ISCApad number 107

May 6th, 2007

MESSAGE from Michael Wagner, member of the board, currently responsible for Industry Relations
Dear ISCA Member,
I am happy to report for ISCApad on my seven years of serving the speech science and technology community on the ISCA Board. When I joined the Board in 2000, I was drafted by then President Roger Moore into the discussions and preparations for the “internationalisation” of ESCA, the European Speech Communication Association, into ISCA as it is now. Subsequently, I was a member of the ISCA Committee that negotiated the merger of ISCA and PC-ICSLP, the then Permanent Council for the organisation of the International Conferences on Spoken Language Processing. I can only guess that the international diplomacy leading to treaties like nuclear non-proliferation were a cinch compared with the complexities of the merger between ISCA and PC-ICSLP, but eventually we achieved our common goal, and we now have the unified Interspeech Conferences (this year in Antwerp, next year in Brisbane!), which emerged from the preceding alternating Eurospeech and ICSLP Conferences. Subsequently, my portfolio on the Board shifted from Internationalisation to Industry Relations, which has included the enjoyable task of organising an “Industry Lunch” each year at the Interspeech Conference and initiating valuable contacts and discussions on strengthening the relationship between academic research, both fundamental and applied, and the research & development undertaken in the speech technology industry. ISCA is now facilitating a vigorous job market for speech science and technology graduates as well as for senior academics and engineers, partly at the Interspeech Conferences and partly through ISCApad. In addition, I stepped into the ISCA Treasurer portfolio when it became vacant in 2005. Initially, I found ISCA’s financial status a little difficult to understand, which was mainly due to the fact that a large proportion of ISCA’s assets are continually rolled over as loans to Interspeech Conferences and ISCA Workshops, only to be repaid one or two years later, a situation not well captured by ISCA’s previous cash accounting system. In 2006, the system was converted, with the invaluable help of ISCA’s Administrator Manu Foxonet, to an accrual accounting system, reflecting ISCA’s financial status more clearly, both for the Board and for you, the ISCA Members. I had to step down as Treasurer at the end of 2006 due to acute work overload – thank you to Christian Wellekens for stepping into the breach! – and, having served two full terms on the Board, I am now looking forward to my last Board meeting in Antwerp in August. After that? Interspeech 2008 in Brisbane is not much more than a year away and the Organising Committee of the Australasian Speech Science and Technology Association (ASSTA) is busily preparing to offer all of you yet another most enjoyable Interspeech Conference – see you in Antwerp in a few months and then again in Brisbane next year!
Michael Wagner
Professor, University of Canberra, Australia
Editorial

Dear Members,
Speech research and development are very active. A growing number of summer schools, master programs in speech processing,...is organized.
Have also a look of our updated list of job offers that shows how many speech experts are needed in our universities and industries.
Consider the major part of speech contributions in the last ICASSP.
Take a note of the coming ITRW worlwide organized and do not hesitate to inform our colleague Professor Sadaoki Furui about new ITRW you are volunteering to organize.
Great news is that ISCA board has now selected Chiba in Japan as the venue for Interpeech 2010: a long way to walk by the organizers but a very short time for preparing interesting results along the guidelines of this conference. But meanwhile we will meet in Antwerp (2007), Brisbane (2008) and in Brighton (2009). The Technical Program Committee is working hard to prepare the next Interspeech (Antwerp 2007).
Also I draw your attention on the extended deadline for applying for the Christian Benoit Prize.

Christian Wellekens

TABLE OF CONTENTS

  1. ISCA News
  2. SIG's activities
  3. Courses, internships
  4. Books, databases, softwares
  5. Job openings
  6. Journals
  7. Future Interspeech Conferences
  8. Future ISCA Tutorial and Research Workshops (ITRW)
  9. Forthcoming Events supported (but not organized) by ISCA
  10. Future Speech Science and technology events

ISCA NEWS


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Call for applications to the Christian Benoit Award
Extended deadline!
The Christian Benoit Award is delivered periodically by the Association Christian Benoit (**). It is given to promising young scientists in the domain of Speech Communication. The Award provides financial support for the development of a multi-media project promoting the work of these young scientists, and is valued at 7,622 Euros.
The first award was delivered to Tony Ezzat from MIT in June 2000, for his research in Audiovisual Speech Synthesis, the second award to Johanna Barry from University of Melbourne in September 2002 for her work on the acquisition of lexical tones in profoundly hearing-impaired speakers using a cochlear implant, and the third award to Olov Engwal from KTH in Stockholm in October 2004 for the elaboration of ARTUR, a multi-modal articulation tutor able to give automatic feedback to real users.
The fourth award will be delivered this year to ANY PROJECT IN THE FIELD OF SPEECH COMMUNICATION. Candidates should be in the final stages of their doctoral research or within the five years following the obtention of their PhD.
The Christian Benoit award will offer financial support to develop a multi-media project which (a) demonstrates the candidate's research in a way that helps launching that candidate's career, and (b) leverages electronic publishing technologies intelligently so as to facilitate the widest possible dissemination of this content.
In the application, the candidate should provide
-- a statement of research interest,
-- a detailed curriculum vitae, and
-- a description of the proposed multi-media project.
If the project already exists, a copy or link should be provided along with the application.
Applications should be sent to Pascal Perrier and received by Friday May 11th, 2007. Electronic submissions are mandatory.
The successful candidate will be notified by June 1st and invited to make a brief presentation of his/her work at the Interspeech 2007 Conference in Antwerp (Belgium).
Travel expenses for attendance at the Award ceremony will be provided by the Christian Benoit Association. For further information, please contact Pascal Perrier.
** The Christian Benoit Association is a nonprofit organization, whose purpose is to facilitate the development of research projects in the field of speech communication. Established in honor of Christian Benoit, French CNRS researcher in the field of speech communication who died on the 26th of April, 1998, at the age of 41, the Award places special emphasis on multimedia representations of ongoing research.

SIG's activities


A list of Speech Interest Groups can be found on our web.

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COURSES, INTERNSHIPS


ELSNET Summer School Belfast 2007

Advanced Dialogue Systems: Affectivity, Adaptability and Multimodality
16 - 27th July 2007
This year's ELSNET Summer School will be held in Belfast, Northern Ireland. It is being hosted by Queen's University at its pleasant tree-lined campus near Belfast's buzzing city centre. ELSNET Summer School Belfast is being organised by Computer Science at Queen's University in association with Computing and Mathematics at the University of Ulster.
The 2007 Summer School focuses on dialogue systems - covering everything from basic prompt and response systems to systems that adapt to the user's level of experience and even the user's emotional state. And the school includes guidance for assessing how well implemented systems are actually working.
Bringing together a teaching team of world experts, the school will cover industry-standard technologies, hosting environments and markup languages for building robust speech-based and multimodal dialogue systems. Alongside practical strands that teach attendees how to set about building both simple and more complex dialogue systems the school includes extensive coverage of the latest trends in dialogue development as viewed by academic and industrial dialogue specialists. At the leading edge of dialogue system development the school considers approaches to emotion-enablement - from analysis of real-world emotionally coloured interactions to ways of conveying affect through the use of computer-generated embodied conversational agents.
In addition to the 2-week schedule of lectures and practicals the summer school will be complemented by a social programme of events and recommended excursions.
Summer School Web Page.

Master in Human Language Technologies and Interfaces at the University of Trento

Website
organized by: University of Trento and Fondazione Bruno Kessler Irst
Call for applications, Academic Year 2007/08
Goal
Human language technology gives people the possibility of using speech and/or natural language to access a variety of automated services, such as airline reservation systems or voicemail, to access and communicate information across different languages, and to keep under control the increasing amount of information available by automatically extracting useful content and summarizing it. This master aims at providing skills in the basic theories, techniques, and applications of this technology through courses taught by internationally recognized researchers from the university, research centers and supporting industry partners. Students enrolled in the master will gain in depth knowledge from graduate courses and from substantial practical projects carried out in research and industry labs.
Courses:
Speech Processing, Machine Learning for NLP, Human Language, Text Processing, Spoken Dialog Systems,Human Computer Interaction, Language Resources, Multilingual Technology
Requisites
Master degree level ( min 4 years) in the area of computer science, electrical engineering, computational linguistics and cognitive science and other related disciplines. English language (official language)
Student Grants
A limited number of fellowships will be available.
Application Deadline
Non EU Students: June, 15
EU Students: end of July
Info
E-mail
University of Trento-Department of Information and Communication
Technologies Via Sommarive, 14-38100 Povo (Trento), Italy

Summer school: Cognitive and physical models of speech production, perception, and perception-production interaction.

Part II : Brain and Speech
Autrans, France
September 16-21, 2007
After the success of the previous summer school held in Lubmin (Germany) 2004, we are happy to announce the second international summer school on Cognitive and physical models of speech production, perception, and perception-production interaction. This year we will pay special attention to the brain. The aim of this summer school is to relate fundamental knowledge on speech production and perception to insights about the organization and function of the brain. Tutorials will be presented by specialists in these domains.
This summer school is intended mainly for graduate students, postdoctoral fellows, and researchers who work in the fields of speech production, perception, perception-production interaction, and the brain (neurolinguistics). Potential topics are:
Speech and language acquisition
Speech and language disorders
Neural basis of speech production
Speech production control
Neural basis of speech perception
Audio-visual speech perception
Plasticity of speech perception
It is intended to provide a platform for interchanges between students, junior and senior researchers, and hence, we would like each participant to feel free to contribute to any of these topics.
Submission
For abstract submission, please include the name(s) of the author(s), affiliations, and a contact e-mail address in the first lines of the body of the message. Texts should be written in English. Since the number of participants is limited to 40, registration will be restricted and based on the scientific quality of the submitted abstract. Authors are invited to present their work in discussion groups or poster sessions at the summer school.
All details can be viewed at the summer school website
Important dates
Deadline for the application is the 2nd of May, 2007!
Notification of acceptance May 21st, 2007
Summer school September 16th-21st,2007
Registration
The number of participants is limited to 40.
There will be no registration fee. Participants will have to pay for lodging and board. We are currently trying to get further funding for participants.
Invited speakers are:
Monica Baciu (LPNC, UPMF, Grenoble)
Grzegorz Dogil (Stuttgart university)
Hélène Loevenbruck (ICP/Gipsa-lab, CNRS, Grenoble)
Marc Sato (CRLMB, McGill university, Montréal)
Jean-Luc Schwartz (ICP/Gipsa-lab, CNRS, Grenoble)
Christophe Pallier (INSERM U562, Gif sur Yvette)
Georg Meyer (School of psychology, university of Liverpool)
Bernd Kröger (UK Aachen)
Organizers
Susanne Fuchs (ZAS, Berlin)
Hélène Loevenbruck (ICP, GIPSA-lab, Grenoble)
Daniel Pape (ZAS, Berlin)
Pascal Perrier (ICP, GIPSA-lab, Grenoble)

Studentships available for 2006/7 at the Department of Computer Science
The University of Sheffield - UK

One-Year MSc in HUMAN LANGUAGE TECHNOLOGY
The Sheffield MSc in Human Language Technology has been carefully tailored to meet the demand for graduates with the highly-specialised multi-disciplinary skills that are required in HLT, both as practitioners in the development of HLT applications and as researchers into the advanced capabilities required for next-generation HLT systems. The course provides a balanced programme of instruction across a range of relevant disciplines including speech technology, natural language processing and dialogue systems.
The programme is taught in a research-led environment. This means that you will study the most advanced theories and techniques in the field, and also have the opportunity to use state- of-the-art software tools. You will also have opportunities to engage in research-level activity through in-depth exploration of chosen topics and through your dissertation.
Graduates from this course are highly valued in industry, commerce and academia. The programme is also an excellent introduction to the substantial research opportunities for doctoral-level study in HLT.
A number of studentships are available, on a competitive basis, to suitably qualified applicants. These awards pay a stipend in addition to the course fees.
See further details of the course
Information on how to apply

