of North Dakota, Upson II Room 160, P.O. Box 7165, Grand Forks, ND 58202-7165, USA; [email protected] Kenneth E. Barner, Department of Electrical and Computer Engineering, University of Delaware, 140 Evans Hall, Newark, DE 19716-3130, USA; [email protected]

http://www.hindawi.com EURASIP JOURNAL ON APPLIED SIGNAL PROCESSING

Special Issue on Transforming Signal Processing Applications into Parallel Implementations

CALL FOR PAPERS There is an increasing need to develop efficient “system-level” models, methods, and tools to support designers to quickly transform signal processing application specification to heterogeneous hardware and software architectures such as arrays of DSPs, heterogeneous platforms involving microprocessors, DSPs and FPGAs, and other evolving multiprocessor SoC architectures. Typically, the design process involves aspects of application and architecture modeling as well as transformations to translate the application models to architecture models for subsequent performance analysis and design space exploration. Accurate predictions are indispensable because next generation signal processing applications, for example, audio, video, and array signal processing impose high throughput, real-time and energy constraints that can no longer be served by a single DSP. There are a number of key issues in transforming application models into parallel implementations that are not addressed in current approaches. These are engineering the application specification, transforming application specification, or representation of the architecture specification as well as communication models such as data transfer and synchronization primitives in both models. The purpose of this call for papers is to address approaches that include application transformations in the performance, analysis, and design space exploration efforts when taking signal processing applications to concurrent and parallel implementations. The Guest Editors are soliciting contributions in joint application and architecture space exploration that outperform the current architecture-only design space exploration methods and tools. Topics of interest for this special issue include but are not limited to: • modeling applications in terms of (abstract) control-dataflow graph, dataflow graph, and process network models of computation (MoC) • transforming application models or algorithmic engineering • transforming application MoCs to architecture MoCs • joint application and architecture space exploration • joint application and architecture performance analysis • extending the concept of algorithmic engineering to architecture engineering • design cases and applications mapped on multiprocessor, homogeneous, or heterogeneous SOCs, showing joint optimization of application and architecture

Authors should follow the EURASIP JASP manuscript format described at the journal site http://www.hindawi.com/journals/asp/. Prospective authors should submit an electronic copy of their complete manuscript through the EURASIP JASP manuscript tracking system at http://www.hindawi.com/mts, according to the following timetable:

Manuscript Due September 1, 2006 Acceptance Notiﬁcation January 1, 2007 Final Manuscript Due April 1, 2007 Publication Date 3rd Quarter 2007

GUEST EDITORS: F. Deprettre, Leiden Embedded Research Center, Leiden University, Niels Bohrweg 1, 2333 CA Leiden, The Netherlands; [email protected] Roger Woods, School of Electrical and Electronic Engineering, Queens University of Belfast, Ashby Building, Stranmillis Road, Belfast, BT9 5AH, UK; [email protected] Ingrid Verbauwhede, Katholieke Universiteit Leuven, ESAT-COSIC, Kasteelpark Arenberg 10, 3001 Leuven, Belgium; [email protected] Erwin de Kock, Philips Research, High Tech Campus 31, 5656 AE Eindhoven, The Netherlands; [email protected]

http://www.hindawi.com EURASIP JOURNAL ON APPLIED SIGNAL PROCESSING

Special Issue on Video Adaptation for Heterogeneous Environments

CALL FOR PAPERS The explosive growth of compressed video streams and repositories accessible worldwide, the recent addition of new video-related standards such as H.264/AVC, MPEG-7, and MPEG-21, and the ever-increasing prevalence of heterogeneous, video-enabled terminals such as computer, TV, mobile phones, and personal digital assistants have escalated the need for efficient and effective techniques for adapting compressed videos to better suit the different capabilities, constraints, and requirements of various transmission networks, applications, and end users. For instance, Universal Multimedia Access (UMA) advocates the provision and adaptation of the same multimedia content for different networks, terminals, and user preferences. Video adaptation is an emerging field that offers a rich body of knowledge and techniques for handling the huge variation of resource constraints (e.g., bandwidth, display capability, processing speed, and power consumption) and the large diversity of user tasks in pervasive media applications. Considerable amounts of research and development activities in industry and academia have been devoted to answering the many challenges in making better use of video content across systems and applications of various kinds. Video adaptation may apply to individual or multiple video streams and may call for different means depending on the objectives and requirements of adaptation. Transcod- ing, transmoding (cross-modality transcoding), scalable content representation, content abstraction and summarization are popular means for video adaptation. In addition, video content analysis and understanding, including low-level feature analysis and high-level semantics understanding, play an important role in video adaptation as essential video content can be better preserved. The aim of this special issue is to present state-of-the-art developments in this flour- ishing and important research field. Contributions in theoretical study, architecture design, performance analysis, complexity reduction, and real-world applications are all welcome. Topics of interest include (but are not limited to): • Heterogeneous video transcoding • Scalable video coding • Dynamic bitstream switching for video adaptation • Signal, structural, and semantic-level video adaptation • Content analysis and understanding for video adaptation • Video summarization and abstraction • Copyright protection for video adaptation • Crossmedia techniques for video adaptation • Testing, field trials, and applications of video adaptation services • International standard activities for video adaptation

Authors should follow the EURASIP JASP manuscript format described at the journal site http://www.hindawi.com/journals/asp/. Prospective authors should submit an electronic copy of their complete manuscript through the EURASIP JASP manuscript tracking system at http://www.hindawi.com/mts, according to the following timetable:

Manuscript Due September 1, 2006 Acceptance Notiﬁcation January 1, 2007 Final Manuscript Due April 1, 2007 Publication Date 3rd Quarter 2007

GUEST EDITORS: Chia-Wen Lin, Department of Computer Science and Information Engineering, National Chung Cheng University, Chiayi 621, Taiwan; [email protected] Yap-Peng Tan, School of Electrical and Electronic Engineering, Nanyang Technological University, Nanyang Avenue, Singapore 639798, Singapore; [email protected] Ming-Ting Sun, Department of Electrical Engineering, University of Washington, Seattle, WA 98195, USA; [email protected] Alex Kot, School of Electrical and Electronic Engineering, Nanyang Technological University, Nanyang Avenue, Singapore 639798, Singapore; [email protected] Anthony Vetro, Mitsubishi Electric Research Laboratories, 201 Broadway, 8th Floor, Cambridge, MA 02138, USA; [email protected]

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Special Issue on Advanced Signal Processing and Computational Intelligence Techniques for Power Line Communications

CALL FOR PAPERS In recent years, increased demand for fast Internet access and new multimedia services, the development of new and feasible signal processing techniques associated with faster and low-cost digital signal processors, as well as the deregulation of the telecommunications market have placed major emphasis on the value of investigating hostile media, such as powerline (PL) channels for high-rate data transmissions. Nowadays, some companies are offering powerline communications (PLC) modems with mean and peak bit-rates around 100 Mbps and 200 Mbps, respectively. However, advanced broadband powerline communications (BPLC) modems will surpass this performance. For accomplishing it, some special schemes or solutions for coping with the following issues should be addressed: (i) considerable differences between powerline network topologies; (ii) hostile properties of PL channels, such as attenuation proportional to high frequencies and long distances, high-power impulse noise occurrences, time-varying behavior, and strong inter-symbol interference (ISI) effects; (iv) electromagnetic compatibility with other well-established communication systems working in the same spectrum, (v) climatic conditions in different parts of the world; (vii) reliability and QoS guarantee for video and voice transmissions; and (vi) different demands and needs from developed, developing, and poor countries. These issues can lead to exciting research frontiers with very promising results if signal processing, digital communication, and computational intelligence techniques are effectively and efficiently combined. The goal of this special issue is to introduce signal processing, digital communication, and computational intelligence tools either individually or in combined form for advancing reliable and powerful future generations of powerline communication solutions that can be suited with for applications in developed, developing, and poor countries. Topics of interest include (but are not limited to) • Multicarrier, spread spectrum, and single carrier techniques • Channel modeling • Channel coding and equalization techniques • Multiuser detection and multiple access techniques • Synchronization techniques • Impulse noise cancellation techniques • FPGA, ASIC, and DSP implementation issues of PLC modems • Error resilience, error concealment, and Joint source-channel design methods for video transmission through PL channels

Authors should follow the EURASIP JASP manuscript format described at the journal site http://asp.hindawi.com/. Prospective authors should submit an electronic copy of their complete manuscripts through the EURASIP JASP manuscript tracking system at http://www.hindawi.com/mts, according to the following timetable:

Manuscript Due October 1, 2006 Acceptance Notiﬁcation January 1, 2007 Final Manuscript Due April 1, 2007 Publication Date 3rd Quarter, 2007

GUEST EDITORS: Moises´ Vidal Ribeiro, Federal University of Juiz de Fora, Brazil; [email protected] Lutz Lampe, University of British Columbia, Canada; [email protected] Sanjit K. Mitra, University of California, Santa Barbara, USA; [email protected] Klaus Dostert, University of Karlsruhe, Germany; [email protected] Halid Hrasnica, Dresden University of Technology, Germany; [email protected]

Volume 2006 Contents and Abstracts

On the Use of Pade´ Approximation for Open Access Performance Evaluation of Maximal Ratio Combining Diversity over Weibull Fading Channels

Mahmoud H. Ismail and Mustafa M. Matalgah DOI: 10.1155/WCN/2006/58501 We use the Pade´ approximation (PA) technique to obtain closed-form approximate expressions for the moment-generating function (MGF) of the Weibull random variable. Unlike previously obtained closed-form exact expressions for the MGF, which are relatively complicated as being given in terms of the Meijer G-function, PA can be used to obtain simple rational expressions for the MGF, which can be easily used in further computations. We illustrate the accuracy of the PA technique by comparing its results to either the existing exact MGF or to that obtained via Monte Carlo simulations. Using the approximate expressions, we analyze the performance of digital modulation schemes over the single channel and the multichannels employing maximal ratio combining (MRC) under the Weibull fading assumption. Our results show excellent agreement with previously published results as well as with simulations.

