The ISCA Board is very pleased to announce that Interspeech 2011 will take place in Florence, Italy, August 27-31, 2011. The Italian bid was chosen among three excellent bids, showing the interest of the speech science and technology community in Interspeech events.  The organizing committe includes Piero Cosi (General Chair), Renato di Mori (General Co-Chair), Claudia Manfredi (Local Chair), Roberto Pieraccini (Technical Program Chair), Maurizio Omologo (Tutorials), Giuseppe Riccardi (Plenary Sessions), among many other well know Italian researchers.  The conference will take place in Palazzo dei Congressi. For more information about this wonderful city, click on www.interspeech2011.org

The ISCA board is always interested in improving its activities and the membership services it provides. To help us with this, could you please send us your ideas/comments/suggestions/impressions? We would be most grateful if you could take a moment to complete the form on the ISCA website : http://www.isca-speech.org/index.php and send us your feedback.

Your message will be sent to the ISCA secretariat : secretariat@isca-speech.org

Please enter ideas/comments/suggestions/impressions you may have on any new (or old) activities and membership services.

Please note: you can send us your comments anonymously, if you so wish.

Eva Hajicova - Membership Services 

 Emmanuelle Foxonet - ISCA Secretariat  for the ISCA board 

INTERSPEECH 2008 incorporating SST 08

September 22-26, 2008

Brisbane Convention & Exhibition Centre

Brisbane, Australia

http://www.interspeech2008.org/

An invitation to INTERSPEECH 2008 and Australia – Registration now open 

 Interspeech is the world's largest and most comprehensive conference on Speech

Science and Speech Technology. We invite original papers in any related area,

including (but not limited to):

Human Speech Production, Perception and Communication;

Speech and Language Technology;

Spoken Language Systems;

Applications, Resources, Standardisation and Evaluation

Important Dates

Paper Submission: Monday, 7 April 2008, 3pm GMT

Notification of Acceptance/Rejection: Monday, 16 June 2008, 3pm GMT

Early Registration Deadline: 7 July 2008, 3pm GMT

Tutorial Day: Monday, 22 September 2008

Main conference: 23-26 September 2008

You will need a VISA

All INTERSPEECH international delegates (with the sole exception of citizens of New Zealand travelling on a New Zealand passport) will need a visa to enter Australia and this must be applied for in advance. Delegates from many countries can apply for an electronic visa, called an ETA (Electronic Travel Authority) for a modest fee of $20 and your application is likely to be approved instantly.  Currently citizens of the following countries can apply electronically at:  www.eta.immi.gov.au

Andorra, Austria , Belgium , Brunei , Canada , Denmark ,France, Finland, Germany , Greece, Hong Kong (SAR), Iceland, Ireland ,  Italy, Japan, Liechtenstein , Luxembourg , Malaysia , Malta, Monaco, Netherlands ,Norway , Portugal , San Marino , Singapore , South Korea, Spain,  Sweden , Switzerland,    Taiwan * UK **,  USA,  Vatican City. 

 If your country is not listed above, you will not be able to apply using the ETA system.  You will need to apply for a visa (a 456 Business Short Stay visa) at an Australian overseas office – for a list of offices go to: http://www.immi.gov.au/contacts/overseas/index.htm).  It is likely that you will be required to provide an INTERSPEECH conference registration confirmation letter with your visa application.  If you do not have one of these already, please write to megan@ccm.com.au and ask for a letter t be sent. The Australian Government’s Department of Immigration and Citizenship website provides full details on visas – please visit:  http://www.immi.gov.au/.

INTERSPEECH is being held in September, which is spring time in Australia! It’s a great time to visit sunny Brisbane and all of the attractions for which Australia is famous:  the Great Barrier Reef, the Gold and Sunshine Coasts, the Outback – they are all at their very best during spring.

See the INTERSPECH 2008 website   www.interspeech2008.org  Information about Brisbane and Australia and links to useful resources.

Start your travels to Australia right here: www.australia.com

BOOK YOUR INTERSPEECH ACCOMMODATION

INTERSPEECH 2008 has negotiated special discount accommodation rates at a variety of hotels and apartment hotels in Brisbane and you can make your booking when you complete the INTERSPEECH registration form.  Be warned, September is a busy time of year and we urge you to book your accommodation as soon as possible.

For a list of accommodation options, please go to the “Accommodation” sidebar on the INTERSPEECH 2008 website:  www.interspeech2008.org .