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BOOKS, DATABASES, SOFTWARES

Databases

HIWIRE database
We would like to draw your attention to the Interspeech 2007 special session "Novel techniques for the NATO non-native Air Traffic Control and HIWIRE cockpit databases"
http://www.interspeech2007.org/Technical/nato_atc.php
that we are co-organizing. For this special session we make available (free of charge) the cockpit database, along with training and testing HTK scripts. Our goal is to investigate feature extraction, acoustic modelling and adaptation algorithms for the problem of (hands-free) speech recognition in the cockpit. A description of the task, database and ordering information can be found at the website of the project We hope that you will be able to participate to this special session.
Alex Potamianos, TUC
Thibaut Ehrette, Thales Research
Dominique Fohr, LORIA
Petros Maragos, NTUA
Marco Matassoni, ITC-IRST
Jose Segura, UGR

- Language Resources Catalogue - Update
ELRA is happy to announce that 3 new Speech Related Resources are now available in its catalogue. Moreover, we are pleased to announce that years 2005 and 2006 from the Text Corpus of "Le Monde" (ELRA-W0015) are now available.
*ELRA-S0235 LC-STAR Hebrew (Israel) phonetic lexicon *The LC-STAR Hebrew (Israel) phonetic lexicon comprises 109,580 words, including a set of 62,431 common words, a set of 47,149 proper names (including person names, family names, cities, streets, companies and brand names) and a list of 8,677 special application words. The lexicon is provided in XML format and includes phonetic transcriptions in SAMPA. More information
*ELRA-S0236 LC-STAR English-Hebrew (Israel) Bilingual Aligned Phrasal lexicon *The LC-STAR English-Hebrew (Israel) Bilingual Aligned Phrasal lexicon comprises 10,520 phrases from the tourist domain. It is based on a list of short sentences obtained by translation from US-English 10,449 phrasal corpus. The lexicon is provided in XML format. More information
*ELRA-S0237 LC-STAR US English phonetic lexicon *The LC-STAR US English phonetic lexicon comprises 102,310 words, including a set of 51,119 common words, a set of 51,111 proper names (including person names, family names, cities, streets, companies and brand names) and a list of 6,807 special application words. The lexicon is provided in XML format and includes phonetic transcriptions in SAMPA. More information
*ELRA-W0015 Text corpus of "Le Monde" *Corpus from "Le Monde" newspaper. Years 1987 to 2002 are available in an ASCII text format. Years 2003 to 2006 are available in .XML format. Each month consists of some 10 MB of data (circa 120 MB per year). More information
For more information on the catalogue, please contact Valérie Mapelli
Our on-line catalogue has moved to the following address. Please update your bookmarks.

Books

Human Communication Disorders/ Speech therapy
This interesting series can be listed on Wiley website

Incurses em torno do ritmo da fala
Author: Plinio A. Barbosa
Publisher: Pontes Editores (city: Campinas)
Year: 2006 (released 11/24/2006)
(In Portuguese, abstract attached.) Website

Speech Quality of VoIP: Assessment and Prediction
Author: Alexander Raake
Publisher: John Wiley & Sons, UK-Chichester, September 2006
Website

Self-Organization in the Evolution of Speech, Studies in the Evolution of Language
Author: Pierre-Yves Oudeyer
Publisher:Oxford University Press
Website

Speech Recognition Over Digital Channels
Authors: Antonio M. Peinado and Jose C. Segura
Publisher: Wiley, July 2006
Website

Multilingual Speech Processing
Editors: Tanja Schultz and Katrin Kirchhoff ,
Elsevier Academic Press, April 2006
Website

Reconnaissance automatique de la parole: Du signal a l'interpretation
Authors: Jean-Paul Haton
Christophe Cerisara
Dominique Fohr
Yves Laprie
Kamel Smaili
392 Pages
Publisher: Dunod

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JOB OPENINGS

We invite all laboratories and industrial companies which have job offers to send them to the ISCApad editor: they will appear in the newsletter and on our website for free. (also have a look at http://www.isca-speech.org/jobs.html as well as http://www.elsnet.org/ Jobs)

Position for a PhD-student in Nijmegen, The Netherlands

The European project "Acoustic reduction in European languages" investigates how speakers and listeners process acoustically reduced words, such as the pronunciation "yesay" for "yesterday" and "onry" for ordinary", in six European languages (Dutch, English, Estonian, Finnish, French, and Spanish). Essential to this research program are corpora of highly spontaneous speech, which exist for English and Dutch, but which will have to be compiled for Estonian, Finnish, French, and Spanish in the course of this project. Complementary to corpus based research, the processing of acoustic reduction will be addressed by means of series of psycholinguistic experiments.
The project has as its principal investigator Dr M. Ernestus. It is funded by a European Young Investigator award, as well as by the Max Planck Institute for Psycholinguistics and by the Radboud University. The research group is located in the building of the Max Planck Institute, on the campus of the Radboud University, in Nijmegen, The Netherlands. This location guarantees a stimulating research environment with excellent experimental facilities. It offers researchers the possibility to develop interdisciplinary skills and to discuss their work with many internationally renowned scholars.
The project is now offering a position for a PhD-student who will investigate acoustic reduction in French and Spanish. The PhD-student will explore which types of reduction occur in these languages, and how the production and comprehension of reduced words are affected by the morphological and phonological properties of these languages. At a more general level, the PhD student will compare synchronic reduction in modern Spanish with the diachronic reduction that has occurred in French.
The PhD student will collaborate closely with the principal investigator. In addition, the PhD-student will be supported in his/her research (including the compilation of the speech corpora) by a team of research assistants.
Applicants should be (near-)native in French and Spanish and also be fluent in English. They should have a master's degree in linguistics or phonetics, or receive one within a few months. Moreover, applicants should have a basic knowledge of the phonology, phonetics, and morphology of French and Spanish. The successful candidate will receive a contract for three and a half years at the Radboud University Nijmegen (www.ru.nl), under the conditions for PhD-students at this university.
For further information, including a description of the complete project, please contact Prof.Mirjam Ernestus (phone: +31-24-3612970).
Application letters, including extensive CVs, should arrive at the latest on 31 May 2007, addressed to Mirjam Ernestus
P.O. Box 310
NL-6500 AH Nijmegen
The Netherlands
or emailed.

Research Positions at Stanford: Robust Dialogue Understanding
The Center for the Study of Language and Information (CSLI) at Stanford University is seeking Research Scientists to work on multimodal spoken-language dialogue systems, starting approximately 1 June 2007.
The ideal candidate is a Computational Linguist with an interest in the computational semantics and pragmatics of dialogue, with experience in formally-inspired and/or statistical/machine-learning approaches to dialogue modeling. The position requires a Ph.D. in computational linguistics, natural language processing, artificial intelligence, cognitive science, or a related field. Applicants should have a demonstrated capacity to define and implement a research plan, and to conduct individual and collaborative research consonant with the dialogue-systems projects underway at CSLI.
Current research into dialogue at CSLI includes both human-human and human-computer dialogue modeling, and employs a variety of techniques, including symbolic and stochastic, theory and data-driven. Proficiency in multiple approaches relevant to current CSLI application areas will be highly valued, as will an ability to participate in implementation.
Research topics of particular interest include:
- robust semantic interpretation from noisy data (e.g. fragment parsing, role detection);
- robust context-based pragmatic interpretation (e.g. anaphora/ ellipsis/fragment resolution);
- multi-party discourse modeling (esp. group decision-making);
- topic/issue detection and tracking;
- probabilistic dialogue state/activity modeling and tracking.
The successful candidate will work with faculty, postdoctoral, and student researchers in the Dialogue Systems Group at CSLI, performing novel research and developing core infrastructure for natural multimodal conversational systems for a range of interactions and applications. Responsibilities will include supervising student research assistants and participating in proposal-preparation to attract new funding.
The CSLI Dialogue Systems Group consists of about thirteen people, and is involved in a number of projects involving close collaboration with other Stanford departments, numerous other academic institutions, government agencies such as NASA, not-for-profit research organizations such as SRI, and various commercial enterprises. Current projects include: understanding multi-party conversational interaction; collaborative control of teams of robotic devices; conversational control of in-vehicle devices; and speech- enabled intelligent tutoring systems.
The position is initially for 1 year, renewable for up to 3 years (contingent on continued funding). Salary is dependent on qualifications and experience, but is expected to be in a range starting from $70,000 for a junior appointee, to $90,000 for a senior appointee. Candidates who may be more appropriate for a more senior appointment are encouraged to contact us directly and may be considered at a higher salary.
Applicants should submit a letter of application and a full resume or curriculum vitae with names and email addresses of at least three references. Please contact Stanley Peters and Matt Purver for further information.
Stanford University is an equal opportunity, affirmative action employer.

Graduated engineer at IRISA (INRIA Rennes)
Context
Irisa (INRIA Rennes) seeks to recruit a recently graduated R&D engineer in spoken document processing to package, develop and improve its rich transcription platform.
The Metiss team at Irisa has developed a recognized know-how in the field of audio processing with emphasis on speaker and speech recognition. Over the last few years, we have developed a spoken document rich transcription platform, Irene, in collaboration with ENST Paris. This platform, mostly built on top of speech processing softwares developed at Irisa (SPro, AudioSeg, Sirocco), is at the centre of our research activities enabling the validation and experimentation of new ideas in a complete application setup. Research and development activities on the Irene platform are in part related to the development of a video content server, Telemex. The latter acts as a key element of our research activities in multimedia document analysis and indexing, carried out at Irisa in collaboration with our industrial partners.
The current version of the Irene platform is an experimental version based on proprietary softwares completed with public ones (HTK, SRI LM). These softwares provide basic functionalities which are linked via a set of scripts. Dedicated to research activities, the current platform does not provide an integrated and easy-to-use solution, thus limiting its distribution as well as its use within a complete application setup.
Job description
The R&D engineer will be in charge of the integration and the development of our rich transcription platform, with the following missions:
- define and implement an integrated architecture to enable the distribution of a verstatile spoken document rich transcription platform;
- implement new functionalities in our proprietary toolkits to enable a fully proprietary platform, in particular concerning our speech recognition software Sirocco;
- integrate and validate the latest technology to improve our rich transcription platform in the framework of the Telemex video server for TV streams transcription.
The work will be carried out in close relation with our academic and industrial partners who have expressed an interest for a spoken document transcription platform.
Skills required
Prospective candidates should have a strong theoretical and practical background in computer science applied to Communication and Information Technology. In particular, strong knowledge in at least one of the following domain is necessary: pattern recognition, graph theory, dynamic programing, statistical modeling (HMM). Fluency in C language and in Perl in a Unix/Linux environment is also required. Knowledge of the French language is welcome but not required.
Practical information
The appointment is for a one year period, renewable for an additional one year, starting fall 2007. The engineer will work primarily in the Metiss team at Irisa, Rennes (France). Net salary is 2020 EUR per month, including social security benefits.
As this job opportunity is reserved to recently graduated students, candidates should hold a Master degree for less than two years.
Interested candidates should contact Guillaume Gravier by mail (guillaume.gravier@irisa.fr) for further information. Application deadline is May 23, 2007.
Related links:
Joboffers
Project SPRO
Project Audioseg
Project Sirocco