A New MAC Protocol with Pseudo-TDMA Open Access Behavior for Supporting Quality of Service in 802.11 Wireless LANs Georgios S. Paschos, Ioannis Papapanagiotou, Stavros A. Kotsopoulos, and George K. Karagiannidis DOI: 10.1155/WCN/2006/65836 A new medium access control (MAC) protocol is proposed for quality-of-service (QoS) support in wireless local area networks (WLAN). The protocol is an alternative to the recent enhancement 802.11e. A new priority policy provides the system with better performance by simulating time division multiple access (TDMA) functionality. Collisions are reduced and starvation of low-priority classes is prevented by a distributed admission control algorithm. The model performance is found analytically extending previous work on this matter. The results show that a better organization of resources is achieved through this scheme. Throughput analysis is veriﬁed with OPNET simulations. Regular Issue, Vol. 2006 137

Joint Frequency Ambiguity Resolution and Open Access Accurate Timing Estimation in OFDM Systems with Multipath Fading Jun Li, Guisheng Liao, and Shan Ouyang DOI: 10.1155/WCN/2006/62173 A serious disadvantage of orthogonal frequency-division multiplexing (OFDM) is its sensitivity to carrier frequency offset (CFO) and timing offset (TO). For many low-complexity algorithms, the estimation ambiguity exists when the CFO is greater than one or two subcarrier spacing, and the estimated TO is also prone to exceeding the ISI-free interval within the cyclic prefix (CP). This paper presents a method for joint CFO ambiguity resolution and accurate TO estimation in multipath fading. Maximum-likelihood (ML) principle is employed and only one pilot symbol is needed. Frequency ambiguity is resolved and accurate TO can be obtained simultaneously by using the fast Fourier transform (FFT) and one-dimensional (1D) search. Both known and unknown channel order cases are considered. Computer simulations show that the proposed algorithm outperforms some others in the multipath fading channels.

An Implementation of Nonlinear Multiuser Open Access Detection in Rayleigh Fading Channel Wai Yie Leong, John Homer, and Danilo P.Mandic DOI: 10.1155/WCN/2006/45647 A blind nonlinear interference cancellation receiver for code-division multiple-access- (CDMA-) based communication systems operating over Rayleigh ﬂat-fading channels is proposed. The receiver which assumes knowledge of the signature waveforms of all the users is implemented in an asynchronous CDMA environment. Unlike the conventional MMSE receiver, the proposed blind ICA multiuser detector is shown to be robust without training sequences and with only knowledge of the signature waveforms. It has achieved nearly the same performance of the conventional training-based MMSE receiver. Several compar- isons and experiments are performed based on examining BER performance in AWGN and Rayleigh fading in order to verify the validity of the proposed blind ICA multiuser detector.

Energy-Efﬁcient Channel Estimation in Open Access MIMO Systems Sarod Yatawatta, Athina P.Petropulu, and Charles J. Graff DOI: 10.1155/WCN/2006/27694 The emergence of MIMO communications systems as practical high-data-rate wireless communications systems has created several technical challenges to be met. On the one hand, there is potential for enhancing system performance in terms of capacity and diversity. On the other hand, the presence of multiple transceivers at both ends has created additional cost in terms of hardware and energy consumption. For coherent detection as well as to do optimization such as water ﬁlling and beamforming, it is essential that the 138 Regular Issue, Vol. 2006

MIMO channel is known. However, due to the presence of multiple transceivers at both the transmitter and receiver, the channel estimation problem is more complicated and costly compared to a SISO system. Several solutions have been proposed to minimize the computational cost, and hence the energy spent in channel estimation of MIMO systems. We present a novel method of minimizing the overall energy consumption. Unlike existing methods, we consider the energy spent during the channel estimation phase which includes transmission of training symbols, storage of those symbols at the receiver, and also channel estimation at the receiver. We develop a model that is independent of the hardware or software used for channel estimation, and use a divide-and-conquer strategy to minimize the overall energy consumption.

Differential Detection of Space-Time Spreading Open Access with Two Transmit Antennas Tao Shi and Lei Cao DOI: 10.1155/WCN/2006/70509 Adifferential detection scheme for space-time spreading with two transmit antennas is proposed. The scheme does not require channel state information at either the transmitter or the receiver. With segmentation and preamble symbols padded at the transmitter, the receiver recovers the information using differential detection. Both phase-shift keying (PSK) and quadrature amplitude modulation (QAM) signals are considered. The proposed scheme achieves two-level transmit diversity gain with low complexity and saves the use of channel estimation, while having about 3 dB performance loss as compared to the coherent detection scheme. When multiple receive antennas exist, additional receive diversity gain can be achieved along with the transmit diversity gain. The scheme works fine under block-fading channel as well as slow Rayleigh fading channel, which is a popular scenario for high-rate data communications. The system performance for different segment sizes, channel fading speeds, modulation methods, and numbers of receive antennas is studied through simulations.

Decision-Directed Recursive Least Squares Open Access MIMO Channels Tracking Ebrahim Karami and Mohsen Shiva DOI: 10.1155/WCN/2006/43275 A new approach for joint data estimation and channel tracking for multiple-input multiple- output (MIMO) channels is proposed based on the decision-directed recursive least squares (DD-RLS) algorithm. RLS algorithm is commonly used for equalization and its application in channel estimation is a novel idea. In this paper, after defining the weighted least squares cost function it is minimized and eventually the RLS MIMO channel estimation algorithm is derived. The proposed algorithm combined with the decision-directed algorithm (DDA) is then extended for the blind mode operation. From the computational complexity point of view being O(3) versus the number of transmitter and receiver antennas, the proposed algorithm is very efficient. Through various simulations, the mean square error (MSE) of Regular Issue, Vol. 2006 139 the tracking of the proposed algorithm for different joint detection algorithms is compared with Kalman filtering approach which is one of the most well-known channel tracking algorithms. It is shown that the performance of the proposed algorithm is very close to Kalman estimator and that in the blind mode operation it presents a better performance with much lower complexity irrespective of the need to know the channel model.

A Multicarrier Multiplexing Method for Very Wide Open Access Bandwidth Transmission Diakoumis Gerakoulis and George Efthymoglou DOI: 10.1155/WCN/2006/64253 The multicarrier orthogonal code division multiplexing (MC-OCDM) introduced here has been designed for very wide bandwidth (VWB) point-to-point and point-to-multipoint transmission. In order to meet VWB transmission requirements, the MC-OCDM design has two components, the basic and the composite. The basic MC-OCDM is a generalized form of the standard orthogonal frequency division multiplexing (OFDM). It has the property of distributing the power of each transmitted symbol into all subcarrier frequencies. Each subcarrier will then carry all transmitted symbols which are distinguished by orthogonal Hadamard sequences. The resulting system is shown to improve the performance of OFDM by introducing frequency and time diversity. As shown, by both analysis and simulation, the basic MC-OCDM combats the effects of narrowband interference (NBI). In particular, the simulation results show that the BER performance of the basic MC-OCDM in the presence of NBI is better than OFDM for both coded and uncoded systems. Further- more, the composite MC-OCDM is a method of orthogonal frequency division multiplexing (OFDM) basic MC-OCDM channels. This allows us to multiplex more than one basic MC-OCDM channel into a VWB transmission system which can have the performance and spectral efficiency required in fixed wireless transmission environments.

On the Geometrical Characteristics of Open Access Three-Dimensional Wireless Ad Hoc Networks and Their Applications Guansheng Li, Pingyi Fan, and Kai Cai DOI: 10.1155/WCN/2006/31467 In a wireless ad hoc network, messages are transmitted, received, and forwarded in a finite geometrical region and the transmission of messages is highly dependent on the loca- tions of the nodes. Therefore the study of geometrical relationship between nodes in wireless ad hoc networks is of fundamental importance in the network architecture design and performance evaluation. However, most previous works concentrated on the networks deployed in the two-dimensional region or in the infinite three-dimensional space, while in many cases wireless ad hoc networks are deployed in the finite three-dimensional space. In this paper, we analyze the geometrical characteristics of the three-dimensional wireless ad hoc network in a finite space in the framework of random graph and deduce an expression to calculate the distance probability distribution between network nodes that are independently and uniformly distributed in a finite cuboid space. Based on the theoretical 140 Regular Issue, Vol. 2006 result, we present some meaningful results on the finite three-dimensional network performance, including the node degree and the max-flow capacity. Furthermore, we investigate some approximation properties of the distance probability distribution function derived in the paper.