Chairman: Denis Burnham, MARCS, University of West Sydney. 

Brighton, UK,
Conference Website
Chairman: Prof. Roger Moore, University of Sheffield.

Chiba, Japan
Conference Website
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."

Call for Papers:
Second IEEE Spoken Language Technology Workshop
Goa, India
December 15-18, 2008

The Second IEEE Spoken Language Technology (SLT) workshop will be held from December 15 to December 18, 2008 in Goa, India. The goal of this workshop is to bring both the speech processing and natural language processing communities together to share and present recent advances in various areas of spoken language technology, with the expectation that such a confluence of the researchers from both communities will foster new ideas, collaborations and new research directions in this area. The SLT 2008 workshop is endorsed by both ISCA and ACL organizations and eligible participants can apply for ISCA grants (http://www.isca-speech.org/grants.html).

Spoken language technology is a vibrant research area, with the potential for significant impact on government and industrial applications especially with the diversity and challenges offered by the multilingual business climates of today's world.

The workshop solicits papers on all aspects of spoken language technology:

 o Spoken language understanding
 o Spoken document summarization
 o Machine translation for speech
 o Spoken dialog systems
 o Spoken language generation
 o Spoken document retrieval
 o Human computer Interactions (HCI)
 o Speech data mining
 o Information extraction from speech
 o Question answering from speech
 o Multimodal processing
 o Spoken language based assistive technologies
 o Spoken language systems and applications
 o Spoken language databases and standards

In addition, this year's workshop will feature three special sessions:

 1) Challenges in Asian spoken language processing with special emphasis on Indian languages
 2) Mining human-human conversations: A resource for building efficient human-machine dialogs
 3) Spoken Language on the go: Challenges and Opportunities for spoken language processing on mobile devices

Submissions for the Technical Program
-------------------------------------
The workshop program will consist of tutorials, oral and poster presentations, and panel discussions. Attendance will be limited with priority for those who will present technical papers; registration is required of at least one author for each paper. Submissions are encouraged on any of the topics listed above. The style guide, templates, and submission form will follow the IEEE ICASSP style. Three members of the Scientific Committee will review each paper. The workshop proceedings will be published on a CD-ROM.

Important Dates
---------------
*Camera-ready paper submission deadline: August 8, 2008
Hotel Reservation and Workshop registration opens: August 8, 2008
Paper Acceptance / Rejection: September 15, 2008
Hotel Reservation and Early Registration closes: October 5, 2008
Workshop: December 15-18, 2008*

For more information visit the SLT 2008 website http://slt2008.org or contact the organizing committee at info@slt2008.org <mailto:info@slt2008.org> if you have any questions.

 This section shows recent books whose titles been have communicated by the authors or editors.

Also some advertisement for recent books in speech are included.

Book presentation is written by the authors and not by this newsletter editor or any  voluntary reviewer.

For Membership Year 2008, we instituted our first-ever discount program to encourage members to join early in the year and to keep their membership current.  Many member organizations participated in the program, but several eligible organizations did not take advantage of the discount.  We are offering those organizations a credit for overpayment of membership fees.  This credit can be applied towards membership or data licensing fees and can used throughout the calendar year.

If your organization is eligible for a credit, you should have already received notification by email.  If you believe you are eligible, but have not been contacted by the LDC, please contact our Membership group at ldc@ldc.upenn.edu or +1 215 573 1275.

Membership Mailbag - What is a Reduced Licensing Fee?

The LDC Membership Office responds to over 4000 emailed queries a year, and, over time, we've noticed that some questions tend to crop up with regularity.  To address the questions that you, our data users, have asked, we'd like to continue our Membership Mailbag series of newsletter articles.  This month, we'll elaborate on the 'What is a Reduced Licensing Fee?’ question on our Members FAQ.

One of the benefits of being a member in the current Membership Year (MY 2008) is the ability to license LDC data from closed MYs at a discount,  what is termed the 'Reduced Licensing Fee'. This fee is listed on each catalog page and is generally 50% of the Non-member Fee.  The Reduced Licensing Fee does not apply to corpora from the previous MY (MY 2007), since that year is still open for joining.  One common misinterpretation is that the Reduced Licensing Fee is discounted pricing for non-commercial organizations.  However, non-members, including educational and not-for-profit organizations, must pay the full Non-member Fee for any data they license.