Facial and Vocal Expression: Two Postdoctoral Fellowships available in Geneva

The Swiss Center for the Affective Sciences and the Geneva Emotion Research Group at the University of Geneva invite applications for two postdoctoral fellowships in the areas of facial and vocal emotion expression. The Swiss Center for Affective Sciences (www.affective-sciences.org) is an interdisciplinary research institute affiliated with the University of Geneva. Its mandate is to study all aspects of human emotion, including psychological and physiological determinants of emotional communication. Together with the Geneva Emotion Research Group, the Center has developed a large corpus of emotional portrayals called GEMEP (GEneva Multimodal Emotion Portrayals). The analysis of the corpus and its subsequent use in experimentation and clinical testing will result in a better understanding of the processes involved in the communication of emotion.
As part of the ongoing work on the GEMEP corpus and the general research program in the Center, the two postdoctoral researchers will participate in either 1) the development of novel methods to digitally analyze the pertinent acoustic parameters in the vocal channel or 2) research on facial expression patterns on the basis of the Component Process Model and collaborative efforts to automatically extract and synthesize facial features.
Position 1) Physiology of voice and speech production, Acoustic measurement, Phonetics.
This fellowship requires a strong interest and research competence in phonetics and acoustics as well as solid knowledge in the physiological bases of voice and speech. As this research is hypothesis driven, candidates with a solid background of linking voice production processes to the measurement of the associated acoustic parameters will be given preference. Position 2) Facial expression, Emotion theory, Nonverbal behavior, FACS.
This fellowship requires a strong interest and research competence in facial expression and nonverbal behavior as well as solid knowledge in the theory of emotions. The candidate must be a certified FACS coder having used FACS in his/her doctoral thesis. Candidates familiar with the timing aspect of the facial expression will be given preference. As this research could include collaboration with other institutions about testing and development of automatic features-extraction software, experience in using different coding and analysis software would be desirable.
The salary corresponds to the position of a post-doctoral researcher as fixed by the University of Geneva (CHF 64’000-72’000 per year. depending on experience). This is a full time position, to be filled as soon as possible.
Please submit your application electronically, before May 31, 2007, including Curriculum Vitae, a letter of intention and supporting documentation to Sylvie Staehli

Maitre de Conference en Reconnaissance et Comprehension de la Parole -Universite Rene Descartes Paris 5
Un poste de MCF (27 MCF 1616) en Reconnaissance et compréhension de la parole est à pourvoir à l'université René Descartes-Paris 5 (UFR de Mathématiques et Informatique) avec le profil suivant en recherche et en enseignement :
RECHERCHE
Le CRIP5 est un laboratoire d'informatique, avec des axes de recherche spécifiques et une production de niveau international. C’est aussi un laboratoire de recherche appliquée, résolument orienté vers les domaines qui font l’originalité de l’université Paris 5 (sciences de la vie et sciences humaines). L’équipe Dialogue et Indexation (Diadex) s’intéresse à tous les domaines de recherche de la reconnaissance et de la compréhension de la parole (modèles acoustiques et linguistiques, stratégies de décodage et optimisation, modèles de langage de genre, planification du dialogue, grammaires formelles pour le langage naturel, …). Le nouveau maître de conférences devra s’intégrer dans l’équipe Diadex et avoir une expérience solide dans un ou plusieurs des domaines précédemment cités. Il devra s’impliquer dans les différents projets de l’équipe et participer à l’encadrement des étudiants du master recherche.
ENSEIGNEMENT
Le nouveau Maître de Conférences prendra en main les enseignements de programmation et participera activement à leur organisation tout au long des trois années de la Licence Mathématiques Informatique et Applications. Il participera également au développement du parcours « Parole et Communication Homme-Machine » du Master Recherche MISV, spécialité Informatique
Contact : Prof. Marie-Jose Caraty
Website

Opening on Speech recognition at Telefonica, Barcelona (Spain)

The speech Technology Group at Telefonica Investigacion y Desarrollo (TID) is looking for a highly qualified candidate for an engineering position on speech recognition and related technologies.
The selected person will become a part of a multidisciplinary team of young highly motivated people in an objective driven, friendly atmosphere located in a central area of Barcelona (Spain).
Minimum requirements ar:
Degree in Computer Science /Electrical Engineering/Computational Linguistics or similar with 2+ years of experience (Ph.D. preferred) on speech technology.
Good knowledge of speech recognition and speech synthesis.
Proven programming expertise in C++ and Java
Good level of English (required) and some knowledge of Spanish (preferred)
High motivation and teamwork spirit
Salary depending on the experience and value of the applicant
Starting date as soon as possible
The speech technology group is a well established group within TID with more than 15 years of experience in research and development of technology for internal use of Telefonica group as well as outside organizations. It is also a very active partner in many National and European projects. TID is the research and development company inside the Telefonica group, currently one of the biggest Telecom companies. It is the biggest private research center in Spain in number of employees and available resources.
Please send your resume and contact information to
Sonia Tejero
Tlf: +34 93 365 3024

Sound to Sense: 18 Fellowships in speech research

Sound to Sense (S2S) is a Marie Curie Research Training Network involving collaborative speech research amongst 13 universities in 10 countries. 18 Training Fellowships are available, of which 12 are predoctoral and 6 postdoctoral (or equivalent experience). Most but not all are planned to start in September or October 2007.
A research training network’s primary aim is to support and train young researchers in professional and inter-disciplinary scientific skills that will equip them for careers in research. S2S’s scientific focus is on cross-disciplinary methods for modelling speech recognition by humans and machines. Distinctive aspects of our approach include emphasis on richly-informed phonetic models that emphasize communicative function of utterances, multilingual databases, multiple time domain analyses, hybrid episodic-abstract computational models, and applications and testing in adverse listening conditions and foreign language learning.
Eleven projects are planned. Each can be flexibly tailored to match the Fellows’ backgrounds, research interests, and professional development needs, and will fall into one of four broad themes.
1: Multilinguistic and comparative research on Fine Phonetic Detail (4 projects)
2: Imperfect knowledge/imperfect signal (2 projects)
3: Beyond short units of speech (2 projects)
4: Exemplars and abstraction (3 projects)
The institutions and senior scientists involved with S2S are as follows:
* University of Cambridge, UK (S. Hawkins (Coordinator), M. Ford, M. Miozzo, D. Norris. B. Post)
* Katholieke Universiteit, Leuven, Belgium (D. Van Compernolle, H. Van Hamme, K. Demuynck)
* Charles University, Prague, Czech Republic (Z. Palková, T. Dub?da, J. Volín)
* University of Provence, Aix-en-Provence, France (N. Nguyen, M. d’Imperio, C. Meunier)
* University Federico II, Naples, Italy (F. Cutugno, A. Corazza)
* Radboud University, Nijmegen, The Netherlands (L. ten Bosch, H. Baayen, M. Ernestus, C. Gussenhoven, H. Strik)
* Norwegian University of Science and Technology (NTNU), Trondheim, Norway (W. van Dommelen, M. Johnsen, J. Koreman, T. Svendsen)
* Technical University of Cluj-Napoca, Romania (M. Giurgiu)
* University of the Basque Country, Vitoria, Spain (M-L. Garcia Lecumberri, J. Cenoz)
* University of Geneva, Switzerland (U. Frauenfelder)
* University of Bristol, UK (S. Mattys, J. Bowers)
* University of Sheffield, UK (M. Cooke, J. Barker, G. Brown, S. Howard, R. Moore, B. Wells)
* University of York, UK. (R. Ogden, G. Gaskell, J. Local)
Successful applicants will normally have a degree in psychology, computer science, engineering, linguistics, phonetics, or related disciplines, and want to acquire expertise in one or more of the others.
Positions are open until filled, although applications before 1 May 2007 are recommended for starting in October 2007.
Further details are available from the web about: + the research network and how to apply: http://www.ling.cam.ac.uk/s2s/s2sJobAd.pdf (92 kB) + the research projects: http://www.ling.cam.ac.uk/s2s/s2sProjects.pdf (328 kB).
Post-doc in Japan - Machine learning, kernel machines, computational statistics, Bayesian statistics, or multimodal processing

DeadLine: 01/05/2007
Website
The Institute of Statistical Mathematics (ISM)
Research Organization of Information and Systems (ROIS)
Postdoctoral position: Applicants are invited to apply to the Transdisciplinary Research Integration Center, ISM/ROIS. ISM is one of member institutes of ROIS along with the National Institute of Informatics, the National Institute of Genetics and the National Institute of Polar Research. The ISM mission includes promoting statistical science and developing an innovative methodology for approaching complex problems related to life science, earth science, environmental science and human sciences from the view point of information and systems (http://www.ism.ac.jp/index_e.html). The position will start as soon as possible after 1 April 2007. The postdoctoral researcher will work on the following project “Discovery of Invariants in Multimodal Data” (http://www.ism.ac.jp/~tmatsui/kinou2_p4/index-en.html). The initial contract is one-year long but could be extended up to three years.
Field of work: Machine learning, kernel machines, computational statistics, Bayesian statistics, or multimodal processing
Project description: Multimodal data available to us through the Internet and other electronic media are explosively increasing both in number and in variety. To handle such massive data for various purposes, new technologies need to be developed. With this in mind, we have started investigating a new methodology that allows us to discover from multimodal data the information relevant to the purpose at hand (which is referred to as “invariants”). To achieve this goal, we will study several qualitatively different problems from different research areas, in which multimodal data play a central role (e.g., visual/audio/text processing, cognitive science, auditory perception and robotics). The problems are to be tackled with some of the recently developed inductive learning machines including automatic model selection mechanism (e.g., Penalized Logistic Regression Machines and Support Vector Machines). The results will be analyzed in order to establish a new methodology for discovery of invarian! ts, which will be applicable to problems across different areas of study.
Job description: The successful candidate will support and coordinate our efforts in the area of investigation of methods for discovery of invariants with multimodal data.
Requirements: Applicants should have a PhD and some knowledge of machine learning and statistics. Applicants must be able to program (C/C++ and Matlab knowledge is an advantage but not requirement) and must also have experience with statistical data analysis.
Payment: The salary will be in the range of 4,500,000 yen 6,000,000 yen (before tax and insurance).
Application: Applicants should send their CV, including a list of publications and the names of two potential referees.
Contact: Prof. Tomoko Matsui
Tel: +1 604 822 9662 (until 9 March 2007 (Vancouver, Canada)
+81 3 5421 8769 (from 10 March 2007 (Japan)