Equalization of Sparse Intersymbol-Interference Open Access Channels Revisited Jan Mietzner, Sabah Badri-Hoeher, Ingmar Land, and Peter A. Hoeher DOI: 10.1155/WCN/2006/29075 Sparse intersymbol-interference (ISI) channels are encountered in a variety of communication systems, especially in high-data-rate systems. These channels have a large memory length, but only a small number of significant channel coefficients. In this paper, equalization of sparse ISI channels is revisited with focus on trellis-based techniques. Due to the large channel memory length, the complexity of maximum-likelihood sequence estimation by means of the Viterbi algorithm is normally prohibitive. In the first part of the paper, a unified framework based on factor graphs is presented for complexity reduction without loss of optimality. In this new context, two known reduced-complexity trellis-based techniques are recapitulated. In the second part of the paper a simple alternative approach is investigated to tackle general sparse ISI channels. It is shown that the use of a linear filter at the receiver renders the application of standard reduced-state trellis-based equalization techniques feasible without significant loss of optimality.

A Conjugate-Cyclic-Autocorrelation Open Access Projection-Based Algorithm for Signal Parameter Estimation Valentina De Angelis, Luciano Izzo, Antonio Napolitano, and Mario Tanda DOI: 10.1155/WCN/2006/86026 A new algorithm to estimate amplitude, delay, phase, and frequency offset of a received signal is presented. The frequency-offset estimation is performed by maximizing, with respect to the conjugate cycle frequency, the projection of the measured conjugate-cyclic- autocorrelation function of the received signal over the true conjugate second-order cyclic autocorrelation. It is shown that this estimator is mean-square consistent, for moderate values of the data-record length, outperforms a previously proposed frequency-offset estimator, and leads to mean-square consistent estimators of the remaining parameters.

A General Theory for SIR Balancing Open Access Holger Boche and Martin Schubert DOI: 10.1155/WCN/2006/60681 We study the problem of maximizing the minimum signal-to-interference ratio (SIR) in a multiuser system with an adaptive receive strategy. The interference of each user is modelled Regular Issue, Vol. 2006 141 by an axiomatic framework, which reflects the interaction between the propagation channel, the power allocation, and the receive strategy used for interference mitigation. Assum- ing that there is a one-to-one mapping between the QoS and the signal-to-interference ratio (SIR), the feasible QoS region is completely characterized by the max-min SIR balancing problem. In the first part of the paper, we derive fundamental properties of this problem for the most general case, when interference is modelled with an axiomatic framework. In the second part, we show more specific properties for interference functions based on a nonnegative coupling matrix. The principal aim of this paper is to provide a deeper understanding of the interaction between power allocation and interference mitigation strategies. We show how the proposed axiomatic approach is related to the matrix-based theory.

Voice and Video Telephony Services in Open Access Smartphone Valeria Loscri’,Mauro Tropea, and Salvatore Marano DOI: 10.1155/WCN/2006/84945 Multimedia telephony is a delay-sensitive application. Packet losses, relatively less critical than delay, are allowed up to a certain threshold. They represent the QoS constraints that have to be respected to guarantee the operation of the telephony service and user satis- faction. In this work we introduce a new smartphone architecture characterized by two process levels called application processor (AP) and mobile termination (MT), respectively. Here, they communicate through a serial channel. Moreover, we focus our attention on two very important UMTS services: voice and video telephony. Through a simulation study the impact of voice and video telephony is evaluated on the structure considered using the protocols known at this moment to realize voice and video telephony.

Efficient Low Bit-Rate Low-Latency Open Access Channelization in DECT Rohit Budhiraja and Bhaskar Ramamurthi DOI: 10.1155/WCN/2006/54148 In a TDMA standard such as DECT, low bit-rate transmission is feasible either at the cost of efficiency (shorter slots with fixed overhead per slot) or increased latency (longer frames). This paper proposes a new scheme for low bit-rate low-latency channelization in the DECT standard, in which data can be efficiently transmitted at rates as low as 10 kbps. This could be useful for sending acknowledgments for a high-speed data communication link, or for vocoder/VoIP traffic. The proposed scheme enables efficient low bit-rate transmission by di- viding a DECT channel into four subbands, and by employing a new slot structure wherein TDMA overhead is kept to a minimum. It is shown that the proposed scheme can coexist with the DECT system and can be implemented using existing IMT-2000 DECT hardware with minor modifications. A comparison is also made of the proposed scheme with existing options for low bit-rate channelization in DECT. 142 Regular Issue, Vol. 2006

Energy-Efficient Medium Access Control Open Access Protocols for Wireless Sensor Networks Qingchun Ren and Qilian Liang DOI: 10.1155/WCN/2006/39814 A key challenge for wireless sensor networks is how to extend network lifetime with dynamic power management on energy-constraint sensor nodes. In this paper, we propose two energy-efficient MAC protocols: asynchronous MAC (A-MAC) protocol and asynchronous schedule-based MAC (ASMAC) protocol. A-MAC and ASMAC protocols are attractive due to their suitabilities for multihop networks and capabilities of removing accu- mulative clock-drifts without any network synchronization. Moreover, we build a traffic- strength- and network-density-based model to adjust essential algorithm parameters adaptively. Simulation results show that our algorithms can successfully acquire the optimum values of power-on/off duration, schedule-broadcast interval, as well as super-time-slot size and order. These algorithm parameters can ensure adequate successful transmission rate, short waiting time, and high energy utilization. Therefore, not only the performance of network is improved but also its lifetime is extended when A-MAC or ASMAC is used.

Energy Efficient AODV Routing in CDMA Open Access Ad Hoc Networks Using Beamforming Nie Nie and Cristina Comaniciu DOI: 10.1155/WCN/2006/76709 We propose an energy aware on-demand routing protocol for CDMA mobile ad hoc networks, for which improvements in the energy consumption are realized by both introducing an energy-based routing measure and by enhancing the physical layer performance using beamforming. Exploiting the cross-layer interactions between the network and the physical layer leads to a significant improvement in the energy efficiency compared with the traditional AODV protocol, and provides an alternative solution of link breakage detection in traditional AODV protocol. Several performance measures are considered for evaluating the network performance, such as data energy consumption, latency, and overhead energy consumption. An optimum SIR threshold range is determined experimentally for various implementation scenarios. EURASIP Journal on Wireless Communications and Networking, Volume 2006 © 2006 Hindawi Publishing Corporation

Special Issue on Quality of Service in Mobile Ad Hoc Networks

Wei Li Department of Electrical Engineering and Computer Science, University of Toledo, Toledo, OH 43606, USA Mohsen Guizani Department of Computer Science, Western Michigan University, Kalamazoo, MI 49008, USA

Demetrios Kazakos Department of Electrical and Computer Engineering, University of Idaho, Moscow, ID 83844, USA