A few of our corpora are only available to members of the LDC.  Organizations that have joined MY 2008 can still license such ‘Members Only’ corpora even if they did not join the MY(s) that apply to the corpora.  Since Members Only corpora do not have corresponding Non-member Fees, their Reduced Licensing Fees are calculated based on corpus size:  US$200 per DVD-ROM, US$150 per CD-ROM, and US$100 per web download.  This pricing information is listed at the bottom of each Members Only catalog entry.

Got a question?  About LDC data?  Forward it to ldc@ldc.upenn.edu.  The answer may appear in a future Membership Mailbag article.

New Browsing Utility for An English Dictionary of the Tamil Verb

A new browsing utility is now available for An English Dictionary of the Tamil Verb (LDC2008L01).  It allow users to interactively search the XML version of the dictionary, presenting the entries in a user friendly format.  Users can conduct searches in English or Tamil and can browse the dictionary as well. This tool works only on Internet Explorer with ActiveX Objects activated.

The utility can be downloaded at no cost from the LDC catalog page for An English Dictionary of the Tamil Verb under 'Updates'.

New Publications

(1) GALE Phase 1 Arabic Broadcast News Parallel Text - Part 2 is the second part of the three-part GALE Phase 1 Arabic Broadcast News Parallel Text, which, along with other corpora, was used as training data in year 1 (Phase 1) of the DARPA-funded GALE program. The corpus contains transcripts and English translations of 10.7 hours of Arabic broadcast news programming selected from various sources. This corpus does not contain the audio files from which the transcripts and translations were generated.

The Arabic broadcast news recordings were selected from four sources and four different programs.   A manual selection procedure was used to choose data appropriate for the GALE program, namely, news and conversation programs focusing on current events. Stories on topics such as sports, entertainment news, and stock market reports were excluded from the data set.  Manual sentence units/segments (SU) annotation was also performed on a subset of files following LDC's Quick Rich Transcription specification. Three types of end of sentence SU were identified: statement SU, question SU, and incomplete SU.

2008 Subscription Members will automatically receive two copies of this corpus on disc. 2008 Standard Members may request a copy as part of their 16 free membership corpora. Nonmembers may license this data for US$1500.

*

(2) The 2005 NIST Language Recognition Evaluation corpus contains the evaluation data, portions of the training data, the evaluation plan, answer keys and scoring script for the 2005 NIST (National Institute of Standards and Technology) Language Recognition Evaluation (LRE). The goal of the LRE is to establish the baseline of current performance capability for language recognition of conversational telephone speech and to lay the groundwork for further research efforts in the field. NIST conducted two previous evaluations in 1996 and 2003. For the 2005 NIST LRE, the emphasis was on research directed toward a general base of technology to be ported to various language recognition tasks with minimum effort and the development of the ability to make more difficult discriminations between similar languages and dialects of the same language.

The task evaluated was the detection of a given target language or dialect. From a test segment of speech and a target language or dialect, the system to be evaluated determined whether the speech was from the target language or dialect. The evaluation consisted of speech from the following languages and dialects:

• English (American)
• English (Indian)
• Hindi
• Japanese
• Korean
• Mandarin (Mainland)
• Mandarin (Taiwan)
• Spanish (Mexican)
• Tamil

The 2005 NIST Language Recognition Evaluation Plan, which includes a description of the evaluation tasks, is included with this release. Further information regarding this evaluation is also available at the NIST Language Recognition Evaluation website.

Each speech file is one side of a telephone conversation . There are 11,106 speech files in sphere (.sph) format for a total of 44.2 hours of speech. The speech data was compiled from LDC's CALLFRIEND corpora and from data collected by Oregon Health and Science University.

Each test segment was prepared using an automatic speech activity detection algorithm to identify areas and durations of speech. Segments were chosen to contain a specified approximate duration of actual speech. The test segments contain three nominal durations of speech: 3 seconds, 10 seconds, and 30 seconds. Performance was evaluated separately for test segments of each duration. Auxiliary information was included in the SPHERE headers to document the source file, start time, and duration of all excerpts that were used to construct the segment.  2005 NIST Language Recognition Evaluation is distributed on one DVD-ROM.

Music and speech share numerous aspects (language, structural, acoustics, cognitive), as long in their production, that in their representation and their perception. This list has for object to warn its users, various events dealing with the study of the links between music and speech. It thus intends to connect several communities, their allowing each to take advantage of a stimulating interaction.