Postdoctoral position- INRIA-LORIA Nancy France

Objective
Despite recent progresses achieved in speech synthesis, it is still very difficult to modify the characteristics linked to the speaker since signals are synthesized by concatenating sounds uttered by a given speaker. It is thus almost impossible to modify acoustic cues of sounds as well as characteristics linked to the speaker.
The objective of the postdoc is to elaborate copy synthesis algorithms that enable a speech signal to be reproduced as faithfully as possible while offering the possibility of modifying acoustic cues. For this reason this postdoctoral work will rely on the utilization of a formant synthesizer derived from that proposed by Klatt[1]. Synthesis thus rests on the filtering by a system of resonators (representing formants) of a sound source, periodic for the voiced sounds as vowels, aperiodic (a noise) for unvoiced sounds as fricatives phonemes.
Work
The work will consist of adapting the synthesizer so that it does lend itself to copy synthesis as well as possible and to develop algorithms to optimizing source and formant parameters.
In order to copy speech sufficiently finely it is necessary to adjust formant and source parameters precisely. The LF source model proposed by Fant and Liljencrants [2] is sufficiently versatile to approximate a natural speech source. The optimization of the four parameters was the subject of a number of works in the case where the vocal tract filter and source are estimated jointly [3,4] or when the source signal is known [5]. The specificity of copy synthesis is that the filter of the vocal tract is only roughly approximated by formants hypothesised and that the ratio of noise in the source has also to be adjusted for each of the formants.
Resonators of a formant synthesizer can be organized in cascade or in parallel. Only the second solution is usable in the case of copy synthesis because it enables formants to be adjusted independently [6]. The frequency, amplitude and bandwidth of each formant have to be specified. One important advantage of the parallel architecture is that it is possible to adjust only amplitude by setting the bandwidth to a default value once the formant frequency is known. The second aspect of the work will be on the elaboration of an algorithm to adjust amplitudes and frequencies. The adjustment of amplitudes must be synchronized on source periods in order to capture fast variations of amplitude, and that of formant frequencies will rest upon the automatic formant tracking previously developed [7]. Improvements will be about the choice of the formant number so as to increase the closeness of the speech copied with respect to the original signal.
The two aspects have been presented independently to simplify the presentation of the work. To a certain extent only they also can be addressed independently. However, it is clear that the improvement of the synthesis quality will be all the better since interactions between these two aspects will have been considered together.
The Parole team mainly works on automatic speech recognition and speech analysis. In the domain of analysis a number of algorithms have been developed (F0 detection, formant tracking, pitch marking, copy synthesis...) and are available in WinSnoori software which already contains a series of tools for copy synthesis and which is developed by the team for several years.
Skill and profile
A good knowledge in speech analysis or in signal processing is required.
References
Copy synthesis tools of WinSnoori are presented here.
[1] D.H. Klatt, “Software for a cascade/parallel formant synthesizer”, J. Acoust. Soc. Amer., 67(3), p. 971-995, March 1980.
[2] G. Fant and J. Liljencrants, “A four parameter model of glottal flow”, STL, QPSR, 4, p. 1-13, 1985
[3] M. Frölich , D. Michaelis and H.W. Strube, “SIM-simultaneous inverse filtering and matching of a glottal flow model for acoustic speech signals”, J. Acoust. Soc. Amer., 115(1), p.337-351, 2003.
[4] D. Vincent, O. Rosec and T. Chonavel, “Estimation of LF glottal source parameters based on an ARX model”, Proc. of Interspeech, p. 333-336, Lisboa, Sep. 2005.
[5] J. Pérez and A. Bonafonte, “Automatic Voice-Source Parametrization of Natural Speech”, Proc. of Interspeech, Lisboa, Sep. 2005.
[6] W. J. Holmes, “Copy synthesis of female speech using the JSRU parallel formant synthesiser”, Proceedings of European Conference on Speech Technology, p. 513-516, Paris, France, Sep., 1989
[7] Y. Laprie, “A concurrent curve strategy for formant tracking”, Proc. of ICSLP, Jegu, Korea, Oct. 2004
Contact
Interested candidates are invited to contact Yves Laprie
Important information
This position is advertised in the framework of the national INRIA campaign for recruiting post-docs. It is a one year position, renewable, beginning fall 2007. The salary is 2,320€ gross per month.
Selection of candidates will be a two step process. A first selection for a candidate will be carried out internally by the PAROLE group. The selected candidate application will then be further processed for approval and funding by an INRIA committee.
Doctoral thesis less than one year old (May 2006) or being defended before end of 2007. If defence has not taken place yet, candidates must specify the tentative date and jury for the defence.
Useful link
Presentation of INRIA postdoctoral positions To apply(be patient, loading this link takes times…)

Research scientist- Speech Technology- Princeton, NJ, USA

Company Profile: Headquartered in Princeton, NJ, ETS (Educational Testing Service)is the world's premier educational measurement institution and a leader in educational research. As an innovator in developing achievement and occupational tests for clients in business, education, and government, we are determined to advance educational excellence for the communities we serve.
Job Description: ETS Research & Development has a Research Scientist opening in the Automated Scoring and Natural Language Processing Group. This group conducts research focusing on the development of new capabilities in automated scoring and NLP-based analysis and evaluation systems, which are used to improve assessments, learning tools and test development practices for diverse groups of users that include K-12 students, college students, English Language Learners and lifelong learners. The Research Scientist position involves applying scientific, technical and software engineering skills to designing and conducting research studies and developing capabilities in support of educational products and services. The job is a full-time job.
Required qualifications
· A Ph.D. in Natural Language Processing, Computational Linguistics, Computer Science, or Electrical Engineering with a focus on speech technology, particularly speech recognition. Knowledge of linguistics is a plus.
· Evidence of at least three years of independent substantive research experience and/or experience in developing and deploying speech technology capabilities, preferably in educational environments.
· Demonstrable contributions to new and/or modified theories of speech processing and their implementation in automated systems.
· Practical expertise with speech recognition systems and fluency in at least one major programming language (e.g., Java, Perl, C/C++, Python).
· Three years of independent substantive research experience and/or experience in developing and deploying speech technology capabilities, preferably in educational environments.
How to apply
Please send copy of your resume, along with cover letter stating salary requirements and job #2965, to e-mail
ETS offers competitive salaries, outstanding benefits, a stimulating work environment, and attractive growth potential. ETS is an Equal Opportunity, Affirmative Action Employer.
Web site

Research Fellow in Speech Synthesis- Centre for Speech Technology Research/ University of Edinburgh

The Centre for Speech Technology Research at the University of Edinburgh is seeking a research fellow to work on the speech synthesis project "Automatically-determined inventories for speech synthesis". This project uses machine learning techniques to automatically discover, from speech data, a set of units for speech synthesis - that is, an alternative to manually-specified phoneme-based units such as diphones. This research is currently being conducted within a concatenative (i.e. unit selection) framework, but we now seek to extend this to the other major synthesis technique: statistical parametric synthesis, based on Hidden Markov Models (i.e., trajectory HMMs). The successful candidate will be expected to contribute, plan and execute new research, as well as extend our existing techniques. You ideally will have a PhD in speech synthesis and experience of trajectory Hidden Markov Models. You will have very good programming skills, preferably in C++, and experience with one or more of: concatenative speech synthesis techniques; statistical models of speech; perceptual evaluations; Festival. An automatic speech recognition background is also appropriate for this position. This post is fixed term for 15 months.
For more information and application instructions, consult our website and enter vacancy number 3006866.

Software Engineer Position at Be Vocal, Mountain View, CA,USA

We are currently looking for a Software Engineer with previous exposure to Speech, to work in our Speech and Natural Language Technology group. This group’s mission is to be the center of excellence for speech and natural language technologies within BeVocal. Responsibilities include assisting in the development of internal tools and processes for building Natural Language based speech applications as well as on ongoing infrastructure/product improvements. The successful candidate must be able to take direction from senior members of the team and will also be given the opportunity to make original contributions to new and existing technologies during the application development process. As such, you must be highly motivated and have the ability to work well independently in addition to working as a team.
Responsibilities
* Develop and maintain speech recognition/NLP tools and supporting infrastructure
* Develop and enhance component speech grammars
* Work on innovative solutions to improve overall Speech/NL performance across BeVocal’s deployments.
Requirements
* BS in Computer Science, Electrical Engineering or Linguistics, an MS is a preferred.
* 2-5 years of software development experience in Perl, Java, C/C++. A willingness and ability to pick up additional software languages as needed is essential.
* Exposure or experience with speech recognition/pattern recognition either from an academic environment or directly related work experience.
* Experience working as part of a world-class speech and language group is highly desirable.
* Experience building natural language applications is preferred.
* Experience building LVCSR speech recognition systems is a plus.
For immediate consideration, please send your resume by email and include "Software Engineer, Speech" in the subject line of your email. Principals only please (no 3rd parties or agencies). Contact for details
BeVocal's policy is to comply with all applicable laws and to provide equal employment opportunity for all applicants and employees without regard to non-job-related factors such as race, color, religion, sex, national origin, ancestry, age, disability, veteran status, marital status or sexual orientation. This policy applies to all areas of employment, including recruitment, hiring, training, promotion, compensation, benefits, transfer, and social and recreational programs.

Postdoctoral Fellow -- Speech Synthesis- Alfred I. Dupont Hospital for Children, Wilmington, DE

The Alfred I. duPont Hospital for Children in Wilmington, DE has an immediate opening for a Postdoctoral Fellow in Speech Synthesis in the Speech Research Laboratory, within the Department of Biomedical Research. The ideal candidate will have a Ph.D. in Computer Science, Linguistics, Psychology, or a related field, demonstrated experience in data-based speech synthesis techniques, and an interest in modeling prosody, particularly intonation, in speech synthesis systems. The primary responsibilities for this position include: Developing a model for intonation that can be trained on and capture the important talker-specific features of an individual's speech while also representing phonologically motivated f0 characteristics; implementing the intonation model for the ModelTalker TTS system; and assisting in the creation of unit concatenation voices for the ModelTalker TTS system. A Ph.D. in Linguistics, Computer Science, Psychology, or closely related field with demonstrated knowledge of and experience in concatenative speech synthesis techniques, speech analysis techniques, and acoustic phonetics is required. Computer programming experience with C or C++, knowledge of additional languages is a plus. Experience with Unix/Linux and Windows operating systems is essential.
This is a two-year grant-funded position. For more information, email Dr Timothy Bunnel or call at (302) 651-6835. Applicants may also post their resume on-line at www.nemours.org or send resume with salary requirements to Dr. Timothy Bunnell, Department of Biomedical Research, Alfred I. duPont Hospital for Children, P.O. Box 269, Wilmington, DE 19899.

Position at Saybot in China

Job title: Speech Scientist
Location: China (Beijing or Shanghai)
Saybot develops software technology and curricula for learning spoken english. Since 2005, we have been building software which features state-of-the-art speech technologies and innovative interactive lessons to help users practice speaking English. We are currently looking for talented speech scientists to help strengthen our R&D team and to develop our next-generation products. Successful candidates would have proven excellence and good work ethics in academic or industry context and demonstrated creativity in building speech systems with revolutionary designs.
* MS/PhD degree in speech technology (or related).
* Expertise in at least one of the following areas and basic knowledge of the others:
o acoustic model training,
o speaker adaptation,
o natural language understanding,
o prosody analysis,
o embedded recognizers.
* Excellent programming skills in both object-oriented languages (C++, C# or Java) and scripting (Perl or Python).
* Good knowledge and experience in at least one commonly used recognizer (HTK, Sphinx, Nuance...).
* Excellent communication skills in written and oral English.
* Experience in machine translation is a plus.
* Experience in VoIP integration is a plus.
* Experience in language teaching is a plus.