Mobile ad hoc networking is a challenging task due to the lack of resources residing in the network as well as the frequent changes in network topology. Although much research has been directed to supporting quality of service (QoS) in the Internet and traditional wireless networks, present results are not suitable for mobile ad hoc network (MANET). QoS support for mobile ad hoc networks remains an open problem, drawing interest from both academia and industry under military and commercial sponsorship. MANETs have certain unique characteristics that pose several difficulties in provisioning QoS, such as dynamically varying network topology, lack of precise state information, lack of central control, error-prone shared radio channels, limited resource availability, hidden terminal problems, and insecure media, and little consensus yet exists on which approaches may be optimal. Future MANETs are likely to be “multimode” or heterogeneous in nature. Thus, the routers comprising a MANET will employ multiple, physical-layer wireless technologies, with each new technology requiring a multiple access (MAC) protocol for supporting QoS. Above the MAC layer, forwarding, routing, signaling, and admission control policies are required, and the best combination of these policies will change as the underlying hardware technology evolves. In response to the above demand for mobile ad hoc networks, this special issue aims at providing a timely and concise reference of the current activities and findings in the relevant technical fields, and focuses as well on the state-of-the-art and up-to-date efforts in design, performance analysis, implementation and experimental results for various QoS issues in MANETs. We believe that all of these papers not only provide novel ideas, new analytical models, simulation and experimental results, and handful experience in this field, but also simulate 144 EURASIP Journal on Wireless Communications and Networking, Vol. 2006 the future research activities in the area of the quality of service for mobile ad hoc networks. Abriefsummaryofeachpaperislistedasfollows. The first paper by Qi He et al. first identifies two critical issues leading to the TCP performance degradation: (1) unreliable broadcast, since broadcast frames are transmitted without the request-to-send and clear-to-send (RTS/CTS) dialog and Data/ACK handshake, so they are vulnerable to the hidden terminal problem; and (2) false link failure which occurs when a node cannot successfully transmit data temporarily due to medium contention. Sec- ondly, the authors propose a scheme to use a narrow-bandwidth, out-of-band busy-tone channel to make reservation for broadcast and link error detection frames only. The proposed scheme is simple and power efficient, because only the sender needs to transmit two short messages in the busy tone channel before sending broadcast or link error detection frames in the data channel. Analytical results show that the proposed scheme can dramatically reduce the collision probability of broadcast and link error detection frames. Exten- sive simulations with different network topologies further demonstrate that the proposed scheme can improve TCP throughput by 23% to 150%, depending on user mobility, and effectively enhance both short-term and long-term fairness among coexisting TCP flows in multi-hop wireless ad hoc networks. The second paper by Deying Li et al. discusses the energy efficient QoS topology control problem for nonhomogeneous ad hoc wireless networks. Given a set of nodes with different energy and bandwidth capacities in a plane, and given the end-to-end traffic demands and delay bounds between node pairs, the problem is to find a network topology that can meet the QoS requirements, and the maximum energy utilization of nodes is minimized. Achieving this objective is vital to the increase of network lifetime. We consider two cases of the problem: (1) the traffic demands are not splittable, and (2) the traffic demands are splittable. For the former case, the problem is formulated as an integer linear programming problem. For the latter case, the problem is formulated as a mixed integer programming problem, and an optimal algorithm has been proposed to solve the problem. The third paper by Hsiao-Hwa Chen et al. proposes autonomous power control MAC protocol (APCMP), which allows mobile nodes dynamically adjusting power level for transmitting DATA/ACK according to the distances between the transmitter and its neighbors. In addition, the power level for transmitting RTS/CTS is also adjustable according to the power level for DATA/ACK packets. In this paper, the performance of APCMP protocol is evaluated by simulation and is compared with that of other protocols. The fourth paper by Yang Yang et al. considers the hybrid problem of the infrastructure and the ad hoc modes in WLAN. They propose in this paper a new coverage improvement scheme that can identify suitable idle MSs in good service zones as trafficagents(TAs)to relay traffic from those out-of-coverage MSs to the AP. The service coverage area of WLAN is then expanded. The QoS requirements (e.g., bandwidth) of those MSs are considered in the selection process of corresponding TAs. Mathematical analysis, verified by computer simulations, shows that the proposed TA scheme can effectively reduce blocking probability when traffic load is light. The fifth paper by S. Ahmed et al. analyzes the performance differentials to compare the commonly used ad hoc network routing protocols. They also analyze the performance over varying loads for each of those protocols using OPNET modeler 10.5. Their findings show that for specific differentials, TORA shows better performance over the two on-demand protocols, that is, dynamic source routing and ad hoc on-demand distance vector routing. Special Issue on Quality of Service in Mobile Ad Hoc Networks 145

Their findings are expected to lead to further performance improvements of various ad hoc networks in the future. The sixth paper by Nagaraja Thanthry et al. analyzes various parameters that affect the performance of TCP in an ad hoc network environment. Congestion and path nonavail- ability are two major factors that affect TCP performance. It was also observed that, in the presence of multiple paths, TCP performance degrades when one of the paths used for forwarding data drops a packet. In the current paper, the authors have proposed establishing multiple connections for every data transfer between the source and the destination. The proposed mechanism would be transparent to the application and session layers; however, it involves the transport layer in multipath routing scheme. The seventh paper by X. Wang et al. develops a modified version that we term CSMA/ CCA (CSMA with copying collision avoidance) in order to mitigate fairness issues arising with CSMA/CA. A station in CSMA/CCA contends for the shared wireless medium by employing a binary exponential backoff similar to CSMA/CA. Different from CSMA/CA, CSMA/CCA copies the contention window (CW) size piggybacked in the MAC header of an overheard data frame within its basic service set (BSS), and updates its backoff counter according to the new CW size. Simulations carried out in several WLAN configurations illustrate that CSMA/CCA improves fairness relative to CSMA/CA and offers considerable advantages for deployment in the 802.11 standard based WLANs.1 The eighth paper by S. Guizani et al. proposes a new technique to compensate the chromatic dispersion optically by applying Talbot effect. Results obtained are inline with what’s proposed. This method is easy to implement and versatile since any type of fiber can be used. Moreover, our technique has the strength to revive a totally deformed signal regardless of the bits transmitted. In closing, we would like to thank the support from the Editor-in-Chief, Phillip Regalia, and the contributions from authors and reviewers, to make this special issue possible.

Wei Li Mohsen Guizani Demetrios Kazakos 146 EURASIP Journal on Wireless Communications and Networking, Vol. 2006 Volume 2006, No. 2 Contents and Abstracts

Improving TCP Performance over Wireless Ad Hoc Open Access Networks with Busy Tone Assisted Scheme

Qi He, Lin Cai, Xuemin (Sherman) Shen, and Pinhan Ho DOI: 10.1155/WCN/2006/51610 It is well known that transmission control protocol (TCP) performance degrades severely in IEEE 802.11-based wireless ad hoc networks. We first identify two critical issues leading to the TCP performance degradation: (1) unreliable broadcast, since broadcast frames are transmitted without the request-to-send and clear-to-send (RTS/CTS) dialog and Data/ACK handshake, so they are vulnerable to the hidden terminal problem; and (2) false link failure which occurs when a node cannot successfully transmit data temporarily due to medium contention. We then propose a scheme to use a narrow- bandwidth, out-of-band busy tone channel to make reservation for broadcast and link error detection frames only. The proposed scheme is simple and power efficient, because only the sender needs to transmit two short messages in the busy tone channel before sending broadcast or link error detection frames in the data channel. Analytical results show that the proposed scheme can dramatically reduce the collision probability of broadcast and link error detection frames. Extensive simulations with different network topologies further demonstrate that the proposed scheme can improve TCP throughput by 23% to 150%, depending on user mobility, and effectively enhance both short-term and long-term fairness among coexisting TCP flows in multihop wireless ad hoc networks.

QoS Topology Control for Nonhomogenous Open Access Ad Hoc Wireless Networks Deying Li, Xiaohua Jia, and Hongwei Du DOI: 10.1155/WCN/2006/82417 This paper discusses the energy-efficient QoS topology control problem for nonhomogenous ad hoc wireless networks. Given a set of nodes with different energy and bandwidth capacities in a plane, and given the end-to-end traffic demands and delay bounds between node-pairs, the problem is to find a network topology that can meet the QoS requirements and the maximum energy utilization of nodes is minimized. Achieving this objective is vital to the increase of network lifetime. We consider two cases of the problem: (1) the trafficde- mands are not splittable, and (2) the traffic demands are splittable. For the former case, the problem is formulated as an integer linear programming problem. For the latter case, the problem is formulated as a mixed integer programming problem, and an optimal algorithm has been proposed to solve the problem. Special Issue on Quality of Service in Mobile Ad Hoc Networks 147

Autonomous Power Control MAC Protocol for Open Access Mobile Ad Hoc Networks Hsiao-Hwa Chen, Zhengying Fan, and Jie Li DOI: 10.1155/WCN/2006/36040 Battery energy limitation has become a performance bottleneck for mobile ad hoc networks. IEEE 802.11 has been adopted as the current standard MAC protocol for ad hoc networks. However, it was developed without considering energy efficiency. To solve this problem, many modifications on IEEE 802.11 to incorporate power control have been proposed in the literature. The main idea of these power control schemes is to use a maximum possible power level for transmitting RTS/CTS and the lowest acceptable power for sending DATA/ACK. However, these schemes may degrade network throughput and reduce the overall energy efficiency of the network. This paper proposes autonomous power control MAC protocol (APCMP), which allows mobile nodes dynamically adjusting power level for transmitting DATA/ACK according to the distances between the transmitter and its neighbors. In addition, the power level for transmitting RTS/CTS is also adjustable according to the power level for DATA/ACK packets. In this paper, the performance of APCMP protocol is evaluated by simulation and is compared with that of other protocols.

Traffic Agents for Improving QoS in Mixed Open Access Infrastructure and Ad Hoc Modes Wireless LAN Yang Yang, Hai-Feng Yuan, Hsiao-Hwa Chen, Wen-Bing Yao, and Yong-Hua Song DOI: 10.1155/WCN/2006/94235 As an important complement to infrastructured wireless networks, mobile ad hoc networks (MANET) are more flexible in providing wireless access services, but more difficult in meeting different quality of service (QoS) requirements for mobile customers. Both infrastructure and ad hoc network structures are supported in wireless local area networks (WLAN), which can offer high data-rate wireless multimedia services to the mobile stations (MSs) in a limited geographical area. For those out-of-coverage MSs, how to effectively connect them to the access point (AP) and provide QoS support is a challenging issue. By mixing the infrastructure and the ad hoc modes in WLAN, we propose in this paper a new coverage improvement scheme that can identify suitable idle MSs in good service zones as traffic agents (TAs) to relay traffic from those out-of-coverage MSs to the AP. The service coverage area of WLAN is then expanded. The QoS requirements (e.g., bandwidth) of those MSs are considered in the selection process of corresponding TAs. Mathematical analysis, verified by computer simulations, shows that the proposed TA scheme can effectively reduce blocking probability when traffic load is light. 148 EURASIP Journal on Wireless Communications and Networking, Vol. 2006

Performance Evaluation of Important Ad Hoc Open Access Network Protocols S. Ahmed and M. S. Alam DOI: 10.1155/WCN/2006/78645 A wireless ad hoc network is a collection of specific infrastructureless mobile nodes forming a temporary network without any centralized administration. A user can move anytime in an ad hoc scenario and, as a result, such a network needs to have routing protocols which can adopt dynamically changing topology. To accomplish this, a number of ad hoc routing protocols have been proposed and implemented, which include dynamic source routing (DSR), ad hoc on-demand distance vector (AODV) routing, and temporally ordered routing algorithm (TORA). Although considerable amount of simulation work has been done to measure the performance of these routing protocols, due to the constant changing nature of these protocols, a new performance evaluation is essential. Accordingly, in this paper, we analyze the performance differentials to compare the above-mentioned commonly used ad hoc network routing protocols. We also analyzed the performance over varying loads for each of these protocols using OPNET Modeler 10.5. Our findings show that for specific differentials, TORA shows better performance over the two on-demand protocols, that is, DSR and AODV. Our findings are expected to lead to further performance improvements of various ad hoc networks in the future.