As a member of the speech or music community, you are invited to
subscribe to musicspeech group. The group will be moderated and
maintained by IRCAM.

Group details:
* Name: musicspeech
* Home page: http://listes.ircam.fr/wws/info/musicspeech
* Email address: musicspeech@ircam.fr

Greg Beller, IRCAM,
moderator, musicspeech list

AT&T - Labs Research is seeking exceptional candidates for Research Staff positions. AT&T is the premiere broadband, IP, entertainment, and wireless communications company in the U.S. and one of the largest in the world. Our researchers are dedicated to solving real problems in speech and language processing, and are involved in inventing, creating and deploying innovative services. We also explore fundamental research problems in these areas. Outstanding Ph.D.-level candidates at all levels of experience are encouraged to apply. Candidates must demonstrate excellence in research, a collaborative spirit and strong communication and software skills. Areas of particular interest are               

• Large-vocabulary automatic speech recognition
• Acoustic and language modeling
• Robust speech recognition
• Signal processing
• Speaker recognition
• Speech data mining
• Natural language understanding and dialog
• Text and web mining
• Voice and multimodal search

AT&T Companies are Equal Opportunity Employers. All qualified candidates will receive full and fair consideration for employment. More information and application instructions are available on our website at http://www.research.att.com/. Click on "Join us". For more information, contact Mazin Gilbert (mazin at research dot att dot com).

Nagoya Institute of Technology is seeking a researcher for a

post-doctoral position in a new European Commission-funded project

EMIME ("Efficient multilingual interaction in mobile environment")

involving Nagoya Institute of Technology and other five European

partners, starting in March 2008 (see the project summary below).

The earliest starting date of the position is March 2007. The initial

duration of the contract will be one year, with a possibility for

prolongation (year-by-year basis, maximum of three years). The

position provides opportunities to collaborate with other researchers

in a variety of national and international projects. The competitive

salary is calculated according to qualifications based on NIT scales.

The candidate should have a strong background in speech signal

processing and some experience with speech synthesis and recognition.

Desired skills include familiarity with latest spectrum of technology

including HTK, HTS, and Festival at the source code level.

For more information, please contact Keiichi Tokuda

(http://www.sp.nitech.ac.jp/~tokuda/).

About us

Nagoya Institute of Technology (NIT), founded on 1905, is situated in

the world-quality manufacturing area of Central Japan (about one hour

and 40 minetes from Tokyo, and 36 minites from Kyoto by Shinkansen).

NIT is a highest-level educational institution of technology and is

one of the leaders of such institutions in Japan. EMIME will be

carried at the Speech Processing Laboratory (SPL) in the Department of

Computer Science and Engineering of NIT. SPL is known for its

outstanding, continuous contribution of developing high-performance,

high-quality opensource software: the HMM-based Speech Synthesis

System "HTS" (http://hts.sp.nitech.ac.jp/), the large vocabulary

continuous speech recognition engine "Julius"

(http://julius.sourceforge.jp/), and the Speech Signal Processing

Toolkit "SPTK" (http://sp-tk.sourceforge.net/). The laboratory is

involved in numerous national and international collaborative

projects. SPL also has close partnerships with many industrial

companies, in order to transfer its research into commercial

applications, including Toyota, Nissan, Panasonic, Brother Inc.,

Funai, Asahi-Kasei, ATR.

Project summary of EMIME

The EMIME project will help to overcome the language barrier by

developing a mobile device that performs personalized speech-to-speech

translation, such that a user's spoken input in one language is used

to produce spoken output in another language, while continuing to

sound like the user's voice. Personalization of systems for

cross-lingual spoken communication is an important, but little

explored, topic. It is essential for providing more natural

interaction and making the computing device a less obtrusive element

when assisting human-human interactions.

We will build on recent developments in speech synthesis using hidden

Markov models, which is the same technology used for automatic speech

recognition. Using a common statistical modeling framework for

automatic speech recognition and speech synthesis will enable the use

of common techniques for adaptation and multilinguality.

Significant progress will be made towards a unified approach for

speech recognition and speech synthesis: this is a very powerful

concept, and will open up many new areas of research. In this

project, we will explore the use of speaker adaptation across

languages so that, by performing automatic speech recognition, we can

learn the characteristics of an individual speaker, and then use those

characteristics when producing output speech in another language.