Contact: Sylvain Chevalier

2 Positions in Research and Development in "Audio description and indexing" at IRCAM-Paris

PRESENTATION OF THE SAMPLE ORCHESTRATOR PROJECT:
The goal of the Sample Orchestrator project is to develop and test new applications for managing and manipulating sound samples based on audio content. On the one hand the commercial availability of large databases of sound samples available on various supports (CD, DVD, online), are currently limited in their applications (synthesizers by sampling). On the other hand, recent scientific and technological development in audio indexing and database management allow the development of new musical functions: database management based on audio content, audio processing driven by audio content, development of orchestration tools.
TASKS:
Two positions are available from April 15th 2007 within the "Equipe Analyse/Synthese" of Ircam for (each) a 12 months total duration (possibility of extending the contracts). The main tasks to be done for the research and development positions are:
- Research and development of new audio features and algorithms for the description of instrumental, percussive and FX sounds.
- Research and development of new audio features and algorithms for the morphological description of sounds
- Research and development of new audio features and algorithms for sounds containing "loops"
- Research and development of algorithms for automatic audio indexing
- Research and development of algorithms for fast search by similarity in large databases
- Participation in the definition of the specification
- Participation in user evaluation and feedback
- Integration into the final application
RESEARCH POSITION:
REQUIRED EXPERIENCE AND COMPETENCE:
- High skills in Audio indexing and signal processing
- High skills in Matlab programming
- High productivity, methodical work, excellent programming style.
- Good knowledge of UNIX, Mac and Windows environments
SALARY:
According to background and experience.
DEVELOPMENT POSITION:
REQUIRED EXPERIENCE AND COMPETENCE:
- Skills in Audio indexing and signal processing
- High skills in C/C++ programming
- High productivity, methodical work, excellent programming style.
- Good knowledge of UNIX, Mac and Windows environments
SALARY:
According to background and experience.
EEC WORKING PAPERS:
In order to start immediately, the candidate should preferably have EEC citizenship or already own valid EEC working papers.
AVAILIBILITY:
The positions are available in the "Analysis/Synthesis" team in the R&D department from April 15th 2007 for (each) a duration of 12 months (possibility of extending the contracts).
TO APPLY:
Please send your resume with qualifications and informations addressing the above issues, preferably by email Xavier Rodet, Analyse/Synthese team manager).
or by fax at: (33 1) 44 78 15 40, care of Xavier.Rodet
or by surface mail to: Xavier Rodet, IRCAM, 1 Place Stravinsky, 75004 Paris.
IRCAM:
IRCAM is a leading non-profit organization dedicated to musical production, R&D and education in acoustics and music, located in the center of Paris (France), next to the Pompidou Center. It hosts composers, researchers and students from many countries cooperating in contemporary music production, scientific and applied research. The main topics addressed in its R&D department are acoustics, psychoacoustics, audio synthesis and processing, computer aided composition, user interfaces, real time systems. Detailed activities of IRCAM and its groups are presented on our WWW server.

RESEARCH AND DEVELOPMENT POSITION IN "AUDIO CONTENT ACCESS" at IRCAM (Paris)

PRESENTATION OF THE MUSICDISCOVER PROJECT :
The goal of the MusicDiscover project is to give access to the contents of musical audios recordings (as it is the case, for example, for texts), i.e. to a structured description, as complete as possible, of the recordings: melody, genre/style, rate/rhythm, instrumentation, musical structure, harmony, etc. The principal objective is thus to develop and evaluate means directed towards the contents, which include techniques and tools for analysis, indexing, representation and search for information. These means will make it possible to build and use such a structured description. This project of the ACI "Masses of Data" is carried out in collaboration between Ircam (Paris), Get-Telecom (Paris) and the LIRIS (Lyon) since October 2004. The principal lines of research are :
- Rhythmic analysis and detection of ruptures
- Recognition of musical instruments and indexing
- Source Separation
- Structured Description
- Research of music by similarity
- Recognition of musical titles
- Classification of musical titles in genre and emotion.
The available position relates to the construction and the use of the Structured Description in collaboration with the other lines of research.
DEVELOPMENTS TASKS:
A position is available from December 1st 2006 within the "Equipe Analyse/Synthese" of Ircam for a 9 months total duration. The contents of work are as follows:
- Participation in the design of a Structured Description
- Software development for construction and use of Structured Descriptions
- Participation in the definition and development of the graphic interface
- Participation in the evaluations
REQUIRED EXPERIENCE AND COMPETENCE:
- Experience of research in Audio Indexing and signal processing
- Experience in Flash, C and C++ and Matlab programming.
- High productivity, methodical work, excellent programming style.
- Good knowledge of UNIX and Windows environments.
AVAILABILITY :
- The position is available in the "Analysis/Synthesis" team in the R&D department from November 1st 2006 for a duration of 9 months.
EEC WORKING PAPERS :
- In order to start immediately, the candidate should preferably have EEC citizenship or already own valid EEC working papers.
SALARY:
- According to background and experience.
TO APPLY:
- Please send your resume with qualifications and informations adressing the above issues, preferably by email to Xavier Rodet, Analyse/Synthese team manager.
or by fax at: (33 1) 44 78 15 40, care of Xavier.Rodet
or by surface mail to: Xavier Rodet, IRCAM, 1 Place Stravinsky, 75004 Paris.
Introducing IRCAM
IRCAM is a leading non-profit organization dedicated to musical production, R&D and education in acoustics and music, located in the center of Paris (France), next to the Pompidou Center. It hosts composers, researchers and students from many countries cooperating in contemporary music production, scientific and applied research. The main topics addressed in its R&D departement are acoustics, psychoacoustics, audio synthesis and processing, computer aided composition, user interfaces, real time systems. Detailed activities of IRCAM and its groups are presented on our WWW server

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JOURNALS

CfP: Speech Communication Journal - Special Issue on “Evaluating new methods and models for advanced speech-based interactive systems”

The aim of this special issue is to explore new evaluation techniques and strategies as applied to advanced dialogue systems, including new models and methods. Original, previously unpublished submissions addressing some (or all) of the following questions are encouraged:
1. What characteristics of spoken language interaction can and should be incorporated into advanced spoken dialogue systems?
2. What are the best methods for designing such systems? To what extent are automatic design methods appropriate or possible?
3. What criteria can be defined for the evaluation of the performance of advanced spoken dialogue systems?
4. Under what circumstances should these criteria be used?
5. How effective are these criteria in isolating problems with a dialogue strategy and in measuring how correction of the problems improves the dialogue?
6. How can these criteria be used to compare and evaluate alternative dialogue strategies and methods for the design and implementation of dialogue systems?
7. How can these criteria be used to compare the pre-modification version and the post-modification version of a dialogue strategy as developers attempt to improve the dialogue strategy?
8. How can the evaluation process be streamlined so that it can be frequently and effectively applied to the improvement and comparison of dialogue strategies?
Guest Editors
Michael McTear, University of Ulster, UK
Kristiina Jokinen, University of Helsinki, Finland
James Larson, Oregon Graduate Institute, Oregon, USA
Important Dates
Submission Deadline: 30th June 2007
Notification of Acceptance: 30th November 2007
Final manuscript due: 31 January 2008
Tentative Publication Date: June 2008
Submission Procedure
Prospective authors should follow the regular guidelines of the Speech Communication journal for electronic submission (http://ees.elsevier.com/specom). During submission authors must select the Article Type: "Special Issue: Spoken Dialogue Technology, not "Regular Paper", and also select Professor Marc Swerts as the handling Editor-in-Chief.
Full text of CFP

CfP: IEEE SIGNAL PROCESSING MAGAZINE- Special Issue on Spoken Language Technology

The evolution of speech and language technologies over the past decade has spawned an exciting new research area known as Spoken Language Tech nology (SLT). Technological advances in SLT promise to provide ubiquit ous and personalized access to information, communication, and entertai nment services. For example, advances in natural language understanding and large vocabulary continuous speech recognition have resulted in a new generation of automated contact center services that offer callers the flexibility to speak their request naturally using their own words as opposed to the words dictated to them by the machine. Advances in ma chine translation technology have resulted in speech-to-speech translat ion products that offer multi-party multi-lingual communication. Advanc es in information search and data mining are providing the means to ext ract intelligence information from large corpora of speech data (e.g., TV programs, call center data) to help improve business operation and s earch for information rapidly without having to listen to conversations .
This special issue on Spoken Language Technology is motivated by the fi rst SLT workshop, Aruba, December 2006, jointly sponsored by IEEE and A CL (www.slt2006.org). The goal is to solicit tutorial articles with com prehensive surveys of important theories, algorithms, tools, and applic ations of SLT on existing and new commercial, academic and government a pplications. Prospective authors should submit a white paper summarizin g the motivation, the significance of the topic, brief history, and an outline of the content. Authors with accepted proposals will be invited to write a full manuscript.
Scope of topics:
Publications in the following areas are strongly encouraged
Spoken language understanding
Dialog management
Spoken language generation
Spoken document retrieval
Information extraction from speech
Question answering from speech
Spoken document summarization
Machine translation of spoken language
Speech data mining and search
Voice-based human computer interfaces
Spoken dialog systems, applications and standards
Multimodal processing, systems and standards
Machine learning for spoken language processing
Speech and language processing in the world wide web
Submission Procedure:
Prospective authors should submit their white papers to the web submiss ion system at http://www.ee.columbia.edu/spm according to the following timetable. The white papers should be three pages maximum
Important dates
White paper due: June 1, 2007
Invitation notification: July 1, 2007
Manuscript due: October 1, 2007
Acceptance Notification: December 1, 2007
Final Manuscript due: January 15, 2008
Publication date: May, 2008
Guest Editors:
Mazin Gilbert
AT&T Labs - Research
180 Park Avenue
Florham Park, NJ, 07932
Kevin Knight
University of Southern California
4676 Admiralty Way
Marina del Rey, CA 90292
Steve Young
Cambridge University
Trumpington Street
Cambridge, CB2 1PZ

Call for Papers- Special Issue of the IEEE Transactions on Audio, Speech and Language Processing on New Approaches to Statistical Speech and Text Processing

Dramatic advances in automatic speech recognition (ASR) technology in recent years has enabled serious growth in spoken language processing research, both for human-computer interaction and spoken document processing. The challenges of working with spoken language, including ASR errors and disfluencies, were major factors in the adoption of statistical techniques in the language processing community. Statistical methods now dominate many areas of text processing as well, enabled by growing collections of linguistic data resources and developments in machine learning. While transfer of methods from spoken- to written- language processing continues, advances in written-language processing also now have a significant impact on spoken-language processing. This issue seeks to highlight the cross-fertilization in speech and text processing by publishing novel statistical modeling and learning methods that span a variety of language processing applications.
We invite papers describing new approaches to statistical language processing of both spoken and written language. Submissions must not have been previously published, with the exception that substantial extensions of conference papers will be considered. Of particular interest are methods that transfer recent developments from text processing to speech processing and vice versa, but new methods in one domain are also welcome. Papers describing new strategies for integrating acoustic and linguistic cues in spoken language processing are also encouraged.
Topics of interest include:
- Unsupervised and semi-supervised learning
- Discriminative learning
- Transfer or adaptation to new domains
- Active learning
- Reinforcement learning
- Memory-based learning and neighborhood methods
- Novel statistical models
- Statistical methods for feature selection or transformation
Specific applications of interest include information extraction, question answering, text segmentation and classification, summarization, translation, language generation and spoken language dialogs. Papers that address component problems of these larger applications are also encouraged, including parsing, discourse analysis, and talker interaction analysis. The issue aims to cover a variety of applications as well as different statistical methods.
Submission procedure:
Prospective authors should prepare manuscripts according to the Information for Authors as published in any recent issue of the Transactions. Note that all rules will apply with regard to submission lengths, mandatory overlength page charges, and color charges. Manuscripts should be submitted electronically through the online IEEE manuscript submission system. When selecting a manuscript type, authors must click on "Special Issue of TASLP on New Approaches to Statistical Speech and Text Processing". Authors should follow the instructions for the IEEE Transactions Audio, Speech and Language Processing and indicate in the Comments to the Editor-in-Chief that the manuscript is submitted for publication in the Special Issue on New Approaches to Statistical Speech and Text Processing. We require a completed copyright form to be signed and faxed to +1-732-562-8905 at the time of submission. Please indicate the manuscript number on the top of the page.
Schedule:
Submission deadline: 15 June 2007
Notification of final acceptance: 15 December 2008
Final manuscript due: 1 February 2008
Publication date: May 2008
Guest Editors:
Dr. Bill Byrne Cambridge University, UK
Dr. Mark Johnson Brown University, USA
Dr. Lillian Lee Cornell University, USA
Dr. Steve Renals University of Edinburgh, UK