TCP-M: Multiflow Transmission Control Open Access Protocol for Ad Hoc Networks Nagaraja Thanthry, Anand Kalamkar, and Ravi Pendse DOI: 10.1155/WCN/2006/95149 Recent research has indicated that transmission control protocol (TCP) in its base form does not perform well in an ad hoc environment. The main reason identified for this behavior involves the ad hoc network dynamics. By nature, an ad hoc network does not support any form of quality of service. The reduction in congestion window size during packet drops, a property of the TCP used to ensure guaranteed delivery, further deteriorates the overall performance. While other researchers have proposed modifying congestion window properties to improve TCP performance in an ad hoc environment, the authors of this paper propose using multiple TCP flows per connection. The proposed protocol reduces the influence of packet drops that occurred in any single path on the overall system performance. The analysis carried out by the authors indicates a significant improvement in overall performance.

CSMA/CCA: A Modified CSMA/CA Protocol Open Access Mitigating the Fairness Problem for IEEE 802.11 DCF Xin Wang and Georgios B. Giannakis DOI: 10.1155/WCN/2006/39604 Carrier sense multiple access with collision avoidance (CSMA/CA) has been adopted by the IEEE 802.11 standards for wireless local area networks (WLANs). Using a distributed Special Issue on Quality of Service in Mobile Ad Hoc Networks 149 coordination function (DCF), the CSMA/CA protocol reduces collisions and improves the overall throughput. To mitigate fairness issues arising with CSMA/CA, we develop a modified version that we term CSMA with copying collision avoidance (CSMA/CCA). A station in CSMA/CCA contends for the shared wireless medium by employing a binary exponential backoff similar to CSMA/CA. Different from CSMA/CA, CSMA/CCA copies the contention window (CW) size piggybacked in the MAC header of an overheard data frame within its basic service set (BSS) and updates its backoff counter according to the new CW size. Sim- ulations carried out in several WLAN configurations illustrate that CSMA/CCA improves fairness relative to CSMA/CA and offers considerable advantages for deployment in the 802.11-standard-based WLANs.

Fiber over Wireless Chromatic Dispersion Open Access Compensation for a Better Quality of Service S. Guizani, M. Razzak, H. Hamam, Y. Bouslimani, and A. Cheriti DOI: 10.1155/WCN/2006/85980 “Anywhere” and, in particular, “anyhow”: these are the two best words that can describe an ad hoc wireless network that is due to the increasing demand for connectivity in such an information society. Ad hoc wireless networks can be described as dynamic multihop wireless networks with mobile nodes. However, the mobility condition can be relaxed, and we can consider an ad hoc wireless network as a reconfigurable network where all the nodes are connected to the local environment through wireless links, and where there is not a central or dominant node—as opposed to, for example, the case of cellular wireless networks where a base station is located in each cell. When ad hoc networks are backboned by fibers, distortion of the optical link presents one of the major issues. In this paper, we will be addressing one of the fundamental problems, namely, chromatic dispersion in the fiber optic prior reaching the access points. This will ensure an adequate quality of service (QoS). EURASIP JOURNAL ON WIRELESS COMMUNICATIONS AND NETWORKING

Special Issue on Satellite Communications

CALL FOR PAPERS Aims and scope of the Special Issue Arthur C. Clarke published the concept of the 24-hour orbit communications satellite in his article Extra-Terrestrial Relays, which appeared in Wireless World in 1945. Twenty years on, satellites were first employed to provide telecommunication services in the mid-1960s, when they were used by Intelsat to provide telephony and television services using the Early Bird satellite. Mobile-satellite services followed in the late 1970s and were the subject of significant investment in the 1990s, with the introduction of multi-satellite, low earth orbit systems. After a period of consolidation, the satellite communications industry is once more shaping to address new market opportunities brought about by the growth in services such as broadband wireless Internet access, digital broadcasting to fixed and mobile terminals, mobile communications, integrated communications and navigation, and security and dis- aster relief. The Advanced Satellite Mobile Task Force, the Satellite Network of Excellence (SatNEx), the Integral Satcom Initiative (ISI) represent the European response to the new challenges posed by these opportunities: researchers and industries from all over the world are cooperating to identify roadmaps and find new technical and theoretical breakthroughs in this exciting field. This Special Issue aims to showcase the latest research and development advances in this framework. Original and unpublished work is solicited in all aspects of satellite communications including experimental platforms, heterogeneous networks, market surveys, network protocol design, performance evaluation, practical applications, radio interface solutions, system design, theoretical studies, and trial campaigns.

Topics Covered: Topics of interest include (but are not limited to): • Coding, modulation, and synchronisation • Cross-layer solutions • Delay tolerant networks • Fading and interference mitigation techniques • Mobility management • Heterogeneous network solutions • Interworking with terrestrial networks • Medium access control • Network layer solutions • Networkprotocoldesignandevaluation • Propagation measurements and channel modelling • Resource management

Authors should follow the EURASIP JWCN manuscript format described at the journal site http://www.hindawi.com/journals/wcn/. Prospective authors should submit an electronic copy of their complete manuscript through the EURASIP JWCN manuscript tracking system at http://www.hindawi.com/mts/, according to the following timetable:

Manuscript Due September 1, 2006 Acceptance Notiﬁcation January 1, 2007 Final Manuscript Due April 1, 2007 Publication Date 2nd Quarter, 2007

GUEST EDITORS: Ray E. Sheriff, School of Engineering, Design and Technology, University of Bradford, Bradford, West Yorkshire, BD7 1DP, UK; r.e.sheriﬀ@bradford.ac.uk Anton Donner, German Aerospace Center (DLR), Institute of Communications and Navigation, Muenchener Street 20, 82234 Wessling/Oberpfaﬀenhofen, Germany; [email protected] Alessandro Vanelli-Coralli, DEIS/ARCES, University of Bologna, Viale Risorgimento, 2-40136 Bologna, Italy; [email protected]

http://www.hindawi.com EURASIP JOURNAL ON WIRELESS COMMUNICATIONS AND NETWORKING

Special Issue on Novel Techniques for Analysis & Design of Cross-Layer Optimized Wireless Sensor Networks

CALL FOR PAPERS Sensor networks have been researched and deployed for decades already; their wireless extension, however, has witnessed a tremendous upsurge in recent years. This is mainly attributed to the unprecedented operating conditions of wireless sensor networks (WSNs), that is, • a potentially enormous amount of sensor nodes, • reliably operating under stringent energy constraints. The wireless sensor networks’ virtually infinite degrees of freedom have ignited feverish research activities, having led to thousands of publications, white papers and patents in less than a decade, with new contributions emerging on a daily basis. The rich mathematical and technical toolboxes already available from the design of wireless cellular and ad hoc systems clearly aided the birth of new ideas tailored to the problems in WSNs. To date, and this may very well change in forthcoming years, the main problem in de- ploying WSNs is their dependence on scarce battery power. A main design criterion is hence to extend the lifetime of the network without jeopardizing reliable and efficient communications between the sensor nodes as well as from the nodes to one or several data sinks. A prominent example of today’s non-optimized WSN deployment experiences is that the start-up alone costs the network half of its battery power. Optimizing every facet of the communication protocols is hence vital and imperative; such stringent design requirements can be met by a plethora of approaches, for example, optimizing each layer of the protocol stack separately (traditional) or jointly (cross-layer), for each node separately (traditional) or for an ensemble of nodes (distributed and cooperative), and so forth. This has led to copious novel distributed signal processing algorithms, energy-efficient medium access control and fault-tolerant routing protocols, self-organizing and self-healing sensor network mechanisms, and so forth. In the light of the above, the main purpose of this special issue is twofold: • to obtain a coherent and concise technical synthesis from the abundance of recently emerged material in the area of WSNs, • to promote novel approaches in analyzing, designing, and optimizing large-scale WSNs, preferably inspired by approaches from other disciplines, such as physics or biology. As for the first one, very few papers are currently available which synthesize the large amount of fairly dispersed technical contributions; a coherent exposure, also touching upon open research issues, will certainly be appreciated by the academic and industrial research community. As for the second one, we believe that novel approaches, potentially inspired by entirely disjoint disciplines, may help considerably in dealing with networks of thousands of nodes.