Our objectives are to:

1. Personalize speech processing systems by learning individual

characteristics of a user's speech and reproducing them in

synthesized speech.

2. Introduce a cross-lingual capability such that personal

characteristics can be reproduced in a second language not spoken

by the user.

3. Develop and better understand the mathematical and theoretical

relationship between speech recognition and synthesis.

4. Eliminate the need for human intervention in the process of

cross-lingual personalization.

5. Evaluate our research against state-of-the art techniques and in a

practical mobile application.

Training Generative Bayesian Networks with Missing Data

Bayesian Methods for Generalization in Reinforcement Learning

Title: Solutions for non-linear acoustic echo.

Background:

Acoustic echo is an annoying disturbance due to the sound feedback between the loudspeaker and the microphone of terminals. Acoustic echo canceller and residual echo cancellation are widely used to reduce the echo signal. The performance of existing echo reduction systems strongly relies on the assumption that the echo path between transducers is linear. However, today’s competitive audio consumer market may favour sacrificing linear performance for the integration of low cost analogue components. The assumption of linearity is not hold anymore, due to the nonlinear distortions introduced by the loudspeakers and the small housing where transducers are placed.

Task:

The PhD thesis will lead research in the field of non-linear system applied for acoustic echo reduction. The foreseen tasks deal first with proper modelling of mobile phone transducers presenting non-linearity, to get a better understanding in which environment echo reduction works. Using this model as a basis, study of performance of linear system will permit to get a good understanding on the problems brought by non-linearity. In a further step, the PhD student will develop and test non-linear algorithms coping with echo cancellation in non-linear environment.

About the Company:

Sophia-Antipolis site is one of the main Infineon Technologies research and development centers worldwide. Located in the high-tech valley of Sophia-Antipolis, near Nice in the south of France, a team of 140 experienced research and development engineers specialized in Mobile Solutions, Embedded SRAM, and Design-Flow Software. The PhD will take place within the Mobile Solution group which is responsible for specifying and designing baseband integrated circuits for cellular phones. The team is specialized in innovative algorithm development, especially in audio, system specification and validation, circuit design and embedded software. Its work makes a significant contribution to the Infineon Technologies wireless chipset portfolio.

Required skills:

-        Master degree

-        Strong background in signal processing.

-        Background in speech signal or non-linear system processing is a plus.

-        Programming: Matlab, C.

-        Knowledge in C-fixed point / DSP implementation is a plus.

-        Language: English

Length of the PhD: 3 years

Place: Infineon Technologies France, Sophia-Antipolis

Contact:

Christophe Beaugeant

Phone: +33 (0)4 92 38 36 30

E-mail : christophe.beaugeant@infineon.com

At the Institut für Theoretische Informatik, Lehrstuhl Prof. Waibel Universität Karlsruhe (TH) a

Ph.D. position

in the field of

Software System Integration of Automatic Speech Recognition and Machine Translation for Speech based Multimedia Indexing

has to be filled immediately with a salary according to TV-L, E13.

The responsibilities include integration, fusion and development of core technologies in the area of automatic speech recognition, simultaneous machine translation, in the context of speech based indexing of multimedia documents within application targeted research projects in the area of multimodal Human-machine interaction.  Set in a framework of internationally and industry funded research programs, the successful candidate is expected to contribute to showcases for state-of-the art of modern recognition and translation systems.

We are an internationally renowned research group with an excellent infrastructure. Examples of our projects for improving Human-Machine and Human-to-Human interaction are: JANUS - one of the first speech translation systems proposed, simultaneous translation of lectures, portable speech translators, meeting browser and lecture tracker.

Within the framework of the International Center for Advanced Communication Technology (interACT), our institute operates in two locations, Universität Karlsruhe (TH), Germany and at Carnegie Mellon University, Pittsburgh, USA.  International joint and collaborative research at and between our centers is common and encouraged, and offers great international exposure and activity. 

Applicants are expected to have:

• an excellent university degree (M.S, Diploma or Ph.D.) in Computer Science, Electrical Engineering, Mathematics, or related fields
• excellent programming skills 
• advanced knowledge in at least one of the fields of Machine Learning, Pattern Recognition, Statistics, or System Integration

For candidates with Bachelor or Master’s degrees, the position offers the opportunity to work toward a Ph.D. degree.

In line with the university's policy of equal opportunities, applications from qualified women are particularly encouraged. Handicapped applicants will be preferred in case of the same qualification.