Call for papers for a special issue of Speech Communication on Iberian Languages

Iberian languages (henceforth IL) are amongst the most widely spoken languages in the world. Nowadays, 628 million people on virtually all continents have Spanish, Portuguese, Catalan, Basque, Galician, etc. as their official language. Consequently, important speech research centers and companies, both public and private, are focusing their interest on those languages. This effort has resulted in novel and generic approaches applicable to any language, as well as in the optimization of existing techniques or systems. It is worth highlighting that the community working on speech science and technology in IL speaking countries has already reached world-class level in many areas and has continuously increased in size in the last 15 years.
Speech technology proposed in the context of a non-Iberian language (e.g., English) may not be directly applicable to IL. All linguistic and paralinguistic dimensions, from phonetics to pragmatics, are amongst the features that certainly distinguish IL from others considered in speech science and technology research. As a result, original work and optimization of existing techniques and systems may be necessary in many areas of Iberian spoken language research.
The purpose of this Special Issue is to present recent progress and significant advances in all areas of speech science and technology research in the context of IL. Submitted papers must address topics specific to IL and/or issues raised by analyses of spoken data that shed light on speech science and linguistic theories regarding these languages. Research which deals with IL data, but makes use of standard techniques should not be submitted for this Special Issue. However, both research presenting relevant optimization of current technology and systems, and work exploring specific features of IL spoken corpora will be considered for submission.
This Special Issue is one of the first initiatives proposed by the recently created SIG-IL (ISCA Special Interest Group on Iberian Languages, URL http://www.il-sig.org). The purposes of the SIG-IL are to promote research activities on IL, to sponsor and/or organise meetings, workshops and other events on related topics, and to make speech corpora publicly available by promoting joint evaluation efforts. Furthermore, the SIG-IL is also strongly committed to encouraging world-class research within its community in order to contribute with new ideas to the field of speech science and technology. Original, previously unpublished submissions for the following areas, involving IL and detailing the language-specific aspects, are encouraged:
Topics
o Linguistics, Phonology and Phonetics
o Prosody
o Paralinguistic & Nonlinguistic Information in Speech
o Discourse & Dialogue
o Speech Production
o Speech Perception
o Physiology & Pathology
o Spoken Language Acquisition, Development and Learning
o Spoken Language Generation & Synthesis
o Language/Dialect Identification
o Speech and Speaker Recognition: acoustic, language and pronunciation modeling.
o Spoken Language Understanding
o Multi-modal / Multi-lingual Processing
o Spoken Language Extraction/Retrieval
o Spoken Language Translation
o Spoken/Multi-modal Dialogue Systems
o Spoken Language Resources and Annotation
o Evaluation and Standardization
o Spoken Language Technology for the Aged and Disabled (e-inclusion)
o Spoken Language Technology for Education (e-learning)
o Interdisciplinary Topics in Speech and Language
o New Applications
Guest Editors
Isabel Trancoso INESC-ID, Portugal
Nestor Becerra-Yoma Univ. de Chile, Chile
Plinio A. Barbosa Univ. of Campinas, Brazil
Rubén San-Segundo UPM, Spain
Kuldip Plaiwal Griffith University, Australia
Important Dates
Submission deadline: May 31st, 2007
Notification of acceptance: October 31st, 2007
Final manuscript due: December 30th, 2007
Tentative publication date: March, 2008
Submission Procedure
Prospective authors should follow the regular guidelines of the Speech Communication Journal for electronic submission (http://ees.elsevier.com/specom). During submission authors must select the Section “Special Issue Paper”, not “Regular Paper”, and the title of the special issue should be referenced in the “Comments” (Special Issue on Iberian Languages) page along with any other information.

Papers accepted for FUTURE PUBLICATION in Speech Communication

Full text available on http://www.sciencedirect.com/ for Speech Communication subscribers and subscribing institutions. Free access for all to the titles and abstracts of all volumes and even by clicking on Articles in press and then Selected papers.

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FUTURE CONFERENCES

Publication policy: Hereunder, you will find very short announcements of future events. The full call for participation can be accessed on the conference websites
See also our Web pages (http://www.isca-speech.org/) on conferences and workshops.

FUTURE INTERSPEECH CONFERENCES

INTERSPEECH 2007-EUROSPEECH
August 27-31,2007,Antwerp, Belgium
Chair: Dirk van Compernolle, K.U.Leuven and Lou Boves, K.U.Nijmegen
Website
INTERSPEECH 2007 is the eighth conference in the annual series of INTERSPEECH events and also the tenth biennial EUROSPEECH conference. The conference is jointly organized by scientists from the Netherlands and Belgium, and will be held in Antwerp, Belgium, August 27-31, 2007, under the sponsorship of the International Speech Communication Association (ISCA).
The INTERSPEECH meetings are considered to be the top international conferences in spoken language processing, with more than 1000 attendees from universities, industry, and government agencies. The conference offers the prospect of meeting the future leaders of our field, exchanging ideas, and exploring opportunities for collaboration, employment, and sales through keynote talks, tutorials, technical sessions, exhibits, and poster sessions.
In recent years the INTERSPEECH meetings have taken place in a number of exciting venues including most recently Pittsburgh, Lisbon, Jeju Island (Korea), Geneva, Denver, Aalborg (Denmark), and Beijing.
AREAS AND TOPICS OF INTEREST:
Interspeech is the world's largest and most comprehensive conference on Speech Science and Speech Technology and it solicits papers in the following areas and topics:
A.Human speech production, perception and communication
Phonology and phonetics
Discourse and dialogue
Prosody (production, perception, prosodic structure)
Paralinguistic and nonlinguistic cues (e.g. emotion and expression)
Speech production
Speech perception
Physiology and pathology
Spoken language acquisition, development and learning
B.Speech and Language technology
Speech and audio processing
Speech enhancement
Speech coding and transmission
Spoken language generation and synthesis
Speech recognition
Spoken language understanding
Accent and language identification
Cross-lingual and multi-lingual processing
Multimodal/multimedia signal processing
Speaker characterization and recognition
C.Spoken language systems and applications
Dialogue systems
Systems for information retrieval
Systems for translation
Applications for aged and handicapped persons
Applications for learning and education
Other applications
D.Resources, standardization and evaluation
Spoken language resources and annotation
Evaluation and standardization
PAPER SUBMISSION
Authors will have to declare that their contribution is original and not being submitted for publication elsewhere (e.g., another conference, workshop, or journal).
Each corresponding author will be notified by e-mail of the acceptance or rejection of his paper by May 25, 2007. Minor updates of accepted papers will be allowed during May 25 - June 3, 2007.
More information is available on the conference website
INVITED KEYNOTE SPEAKERS
Keynote speaker: ISCA Medalist Prof. Victor Zue (MIT, Cambridge, MA)
Title: On Organic Interfaces
Keynote speaker: Prof. Sophie Scott (UCL, London, UK)
Title: How the Brain Decodes Speech – Some Perspectives from Functional Imaging
Keynote speaker: Prof. Alex Waibel (CMU, Pittsburgh, PA; University of Karlsruhe, Germany)
Title: Computer-Supported Human-Human Multilingual Communication
Keynote speaker: Prof. Luc Steels (Free University Brussels, Belgium; Sony Computer Science Laboratory Paris, France)
Title: Can Robots Invent their Own Language?
IMPORTANT DATES
Full paper submission deadline: March 23, 2007
Notification of paper acceptance/rejection May 25, 2007
Early registration deadline: June 22, 2007
Further information via website or email.
ORGANIZERS
Professor Dirk Van Compernolle (General Chair)
Professor Lou Boves (General Co-Chair)
c/o Annitta De Messemaeker
Katholieke Universiteit Leuven
Department of Electrical Engineering
Kasteelpark Arenberg 10
B3001 Heverlee
Belgium
Fax: +32 16 321723
Email
Website

INTERSPEECH 2008-ICSLP
September 22-26, 2008, Brisbane, Queensland, Australia
Chairman: Denis Burnham, MARCS, University of West Sydney.

INTERSPEECH 2009-EUROSPEECH
Brighton, UK,
Chairman: Prof. Roger Moore, University of Sheffield.

INTERSPEECH 2010-ICSLP
Chiba, Japan
ISCA is pleased to announce that INTERSPEECH 2010 will take place in Makuhari-Messe, Chiba, Japan, September 26-30, 2010. The event will be chaired by Keikichi Hirose (Univ. Tokyo), and will have as a theme "Towards Spoken Language Processing for All - Regardless of Age, Health Conditions, Native Languages, Environment, etc."

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FUTURE ISCA TUTORIAL AND RESEARCH WORKSHOP (ITRW)

Third ITRW on NON-LINEAR SPEECH PROCESSING (NOLISP'07)

May 22-25, 2007 , Paris, France
Website
Many specifics of the speech signal are not well addressed by the conventional models currently used in the field of speech processing. The purpose of the workshop is to present and discuss novel ideas, work and results related to alternative techniques for speech processing, which depart from mainstream approaches.
SUBMISSION
Prospective authors are invited to submit a 3 to 4-page paper proposal in English, which will be evaluated by the Scientific Committee. Final papers will be due 1 month after the workshop, for inclusion in the CD-ROM proceedings. A special issue in Speech Communication (Elsevier) will follows.
KEY DATES
Submission (full paper): 15 January 2007
Notification of acceptance: 23 February 2007
Workshop: 22-25 May 2007
Final (revised) paper: 25 June

6th ISCA Speech Synthesis Research Workshop (SSW-6)

University of Bonn (Germany), August 22-24, 2007
A satellite of INTERSPEECH 2007 (Antwerp)in collaboration with SynSIG and IfK (University of Bonn)
Organized shortly after the 16th International Congress on Phonetic Sciences (Saarbrücken, Germany, August 6-10, 2007). Like its predecessors in Autrans (France) 1990, New Paltz (NY, USA) 1994, Jenolan (Australia) 1998, Pitlochry (UK) 2001, and Pittsburgh (PA, USA) 2004, SSW-6 will cover all aspects of speech synthesis and adjacent fields, such as:
TOPICS (updated list)
* Text processing for speech synthesis
* Prosody Generation for speech synthesis
* Speech modeling for speech synthesis applications
* Signal processing for speech synthesis
* Concatenative speech synthesis (diphones, polyphones, unit selection)
* Articulatory synthesis
* Statistical parametric speech synthesis
* Voice transformation/conversion/adaptation for speech synthesis
* Expressive speech synthesis
* Multilingual and/or multimodal speech synthesis
* Text-to-speech and content-to-speech
* Singing speech synthesis
* Systems and applications involving speech synthesis
* Techniques for assessing synthetic speech quality
* Language resources for speech synthesis
* Aids for the handicapped involving speech synthesis.
Deadlines (updated)
* Full-paper submission (up to 6 pages) - May 14, 2007 (EXTENDED DEADLINE!)
* Notification of acceptance - June 25, 2007
* Deadline for paper modification - July 15, 2007
Please send your papers, preferably as PDF files, as an e-mail attachment. Further information can soon be obtained from the website of the workshop,
Contact
Prof. Wolfgang Hess