Topics of Interest Topics of interest in the area of energy-constraint WSNs include (but are not limited to): • Network capacity w/o imperfections • Joint source and channel coding • Cooperative and distributed signal Data fusion and dta aggregation processing • Novel PHY, MAC, and network paradigms • Cross-layer and cross-functionality design • Security, robustness, and reliability • Self-healing, self-stabilization, and self-organization • Applications, architectures, and topologies • (Macroscopic) information ﬂows • Physically and biologically inspired approaches

Authors should follow the EURASIP JWCN manuscript format described at http:// www.hindawi.com/journals/wcn/. Prospective authors should submit an electronic copy of their complete manuscript through the EURASIP JWCN manuscript tracking system at http://www.hindawi.com/mts/, according to the following timetable:

Manuscript Due October 1, 2006 Acceptance Notiﬁcation February 1, 2007 Final Manuscript Due May 1, 2007 Publication Date 3rd Quarter, 2007

GUEST EDITORS: Mischa Dohler, France Telecom R & D, France; [email protected] Taieb Znati, University of Pittsburgh, USA; [email protected] Stavros Toumpis, University of Cyprus, Cyprus; [email protected] Lionel M. Ni, Hong Kong University of Science and Technology, Hong Kong, China; [email protected]

http://www.hindawi.com EURASIP JOURNAL ON WIRELESS COMMUNICATIONS AND NETWORKING

Special Issue on Trust and Digital Rights Management in Wireless Multimedia Networks and Systems

CALL FOR PAPERS With the widespread infusion of digital technologies and the ensuing ease of digital content transport over the Internet, multimedia data distribution is experiencing exponential growth. The use of emerging technologies and systems based on wireless networks has further facilitated the ubiquitous presence of multimedia data. These rapid advances are neither without cost nor without negative impact. With the increasing sophistication and ubiquity of sharing and dissemination of data over a plethora of networks, the complexity and challenges of untrustworthy behavior as well as cyber attacks may grow significantly. Moreover, the emerging unstructured, mobile, and ad hoc nature of today’s heterogeneous network environment is leading to problems such as the exploitation of resources due to selfish and malicious behavior by users and their agents in the networks. Trust and digital rights management (DRM) of data and the underlying systems and networks have therefore become of critical concern. Moreover, satisfying users’ quality of service (QoS) requirements while implementing trust and DRM mechanisms may overbur- den the already resourceconstrained wireless networks. The objective of this solicitation is to encourage cuttingedge research in trust and digital rights management in wireless networks and systems. Dissemination of research results in formulating the trust and DRM issues, and emerging solutions in terms of technologies, protocols, architecture, and models are expected to contribute to the advancement of this field in a significant way. Topics of interests include but are not limited to:

• DRM issues (copyright protection, tracking, tracing, fingerprinting, authentication, concealment, privacy, access control, etc.) in wireless multimedia • Wireless multimedia traffic modeling, analysis, and management • Tradeoff between QoS, security, dependability, and performability requirements • Context, behavior, and reputation specification, modeling, identification, and management • Trust and DRM models, architectures, and protocols • Trust and DRM in applications (telemedicine, ubiquitous commerce, etc.) • Trust and DRM in wireless ad hoc, mesh, sensor and heterogeneous networks • Trust and DRM technologies for wireless multimedia (digital watermarking, encryption, coding, and compression, and their interplay) • Test beds for experimental evaluation of trust and DRM models.

Manuscript Due December 1, 2006 Acceptance Notiﬁcation April 1, 2007 Final Manuscript Due June 1, 2007 Publication Date 4th Quarter, 2007

GUEST EDITORS: Farid Ahmed, Department of Electrical Engineering and Computer Science The Catholic University of America, Washington, DC, 20064, USA; [email protected] Mohamed Eltoweissy, The Bradley Department of Electrical and Computer Engineering, Virginia Tech, Advanced Research Institute, Arlington, VA 22203, USA; [email protected] Kamesh Namuduri, Department of Electrical and Computer Engineering Wichita State University, Wichitan KS 67260, USA; [email protected]

Volume 2006 Contents and Abstracts

Modular Inverse Algorithms Without Multiplications for Cryptographic Applications Laszlo Hars DOI: 10.1155/ES/2006/32192 Hardware and algorithmic optimization techniques are presented to the left-shift, right- shift, and the traditional Euclidean-modular inverse algorithms. Theoretical arguments and extensive simulations determined the resulting expected running time. On many computational platforms these turn out to be the fastest known algorithms for moderate operand lengths. They are based on variants of Euclidean-type extended GCD algorithms. On the considered computational platforms for operand lengths used in cryptography, the fastest presented modular inverse algorithms need about twice the time of modular multiplications, or even less. Consequently, in elliptic curve cryptography delaying modular divisions is slower (aﬃne coordinates are the best) and the RSA and ElGamal cryptosystems can be accelerated. EURASIP Journal on Embedded Systems, Volume 2006 © 2006 Hindawi Publishing Corporation

Special Issue on Processing with High Complexity: Prototyping and Industrial Design

Markus Rupp Institute for Communication and RF Engineering, Vienna University of Technology, Gusshausstrasse 25/389, 1040 Vienna, Austria Thomas Kaiser Institut fur¨ Kommunikationstechnik, Leibniz Universitat¨ Hannover, Appelstrasse 9a, 30167 Hannover, Germany Jean-Francois Nezan IETR/Image Group Lab, France

Gerhard Schmidt Harman/Becker Automotive Systems, 89077 Ulm, Germany

Some modern applications require an extraordinary large amount of complexity in signal processing algorithms. For example, the 3rd generation of wireless cellular systems is expected to require 1000 times more complexity when compared to its 2nd generation pre- decessors, and future 3GPP standards will aim for even more number-crunching applications. Video and multimedia applications do not only drive the complexity to new peaks in wired and wireless systems but also in personal and home devices. Also in acoustics, modern hearing aids, or algorithms for dereverberation of rooms, blind source separation and multichannel echo cancellation are complexity hungry. At the same time the anticipated products also put on additional constraints like size and power consumption when mobile and thus battery powered. Furthermore, due to new developments in electro-acoustic trans- ducer design, it is possible to design very small and effective loudspeakers. Unfortunately, the linearity assumption does not hold any more for this kind of loudspeakers, leading to computationally demanding nonlinear cancellation and equalization algorithms. Since standard design techniques would either consume too much time or not result in solutions satisfying all constraints, more efficient development techniques are required to speed up this crucial phase. In general such developments are rather expensive due to the required extraordinary high complexity. Thus, de-risking of a future product based on 158 EURASIP Journal on Embedded Systems, Vol. 2006 rapid prototyping is often an alternative approach. However, since prototyping would delay the development, it often makes only sense when it is well embedded in the product design process. Rapid prototyping has thus evolved by applying new design techniques more suitable to support a quick time to market requirement. This special issue focuses on new development methods for applications with high complexity in signal processing and on showing the improved design obtained by such methods. Examples of such methods are virtual prototyping, HW/SW partitioning, automatic design flows, float to fix conversions, and automatic testing and verification. We received seven submissions of which only four were accepted. In Rapid industrial prototyping and SoC design of 3G/4G wireless systems using an HLS methodology the authors Yuanbin Guo et al. present their industrial rapid prototyping experiences on 3G/4G wireless systems using advanced signal processing algorithms in MIMO- CDMA and MIMO-OFDM systems. Advanced receiver algorithms suitable for implementation are proposed for synchronization, MIMO equalization, and detection, VLSI-oriented complexity reduction is presented. This design experience demonstrates that it is possible to enable an extensive architectural analysis in a short time frame using HLS methodology by abstracting the hardware design iterations to an algorithmic C/C++ fixed-point design, which in turn significantly shortens the time to market for wireless systems. In Generation of embedded hardware/software from systemC the authors Salim Ouad- jaout and Dominique Houzet present a design flow to reduce the SoC design cost. This design flow unifies hardware and software using a single high level language and thus decreases the manual errors by rewriting design code. It integrates hardware/software (HW/SW) generation tools and an automatic interface synthesis through a custom library of adapters. The approach is validated on a hardware producer/consumer case study and on the design of a given software-radio communication application. In Efficient design methods for embedded communication systems the authors Martin Holzer et al. analyze a complete design process to exhibit inefficiencies. The lack of an integrated design methodology is argued. High level characterisation, virtual prototyping, automated hardware/software partitioning, and floating-point to fixed-point data conversion are bottlenecks to solve in such a methodology. For each point, authors present and compare several tools and algorithms leading to an efficient fast prototyping framework. Examples are given in the field of high-complexity communication systems but can be extended to other complex application fields. In Fixed-point configurable hardware components the authors Romuald Rocher et al. propose a flexible scheme for fixed-point optimization in order to better exploit advances in VLSI technology. After determining the dynamic range and the binary point, a data wordlength optimization follows by introducing a suitable user-defined cost function. This central cost function, which, for example, depends on chip area and/or energy consumption, is to be minimized under the constraint of a pre-defined thresholded signal-to-quantization noise ratio (SQNR). Through use of analytical models the design time can be significantly reduced. A 128-tap LMS filter design exemplarily explores the fixed-point search space and demonstrates the benefits of the proposed scheme. Markus Rupp Thomas Kaiser Jean-Francois Nezan Gerhard Schmidt Special Issue on Processing with High Complexity: Prototyping and Industrial Design 159 Volume 2006, No. 1 Contents and Abstracts

Rapid Industrial Prototyping and SoC Design of Open Access 3G/4G Wireless Systems Using an HLS Methodology Yuanbin Guo, Dennis McCain, Joseph R. Cavallaro, and Andres Takach DOI: 10.1155/ES/2006/14952 Many very-high-complexity signal processing algorithms are required in future wireless systems, giving tremendous challenges to real-time implementations. In this paper, we present our industrial rapid prototyping experiences on 3G/4G wireless systems using advanced signal processing algorithms in MIMO-CDMA and MIMO-OFDM systems. Core system design issues are studied and advanced receiver algorithms suitable for implementation are proposed for synchronization, MIMO equalization, and detection. We then present VLSI-oriented complexity reduction schemes and demonstrate how to interact these high- complexity algorithms with an HLS-based methodology for extensive design space exploration. This is achieved by abstracting the main effort from hardware iterations to the algorithmic C/C++ fixed-point design. We also analyze the advantages and limitations of the methodology. Our industrial design experience demonstrates that it is possible to enable an extensive architectural analysis in a short-time frame using HLS methodology, which significantly shortens the time to market for wireless systems.