8th Workshop on Discourse and Dialogue (SIGdial), Antwerp, Belgium

Antwerp, September 2-3, 2007
Held immediately following Interspeech 2007
Continuing with a series of successful workshops in Sydney, Lisbon, Boston, Sapporo, Philadelphia, Aalborg, and Hong Kong, this workshop spans the ACL and ISCA SIGdial interest area of discourse and dialogue. This series provides a regular forum for the presentation of research in this area to both the larger SIGdial community as well as researchers outside this community. The workshop is organized by SIGdial, which is sponsored jointly by ACL and ISCA.
Topics of Interest
We welcome formal, corpus-based, implementation or
analytical work on discourse and dialogue including but not restricted to the following three themes:
1. Discourse Processing and Dialogue Systems
Discourse semantic and pragmatic issues in NLP applications such as text summarization, question answering, information retrieval including topics like:
· Discourse structure, temporal structure, information structure
· Discourse markers, cues and particles and their use
. (Co-)Reference and anaphora resolution, metonymy and bridging resolution
· Subjectivity, opinions and semantic orientation
Spoken, multi-modal, and text/web based dialogue systems including topics such as:
· Dialogue management models;
· Speech and gesture, text and graphics integration;
· Strategies for preventing, detecting or handling miscommunication (repair and correction types, clarification and under-specificity, grounding and feedback strategies);
· Utilizing prosodic information for understanding and for disambiguation;
2. Corpora, Tools and Methodology
Corpus-based work on discourse and spoken, text-based and multi-modal dialogue including its support, in particular:
· Annotation tools and coding schemes;
· Data resources for discourse and dialogue studies;
· Corpus-based techniques and analysis (including machine learning);
· Evaluation of systems and components, including methodology, metrics and case studies;
The pragmatics and/or semantics of discourse and dialogue (i.e. beyond a single sentence) including the following issues:
· The semantics/pragmatics of dialogue acts (including those which are less studied in the semantics/pragmatics framework);
· Models of discourse/dialogue structure and their relation to referential and relational structure;
· Prosody in discourse and dialogue;
· Models of presupposition and accommodation; operational models of conversational implicature.
Submissions
The program committee welcomes the submission of long papers for full plenary presentation as well as short papers and demonstrations. Short papers and demo descriptions will be featured in short plenary presentations, followed by posters and demonstrations.
· Long papers must be no longer than 8 pages, including title, examples, references, etc. In addition to this, two additional pages are allowed as an appendix which may include extended example discourses or dialogues, algorithms, graphical representations, etc.
· Short papers and demo descriptions should aim to be 4 pages or less (including title, examples, references, etc.). Please use the official ACL style files. Submission/Reviewing will be managed by the START system. Link to follow. Papers that have been or will be submitted to other meetings or publications must provide this information (see submission format). SIGdial 07 cannot accept for publication or presentation work that will be (or has been) published elsewhere. Authors are encouraged to make illustrative materials available, on the web or otherwise. For example, excerpts of recorded conversations, recordings of human-computer dialogues, interfaces to working systems, etc.
Important Dates (subject to change)
Submission May 2, 2007
Notification June 13, 2007
Final submissions July 6, 2007
Workshop September 2-3, 2007
Websites
Workshop website:To be announced
Submission website:To be announced
Sigdial website
Interspeech 2007 website
Email
Program Committee (confirmed)
Harry Bunt, Tilburg University, Netherlands (co-chair)
Tim Paek, Microsoft Research, USA (co-chair)
Simon Keizer, Tilburg University, Netherlands (local chair)
Wolfgang Minker, University of Ulm, Germany
David Traum, USC/ICT, USA

CfP-SLaTE Workshop on Speech and Language Technology in Education
ISCA Tutorial and Research Workshop

The Summit Inn, Farmington, Pennsylvania USA October 1-3, 2007.
Website
Speech and natural language processing technologies have evolved from being emerging new technologies to being reliable techniques that can be used in real applications. One worthwhile application is Computer-Assisted Language Learning. This is not only helpful to the end user, the language learner, but also to the researcher who can learn more about the technology from observing its use in a real setting. This workshop will include presentations of both research projects and real applications in the domain of speech and language technology in education.
IMPORTANT DATES
Full paper deadline: May 1, 2007.
Notification of acceptance: July 1, 2007.
Early registration deadline: August 1, 2007.
Preliminary programme available: September 1, 2007.
Workshop will take place: October 1-3, 2007.
LOCATION
The workshop will be held in the beautiful Laurel Highlands. In early October the vegetation in the Highlands puts on a beautiful show of colors and the weather is still not too chilly. The event will take place at the Summit Inn, situated on one of the Laurel Ridges. It is close to the Laurel Caverns where amateur spelunkers can visit the underground caverns. The first night event will be a hayride and dinner at a local winery and the banquet will take place at Frank Lloyd Wright’s wonderful Fallingwater.
TOPICS
The workshop will cover all topics which come under the purlieu of speech and language technology for education. In accordance with the spirit of the ITRWs, the upcoming workshop will focus on research and results, give information on tools and welcome prototype demons
trations of potential future applications. The workshop will focus on research issues, applications, development tools and collaboration. It will be concerned with all topics which fit under the purview of speech and language technology for education. Papers will discuss theories, applications, evaluation, limitations, persistent difficulties, general research tools and techniques. Papers that critically evaluate approaches or processing strategies will be especially welcome, as will prototype demonstrations of real-world applications.
The scope of acceptable topic interests includes but is not limited to:
- Use of speech recognition for CALL
- Use of natural language processing for CALL
- Use of spoken language dialogue for CALL
- Applications using speech and/or natural language processing for CALL
- CALL tutoring systems
- Assessment of CALL tutors

ORGANIZATION-CONTACT
The workshop is being organized by the new ISCA Special Interest Group, SLaTE. The general chair is Dr. Maxine Eskenazi from Carnegie Mellon University .
PROGRAMME
As per the spirit of ITRWs, the format of the workshop will consist of a non-overlapping mixture of oral, poster and demo sessions. Internationally recognized experts from pertinent areas will deliver several keynote lectures on topics of particular interest. All poster sessions will be opened by an oral summary by the session chair. A number of poster sessions will be succeeded by a discussion session focussing on the subject of the session. The aim of this structure is to ensure a lively and valuable workshop for all involved. Furthermore, the organizers would like to encourage researchers and industrialists to bring along their applications, as well as prototype demonstrations and design tools where appropriate. The official language of the workshop is English. This is to help guarantee the highest degree of international accessibility to the workshop. At the opening of the workshop hardcopies and CD-ROM of the abstracts and proceedings will be available.
CALL FOR PAPERS
We seek outstanding technical articles in the vein discussed above. For those who intend to submit papers, the deadline is May 1, 2007. Following preliminary review by the committee, notification will be sent regarding acceptance/rejection. Interested authors should send full 4 page camera-ready papers.
REGISTRATION FEE
The fee for the workshop, including a booklet of Abstracts, the Proceedings on CD-ROM is:
- $325 for ISCA members and
- $225 for ISCA student members with valid identification
Registrations after August 1, 2007 cannot be guaranteed.
ADDITIONAL REGISTRATION INFORMATION
All meals except breakfast for the two and a half days as well as the two special events are included in this price. Hotel accommodations are $119 per night , and breakfast is about $10. Upon request we will furnish bus transport from the Greater Pittsburgh Airport and from Pittsburgh to Farmington at a cost of about $30. ISCA membership is 55 Euros. You must be a member of ISCA to attend this workshop.

ITRW Odyssey 2008

The Speaker and Language Recognition Workshop
21-25 January 2008, Stellenbosch, South Africa
Topics
* Speaker recognition(identification, verification, segmentation, clustering)
* Text dependent and independent speaker recognition
* Multispeaker training and detection
* Speaker characterization and adaptation
* Features for speaker recognition
* Robustness in channels
* Robust classification and fusion
* Speaker recognition corporaand evaluation
* Use of extended training data
* Speaker recognition with speaker recognition
* Forensics, multimodality and multimedia speaker recogntion
* Speaker and language confidence estimation
* Language, dialect and accent recognition
* Speaker synthesis and transformation
* Biometrics
* Human recognition
* Commercial applications
Paper submission
Proaspective authors are invited to submit papers written in English via the Odyssey website. The style guide, templates,and submission form can be downloaded from the Odyssey website. Two members of the scientific committee will review each paper. Each accepted paper must have at least one registered author. The Proceedings will be published on CD
Schedule
Draft paper due July 15, 2007
Notification of acceptance September 15,2007
Final paper due October 30, 2007
Preliminary program November 30, 2007
Workshop January 21-25, 2008
Futher informations: venue, registation...
On the workshop website
Chairs
Niko Brummer, Spescom Data Voice, South Africa
Johan du Preez.Stellenbosch University,South Africa

ITRW on Evidence-based Voice and Speech Rehabilitation in Head & Neck Oncology

May 2008, Amsterdam, The Netherlands,
Cancer in the head and neck area and its treatment can have debilitating effects on communication. Currently available treatment options such as radiotherapy, surgery, chemo-radiation, or a combination of these can often be curative. However, each of these options affects parts of the vocal tract and/or voice to a more or lesser degree. When the vocal tract or voice no longer functions optimally, this affects communication. For example, radiotherapy can result in poor voice quality, limiting the speaker’s vocal performance (fatigue from speaking, avoidance of certain communicative situations, etc.). Surgical removal of the larynx necessitates an alternative voicing source, which generally results in a poor voice quality, but further affects intelligibility and the prosodic structure of speech. Similarly, a commando procedure (resection involving portions of the mandible / floor of the mouth / mobile tongue) can have a negative effect on speech intelligibility. This 2 day tutorial and research workshop will focus on evidence-based rehabilitation of voice and speech in head and neck oncology. There will be 4 half day sessions, 3 of which will deal with issues concerning total laryngectomy. One session will be devoted to research on rehabilitation of other head and neck cancer sites. The chairpersons of each session will prepare a work document on the specific topic at hand (together with the two keynote lecturers assigned), which will be discussed in a subsequent round table session. After this there will be a 30’ poster session, allowing 9-10 short presentations. Each presentation consists of maximally 4 slides, and is meant to highlight the poster’s key points. Posters will be visited in the subsequent poster visit session. The final work document will refer to all research presently available, discuss its (clinical) relevance, and will attempt to provide directions for future research. The combined work document, keynote lectures and poster abstracts/papers will be published under the auspices of ISCA.
Organizers
prof. dr. Frans JM Hilgers
prof. dr. Louis CW Pols,
dr. Maya van Rossum.
Sponsoring institutions:
Institute of Phonetic Sciences - Amsterdam Center for Language and Communication,
The Netherlands Cancer Institute – Antoni van Leeuwenhoek Hospital
Dates and submission details as well as a website address will be announced in a later issue.

Audio Visual Speech Processing Workshop (AVSP 2008)

Tentative location:Queensland coast near Brisbane (most likely South Stradbroke Island)
Tentative date: 27-29 September 2008 (immediately after Interspeech 2008)
Following in the footsteps of previous AVSP workshops / conferences, AVSP workshop (ISCA Research and Tutorial Workshop) will be hold concomitantly to Interspeech2008, Brisbane, Australia, 22-26 September 2008. The aim of AVSP2008 is to bring together researchers and practitioners in areas related to auditory-visual speech processing. These include human and machine AVSP, linguistics, psychology, and computer science. One of the aims of the AVSP workshops is to foster collaborations across disciplines, as AVSP research is inherently multi-disciplinary. The workshop will include a number of tutorials / keynote addresses by internationally renowned researchers in the area of AVSP.
Organizers
Roland Goecke, Simon Lucey, Patrick Lucey
Australian National University,RSISE, Bldg. 115, Australian National University, Canberra, ACT 0200, Australia

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FORTHCOMING EVENTS SUPPORTED (but not organized) by ISCA

CFP- ETSI Workshop: Speech and Noise in Wideband Communication

22nd & 23rd May 2007, at ETSI Headquarters in Sophia Antipolis, France.
As new types of voice coders, noise cancellation algorithms, transmission technologies and consequently transmission impairments enter the scene and convergence becomes ever more a reality, the standardization community faces new challenges.
Being organised by TC STQ, STF 294 and Mesaqin, under contract to ETSI, the main objectives of the workshop are to:
* Discuss the status, latest advances and trends in wideband speech and audio coding, in particular in the presence of interfering sounds and noise
* Present the results of STF 294: Improving the quality of eEurope wideband speech applications by developing a standardised performance testing and evaluation methodology for background noise transmission
* Exchange information and establish relationships between research, state and industrial organizations involved in the topic
Topics that will be addressed will include speech and audio wideband coding, noise suppression and its artefacts, and quality assessment.
A round table discussion will permit participants to offer views on the current issues and challenges that we will be facing in the future.
Participation in the workshop is free of charge, and open to everyone.
Candidate speakers are invited to send an abstract of their presentation to Jan Holub by Friday 16th March 2007.
For further details, consult the workshop Website For registration please see our web

AVSP 2007

International Conference on Auditory-Visual Speech Processing 2007,
August 31 - September 3, 2007
Kasteel Groenendael, Hilvarenbeek, The Netherlands
The next International Conference on Auditory-Visual Speech Processing (AVSP 2007) will be organised by different members of Tilburg University (The Netherlands). It will take place in Kasteel Groenendael in Hilvarenbeek (The Netherlands) from August 31, 2007 till September 3, 2007, immediately following Interspeech 2007 in Antwerp (Belgium). Hilvarenbeek is located at close distance from Antwerp, so that attendance at AVSP 2007 can easily be combined with participation in Interspeech 2007.
Auditory-visual speech production and perception by human and machine is an interdisciplinary and cross-linguistic field which has attracted speech scientists, cognitive psychologists, phoneticians, computational engineers, and researchers in language learning studies. Since the inaugural workshop in Bonas in 1995, Auditory-Visual Speech Processing workshops have been organised on a regular basis (see an overview at the avisa website). In line with previous meetings, this conference will consist of a mixture of regular presentations (both posters and oral), and lectures by invited speakers. All presentations will be plenary.
We are happy to announce that the following experts have agreed to give a keynote lecture at our conference: Sotaro Kita (Birmingham)
Asif Ghazanfar (Princeton)
More details about the conference can be found on the website
Further information

CfP SPECOM 2007

The 12th International Conference on Speech and Computer
October 15-18, 2007
Organized by Moscow State Linguistic University
General Chair:
Prof. Irina Khaleeva (Moscow State Linguistic University)
Chair:
Prof. Rodmonga Potapova (Moscow State Linguistic University)
SPECOM'07 is the twelfth conference in the annual series of SPECOM events. It is organized by Moscow State Linguistic University and will be held in Moscow, Russia, under the sponsorship of Russian Foundation for Basic Research (RFBR), Ministry of Education and Science of Russian Federation, the International Speech Communication Association (ISCA) and others. SPECOM'07 will cover various aspects of speech science and technology. The program of the conference will include keynote lectures by internationally renowned scientists, parallel oral and poster sessions and an exhibition. The sci-tech exhibition that will be held during the conference will be open to companies and research institutions. The official language of the Conference will be English.
Important Dates (Extended)
Paper submission opening February 1, 2007
Full paper deadline May 25, 2007
Notification of paper acceptance June 15, 2007
Conference October 15-18, 2007
Topics
o Speech signal coding and decoding; multi-channel transmitted speech intelligibility; speech information security
o Speech production and perception modeling
o Automatic processing of multilingual, multimodal and multimedia information
o Linguistic, para- and extralinguistic communicative strategies
o Development and testing of automatic voice and speech systems for speaker verification; speaker psychoemotional state and native language identification
o Automatic speech recognition and understanding systems
o Language and speech information processing systems for robotechnics
o Automated translation systems
o New information technologies for spoken language acquisition, development and learning
o Text-to-speech conversion systems
o Spoken and written natural language corpora linguistics
o Multifunctional expert and information retrieval systems
o Future of multi-purpose and anti-terrorist speech technologies
PAPER SUBMISSION
The deadline for full paper submission (4-6 pages) is April 25, 2007. Papers are to be sent by e-mail to specom2007@mail.ru. All manuscripts must be in English. Please note that the size of a single letter must not exceed 10 Megabytes (that is, the total size of all the attached files should not be greater than 7 Megabytes to leave room for recoding operations performed by the e-mail software). In case the paper files are larger than 7 Megabytes, it is recommended to pack them into a split WinRar or WinZip archive and send part by part in a series of letter.
All the papers will be reviewed by an international scientific committee. Each author will be notified by e-mail of the acceptance or rejection of her/his paper by May 30, 2007. Minor updates of accepted papers will be allowed during May 30 - June 15, 2007.
EVALUATION CRITERIA
Submission of a paper or poster is more likely to be accepted if it is original, innovative, and contributes to the practice of worldwide scientific communication. Quality of work, clarity and completeness of the submitted materials will be considered.
REGISTRATION
Registration will be available at the Conference on arrival. The registration fees are planned to be approximately as follows:
Regular 500 EUR
Students/PG Students 200 EUR
NIS (New Independent States), Regular 300 EUR
NIS, Students/PG Students 100 EUR
Russia, Regular 150 EUR
Russia, Students/PG Students (no Proceedings) Free Extra Copy of Proceedings (hard copy) 20 EUR
Extra Proceedings CD/DVD 10 EUR
Information regarding accommodation costs will be available later. All the registration and accommodation payments will be accepted in cash during the registration procedure on arrival.
PAPER PREPARATION GUIDELINES
In the following you will find guidelines for preparing your full paper to SPECOM'07 electronically.
· To achieve the best viewing experience both for the Proceedings and the CD (or DVD), we strongly encourage you to use Times Roman font. This is needed in order to give the Proceedings a uniform look. Please use the attached printable version of this newsletter as a model.
· Authors are requested to submit PDF files of their manuscripts, generated from the original Microsoft Word sources. PDF files can be generated with commercially available tools or with free software such as PDFCreator.
· Paper Title - The paper title must be in boldface. All non-function words must be capitalized, and all other words in the title must be lower case. The paper title is centered.
· Authors' Names - The authors' names (italicized) and affiliations (not italicized) appear centered below the paper title.
· Abstract - Each paper must contain an abstract that appears at the beginning of the paper.
· Major Headings - Major headings are in boldface.
· Sub Headings - Sub headings appear like major headings, except that they are in italics and not bold face.
· References - Number and list all references at the end of the paper. The references are numbered in order of appearance in the document. When referring to them in the text, type the corresponding reference number in square brackets as shown at the end of this sentence [1].
· Illustrations - Illustrations must appear within the designated margins, and must be positioned within the paper margins. Caption and number every illustration. All half-tone or color illustrations must be clear when printed in black and white. Line drawings must be made in black ink on white paper.
· Do NOT include headers and footers. The page numbers, session numbers and conference identification will be inserted automatically in a post processing step, at the time of printing the Proceedings.
· Apart from the paper in PDF format, authors can upload multimedia files to illustrate their submission. Multimedia files can be used to include materials such as sound files or movies. The proceedings CD (DVD) will NOT contain readers or players, so only widely accepted file formats should be used, such as MPEG, Windows WAVE PCM (.wav) or Windows Media Video (.wmv), using only standard codecs to maximize compatibility. Authors must ensure that they have sufficient author rights to the material that they submit for publication. Archives (RAR, ZIP or ARJ format) are allowed. The archives will be unpacked on the CD (DVD), so that authors can refer to the file name of the multimedia illustration from within their paper. The submitted files will be accessible from the abstract card on the CD (DVD) and via a bookmark in the manuscript. We advise to use SHORT but meaningful file names. The total unzipped size of the multimedia files should be reasonable. It is recommended that they do not exceed 32 Megabytes.
· Although no copyright forms are required, the authors must agree that their contribution, when accepted, will be archived by the Organizing Committee.
· Authors must proofread their manuscripts before submission and they must proofread the exact files which they submit.
POSTERS AND PRESENTATIONS
Only electronic presentations are accepted. PowerPoint presentations can be supplied on CD, DVD, FD or USB Flash drives. Designated poster space will be wooden or felt boards. The space allotted to one speaker will measure 100 cm (width) x 122 cm (height). Posters will be attached to the boards using pushpins. Pins will be provided. Thanks for following all of these instructions carefully! If you have any questions or comments concerning the submission, please don't hesitate to contact the conference organizers. Please address all technical issues or questions regarding paper submission or presentation to our technical assistant Nikolay Bobrov.

CFP IEEE ASRU 2007

Automatic Speech Recognition and Understanding Workshop
The Westin Miyako Kyoto, Japan
December 9 -13, 2007
Conference website
The tenth biannual IEEE workshop on Automatic Speech Recognition and Understanding (ASRU) cooperated by ISCA will be held during December 9-13, 2007. The ASRU workshops have a tradition of bringing together researchers from academia and industry in an intimate and collegial setting to discuss problems of common interest in automatic speech recognition and understanding.
WORKSHOP TOPICS
Papers in all areas of human language technology are encouraged to be submitted, with emphasis placed on:
- automatic speech recognition and understanding technology
- speech to text systems
- spoken dialog systems
- multilingual language processing
- robustness in ASR
- spoken document retrieval
- speech-to-speech translation
- spontaneous speech processing
- speech summarization,
- new applications of ASR.
SUBMISSIONS FOR THE TECHNICAL PROGRAM
The workshop program will consist of invited lectures, oral and poster presentations, and panel discussions. Prospective authors are invited to submit full-length, 4-6 page papers, including figures and references, to the ASRU 2007 website. All papers will be handled and reviewed electronically. The website will provide you with further details. There is also a demonstration session, which has become another highlight of the ASRU workshop. Demonstration proposals will be handled separately. Please note that the submission dates for papers are strict deadlines.
IMPORTANT DATES
Paper submission deadline July 16, 2007
Paper acceptance/rejection notification September 3, 2007
Demonstration proposal deadline September 24, 2007
Workshop advance registration deadline October 15, 2007
Workshop December 9-13, 2007
REGISTRATION AND INFORMATION
Registration will be handled via the ASRU 2007 website .
ORGANIZING COMMITTEE
General Chairs:
Sadaoki Furui (Tokyo Inst. Tech.)
Tatsuya Kawahara (Kyoto Univ.)
Technical Chairs:
Jean-Claude Junqua (Panasonic)
Helen Meng (Chinese Univ. Hong Kong)
Satoshi Nakamura (ATR)
Publication Chair:
Timothy Hazen, MIT, USA
Publicity C