Generation of Embedded Hardware/Software Open Access From SystemC Salim Ouadjaout and Dominique Houzet DOI: 10.1155/ES/2006/18526 Designers increasingly rely on reusing of Intellectual Property (IP) and on raising the level of abstraction to respect System-on-Chip (SoC) market characteristics. However, most hardware and embedded software codes are recoded manually from system level. This recoding step often results in new coding errors that must be identified and debugged. Thus, shorter time to market requires automation of the system synthesis from high level specifications. In this paper, we propose a design flow intended to reduce the SoC design cost. This design flow unifies hardware and software using a single high level language. It integrates hardware/software (HW/SW) generation tools and an automatic interface synthesis through a custom library of adapters. We have validated our interface synthesis approach on a hardware producer/consumer case study and on the design of a given software radio- communication application. 160 EURASIP Journal on Embedded Systems, Vol. 2006

Efficient Design Methods for Embedded Open Access Communication Systems M. Holzer, B. Knerr, P.Belanovic,´ and M. Rupp DOI: 10.1155/ES/2006/64913 Nowadays, design of embedded systems is confronted with complex signal processing algorithms and a multitude of computational intensive multimedia applications, while time to product launch has been extremely reduced. Especially in the wireless domain, those challenges are stacked with tough requirements on power consumption and chip size. Unfortu- nately, design productivity did not undergo a similar progression, and therefore fails to cope with the heterogeneity of modern architectures. Electronic design automation tools exhibit deep gaps in the design flow like high-level characterization of algorithms, floating-point to fixed-point conversion, hardware/software partitioning, and virtual prototyping. This tutorial paper surveys several promising approaches to solve the widespread design problems in this field. An overview over consistent design methodologies that establish a framework for connecting the different design tasks is given. This is followed by a discussion of solutions for the integrated automation of specific design tasks.

Fixed-Point Conﬁgurable Hardware Components Open Access

Romuald Rocher, Daniel Menard, Nicolas Herve, and Olivier Sentieys DOI: 10.1155/ES/2006/23197 To reduce the gap between the VLSI technology capability and the designer productivity, design reuse based on IP (intellectual properties) is commonly used. In terms of arithmetic accuracy, the generated architecture can generally only be configured through the input and output word lengths. In this paper, a new kind of method to optimize fixed-point arithmetic IP has been proposed. The architecture cost is minimized under accuracy constraints defined by the user. Our approach allows exploring the fixed-point search space and the algorithm-level search space to select the optimized structure and fixed-point specification. To significantly reduce the optimization and design times, analytical models are used for the fixed-point optimization process. EURASIP JOURNAL ON EMBEDDED SYSTEMS

Special Issue on Embedded Systems for Intelligent Vehicles

CALL FOR PAPERS The transport sector is seeking new technology to improve safety, driver comfort, and efficient use of infrastructures. Computer vision, range sensors, adaptive control, and networking, among the others, target problems like traffic flow control, pedestrian protection, lane-departure monitoring, smart parking facilities, and driver assistance in general. Em- bedded systems are sought after to implement technologically advanced solutions in smart vehicles. The automotive industry addresses mass markets in which embedded systems have a dramatic impact on the final consumer market price. From the point of view of academic research, intelligent vehicles represent a complete and sufficiently complex benchmark for integrating sensors, actuators, and control to test prototypes of autonomous systems. Ad- ditionally, intelligent vehicles are a challenging environment with a direct applicative aspect for research on autonomous systems, intended as systems reacting in a closed loop with the environment. Topics of interest include smart sensors, sensor fusion, embedded vehicle controls, autonomous vehicles, centralized and local traffic control, GSM and ad hoc networking, Blue- tooth and IEEE 802.15.4 technologies, driver-computer interface, signal processing for embedded environments, autonomous components, and intelligent control. This special issue focuses on new results of research work in the field of embedded systems for intelligent vehicles. Several main keywords are: • Intelligent vehicles • Autonomous vehicles • Embeddedsystemsversusautonomoussystems • Computer vision in embedded systems • Laser/radar range sensors • Multiple sensor embedded architectures • Sensor networks for automotive applications • Vehicle networking • Obstacle detection and tracking • GPS-based navigation • Design methodologies • FPGA for embedded systems with application to intelligent vehicles Authors should follow the EURASIP JES manuscript format described at the journal site http://www.hindawi.com/journals/es/. Prospective authors should submit an electronic copy of their complete manuscript through the EURASIP JES manuscript tracking system at http://www.hindawi.com/mts/, according to the following timetable:

Manuscript Due October 15, 2006 Acceptance Notiﬁcation February 15, 2007 Final Manuscript Due May 15, 2007 Publication Date 3rd Quarter, 2007

GUEST EDITORS: Samir Bouaziz, Institut d’Electronique Fondamentale, Universite´ Paris-Sud XI, Bat.ˆ 220, 91405 Orsay Cedex, France; [email protected] Paolo Lombardi, Institute for the Protection and Security of the Citizen, European Commission Joint Research Centre, TP210, Via Fermi1, 21020 Ispra, Italy; [email protected] Roger Reynaud, Institut d’Electronique Fondamentale, Universite´ Paris-Sud XI, Bat.ˆ 220, 91405 Orsay Cedex, France; [email protected] Gunasekaran S. Seetharaman, Department of Electrical and Computer Engineering, Air Force Institute of Technology, Dayton, OH, 45433, USA; [email protected]

Volume 2006 Contents and Abstracts

An Improved Algorithm for the Piecewise-Smooth Mumford and Shah Model in Image Segmentation Yingjie Zhang DOI: 10.1155/BSB/2006/84397 An improved algorithm for the piecewise-smooth Mumford and Shah functional is presented. Compared to the previous work of Chan and Vese, and Choi et al., extensions of the key functions u± are replaced by updating the level set function based on an artificial image that is composed of the diffused image and the original image. The low convergence problem of the classical algorithm is efficiently solved in the proposed approach. The resulting algorithm has also been demonstrated by several cases. %52!3)0*OURNALON /PEN!CCESS !UDIO 3PEECH AND-USIC0ROCESSING

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HTTPWWWHINDAWICOMJOURNALSIS EURASIP BOOK SERIES ON SIGNAL PROCESSING AND COMMUNICATIONS

EURASIP Book Series on Signal Processing and Communications publishes monographs, edited volumes, and textbooks on signal processing and communications. Published Titles • High-Fidelity Multichannel Audio Coding, Dai Tracy Yang, Chris Kyriakakis, and C.-C. Jay Kuo • Genomic Signal Processing and Statistics, Editedby:EdwardR.Dougherty, Ilya Shmulevich, Jie Chen, and Z. Jane Wang • Smart Antennas—State of the Art, Edited by: Thomas Kaiser, Andr´e Bourdoux, Holger Boche, Javier Rodr´ıguez Fonollosa, Jorgen¨ Bach Andersen, and Wolfgang Utschick • Multimedia Fingerprinting Forensics for Traitor Tracing, K. J. Ray Liu, Wade Trappe, Z. Jane Wang, Min Wu, and Hong Zhao • UWB Communication Systems—A Comprehensive Overview, Edited by: Andreas Molisch, Ian Oppermann, Maria Gabriella Di Benedetto, Domenico Porcino, David Bateman, Phillip Rouzet, and Thomas Kaiser • Advances on Nonlinear Signal and Image Processing, Edited by: Stephen Marshall and Giovanni L. Sicuranza

Forthcoming Titles • Signal Processing for the Acoustic Human-Machine Interface, Walter Kellermann and Herbert Buchner • Genetic and Evolutionary Computation for Image Processing and Analysis, Edited by: Stefano Cagnoni, Evelyne Lutton, and Gustavo Olague

Editor-in-Chief Alex Gershman, Darmstadt University of Technology, Germany; [email protected] Editorial Board Zhi Ding, University of California, USA; [email protected] Moncef Gabbouj, Tampere University of Technology, Finland; moncef.gabbouj@tut.ﬁ Peter Grant, University of Edinburgh, UK; [email protected] Ferran Marqu´es, ETSETB Polytechnic University of Catalonia, Spain; [email protected] Marc Moonen, Katholieke Universiteit Leuven, Belgium; [email protected] Hideaki Sakai, Kyoto University, Japan; [email protected] Giovanni Sicuranza, Universita` di Trieste, Italy; [email protected] Bob Stewart, University of Strathclyde, UK; [email protected] Sergios Theodoridis, University of Athens, Greece; [email protected] For more information, please contact [email protected].

http://www.hindawi.com/books/spc/ Forthcoming Volumes in the EURASIP Book Series on Signal Processing and Communications

Signal Processing for the Acoustic Human-Machine Interface

Walter Kellermann and Herbert Buchner; ISBN: 977-5945-16-X

This monograph presents a coherent treatment of the state of the art in acoustic signal processing for speech and audio, especially for ‘natural’ human-machine interfaces, where users are untethered and mobile. This area has attracted many researchers in recent years due to that novel highly advanced signal processing algorithms promise significant practical benefits. For a direct access to the state of the art, this book is designed as a textbook and reference for graduate students, practicing engineers, and researchers with some background in digital signal processing. The book starts with a discussion of the properties of acoustic systems and speech and audio signals involved in the scenario, and then defines the fundamental problems and basic signal processing concepts for both reproduction and acquisition of such signals. For each of these concepts, we start with single-channel algorithms and then strongly emphasize the increasingly important multichannel techniques, which open the door to new solutions by exploiting the spatial domain. The first of these concepts to describe is the extension of acoustic echo cancellation from the single-channel case to multiple loudspeaker channels, to the combination with microphone arrays, and also to nonlinear echo paths. For the enhancement of speech and audio signals as acquired in noisy and reverberant environments, we first examine single-channel noise reduction and dereverberation schemes. Multichannel schemes are discussed in three chapters on statistical multichannel signal enhancement, classical spatial filtering (supervised beamforming), and blind signal separation for convolutive mixtures, which can be seen as blind beamforming. As an additional building block of natural human-machine interfaces, localization of sources, e.g., as needed for supervised beamforming, is addressed and the current state of the art is reviewed. Finally, the integration of several of the above algorithmic modules into real-world systems is discussed, by way of both general strategies and specific examples.

http://www.hindawi.com/books/spc/

Hindawi Publishing Corporation, 410 Park Avenue, 15th Floor, #287 pmb, INDAWI New York, NY 10022, USA; Fax 1-866-446-3294 (USA, toll-free) Forthcoming Volumes in the EURASIP Book Series on Signal Processing and Communications

Advances on Nonlinear Signal and Image Processing

Edited by: Stephen Marshall and Giovanni L. Sicuranza ISBN: 977-5945-21-6

The interest in nonlinear methods in signal processing is steadily increasing, since nowadays the advances in computational capacities make it possible to implement sophisticated nonlinear processing techniques which in turn allow remarkable improvements with respect to standard and well-consolidated linear processing approaches. The aim of the book is to present a review of emerging new areas of interest involving nonlinear signal and image processing theories, techniques, and tools. More than 30 leading researchers have contributed to this book covering the major topics relevant to nonlinear signal processing. These topics include recent theoretical contributions in diﬀerent areas of digital ﬁltering and a number of applications in genomics, speech analysis and synthesis, communication system, active noise control, digital watermarking, feature extraction, texture analysis, and color image processing. The book is intended as a reference for recent advances and new applications of theories, techniques, and tools in the area of nonlinear signal processing. The target audience are graduate students and practitioners working on modern signal processing applications.

http://www.hindawi.com/books/spc/

Genetic and Evolutionary Computation for Image Processing and Analysis

Edited by: Stefano Cagnoni, Evelyne Lutton, and Gustavo Olague ISBN: 977-5945-22-4

Image analysis and processing is steadily gaining relevance within the large number of application fields to which genetic and evolutionary computation (GEC) techniques are applied. Although more and more examples of such applications can be found in literature, they are scattered, apart from a few exceptions, in proceedings and journals dedicated to more general topics. This book is the first attempt to offer a panoramic view on the field, by describing applications of most mainstream GEC techniques to a wide range of problems in image processing and analysis. More than 20 leading researchers in the field have contributed to this book, covering topics ranging from low-level image processing to high-level image analysis in advanced computer vision applications. Although the book is mainly application- oriented, particular care has been given to introducing GEC methods, in each chapter, at a level which makes them accessible to a wide audience. The expected target of the book comprises practitioners and researchers in image analysis and processing who may not be familiar with GEC techniques. At the same time, the book can as well be of interest for researchers in evolutionary computation, since most contributions focus on applications of genetic and evolutionary techniques which are based on nontrivial implementations of such methods. This feature reflects the nature of the contributions which are authored both by researchers for which GEC is the main field of interest and by researchers whose work is mainly focused on image processing and analysis.

http://www.hindawi.com/books/spc/

Advances in Signal Transforms: Theory and Applications

Edited by: J. Astola and L. Yaroslavsky ISBN: 977-5945-11-9 Digital signal transforms are of a fundamental value in digital signal and image processing. Their role is manifold. Transforms selected appropriately enable substantial compressing signals and images for storage and transmission. No signal recovery, image reconstruction and restoration task can be efficiently solved without using digital signal transforms. Transforms are successfully used for logic design and digital data encryption. Fast transforms are the main tools for acceleration of computations in digital signal and image processing. The volume collects in one book most recent developments in the theory and practice of the design and usage of transforms in digital signal and image processing. It emerged from the series of reports published by Tampere International Centre for Signal Processing, Tampere University of Technology. For the volume, all contributions are appropriately updated to represent the state of the art in the field and to cover the most recent developments in different aspects of the theory and applications of transforms. The book consists of two parts that represent two major directions in the field: development of new transforms and development of transform based signal and image processing algorithms. The first part contains four chapters devoted to recent advances in transforms for image compression and switching and logic design and to new fast transforms for digital holography and tomography. In the second part, advanced transform based signal and image algorithms are considered: signal and image local adaptive restoration methods and two complementing families of signal and image re-sampling algorithms, fast transform based discrete sinc- interpolation and spline theory based ones.

http://www.hindawi.com/books/spc/

Hindawi Publishing Corporation, 410 Park Avenue, 15th Floor, #287 pmb, INDAWI New York, NY 10022, USA; Fax 1-866-446-3294 (USA, toll-free) How to Become a EURASIP Member

EURASIP membership is open to all persons and institutions active or interested in signal processing within Europe or outside. Membership benefits include free subscription to EURASIP Newsletter; reduced subscription to the journals “Signal Processing,” “Speech Communication,” “Signal Processing: Image Communication,” “EURASIP Journal on Ap- plied Signal Processing,” and “EURASIP Journal on Wireless Communications and Net- working”; reduced fees for conferences and workshops that are sponsored or cosponsored by the Association; and reduced fees for EURASIP’s short courses. Personal (individual) membership must be paid by personal funds. To validate their category, students should provide an endorsement from school of ricials stating that they are enrolled in regular academic programs. Please send an application letter containing your name, title, position, company/institution, full mailing address, country, phone number, telefax number, and the desired journals to EURASIP, European Association for Signal, Speech, and Image Processing, EPFL-STI-LTS, CH-1015 Lausanne, Switzerland, together with your remittance. Your payments to EURASIP can be made in any of the following methods. In all cases, Euro is the only acceptable currency: 1. Using credit cards: MasterCard, American Express or Visa only. Please fill in your Card No., Expiring date, and signature. 2. Using cheques drawn in Euro mailed to EURASIP. 3. Using international money orders drawn in Euro mailed to EURASIP. 4. Requesting your bank to write transfer funds including your name to EURASIP at the Swiss Credit Bank Lausanne, Switzerland, account no. 322.294-31. 5. Using international postal money transfer to EURASIP postal account no. 10-3279-8 in Lausanne, Switzerland. For membership inquiries contact Fabienne Vionnet at sigpro@epfl.ch Please remember to mention your name and your EURASIP membership number on all payment forms.

2006 Print Only Journal Prices for EURASIP Members (Elsevier journals)

Institutional Individual Signal Processing €2461 €116 Speech Communication €952 €101 Signal Processing: Image Communication €792 €96

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Institutional Individual Signal Processing – €87 Speech Communication – €87 Signal Processing: Image Communication – €85

All EURASIP members that subscribe to an Elsevier journal have access to its electronic version by the Elsevier Website. How to Become a Member of EURASIP 175

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Institutional Individual EURASIP JASP €1285 €46 EURASIP JASMP Open Access Open Access EURASIP JBSB Open Access Open Access EURASIP JES Open Access Open Access EURASIP JWCN Open Access Open Access

2006 Print and Online Journal Prices for EURASIP Members (Hindawi Journals)

Institutional Individual EURASIP JASP €1540 €171 Subscribers to the online edition of any journal shall receive perpetual online access to the current volume as well as all back volumes of that journal even if they cancel their subscription in the future. EURASIP MEMBERSHIP APPLICATION

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