Questions may be directed to: Sebastian Stüker, Tel. +49 721 608 6284, E-Mail: stueker@ira.uka.de,  http://isl.ira.uka.de

The application should be sent to Professor Waibel, Institut für Theoretische Informatik, Universität Karlsruhe (TH), Adenauerring 4, 76131 Karlsruhe, Germany

----------------------------------------------------------------------------------------------------------------------------------

Ph.D. position

in the field of

Multimodal Dialog Systems

Is to be filled immediately with a salary according to TV-L, E13.

The responsibilities include basic research in the area of multimodal dialog systems, especially multimodal human-robot interaction and learning robots, within application targeted research projects in the area of multimodal Human-machine interaction.  Set in a framework of internationally and industry funded research programs, the successful candidate(s) are expected to contribute to the state-of-the art of modern spoken dialog systems, improving natural interaction with robots.

We are an internationally renowned research group with an excellent infrastructure. Current research projects for improving Human-Machine and Human-to-Human interaction are focus on dialog management for Human-Robot interaction.

Within the framework of the International Center for Advanced Communication Technology (interACT), our institute operates in two locations, Universität Karlsruhe (TH), Germany and at Carnegie Mellon University, Pittsburgh, USA.  International joint and collaborative research at and between our centers is common and encouraged, and offers great international exposure and activity. 

Applicants are expected to have:

• an excellent university degree (M.S, Diploma or Ph.D.) in Computer Science, Computational Linguistics, or related fields
• excellent programming skills 
• advanced knowledge in at least one of the fields of Speech and Language Processing, Pattern Recognition, or Machine Learning

For candidates with Bachelor or Master’s degrees, the position offers the opportunity to work toward a Ph.D. degree.

In line with the university's policy of equal opportunities, applications from qualified women are particularly encouraged. Handicapped applicants will be preferred in case of the same qualification.

Questions may be directed to: Hartwig Holzapfel, Tel. +49 721 608 4057, E-Mail: hartwig@ira.uka.de,  http://isl.ira.uka.de

The application should be sent to Professor Waibel, Institut für Theoretische Informatik, Universität Karlsruhe (TH), Adenauerring 4, 76131 Karlsruhe, Germany

Job opening at TFH Berlin University of Applied Sciences, Department of Computer Sciences and Media, Germany: Post-graduate position (part-time) for a computer scientist or engineer with a background in ASR and/or TTS in a three-year project in Computer-Aided Language Learning funded by the German Ministry of Education and Research. Start: 1 July 2008. The task will be the development and evaluation of a software system for teaching Mandarin pronunciation to Germans, as well as administrative duties with the funding body. Knowledge of E-Learning applications, German and/or Mandarin are welcome, good English skills mandatory. Candidates will have the opportunity to pursue a PhD degree and should be preferably EU citizens. The position is paid according to BAT 2a/2 (German pay scale for federal employees), about €28.000/year depending on age and marital status. Please direct further enquiries to Prof. Dr. Hansjörg Mixdorff at mixdorff@tfh-berlin.de. 

Offre d’Allocation de Recherche – Rentrée Universitaire 2008

Les cartes sensorimotrices de la parole: Corrélats neuroanatomiques des systèmes de perception et de production des voyelles et consonnes du Français.

Marc Sato, Chargé de Recherche CNRS

Jean-Luc Schwartz, Directeur de Recherche CNRS

GIPSA-Lab, UMR CNRS 5216, Département Parole et Cognition, Equipe "Perception, Multimodalité, Développement", Grenoble France (http://gipsa-lab.inpg.fr).

Pour la rentrée universitaire 2008, nous proposons une Allocation Doctorale de Recherche dans le cadre de l’Ecole Doctorale Ingénierie pour la santé, la Cognition et l’Environnement, habilitée par les universités Pierre Mendès France, Joseph Fourier et l’Institut National Polytechnique de Grenoble (EDISCE – ED216, http://www-sante.ujf-grenoble.fr/edisce/).

Dans le cadre théorique d’un possible couplage fonctionnel entre systèmes de perception et de production de la parole, ce projet de recherche a pour but de tester l’existence de connectivités dynamiques fonctionnelles spécifiques entre régions sensorielles et motrices lors de la perception et de la production des voyelles et consonnes du Français. Dans ce but, une série d’expériences en imagerie par résonance magnétique fonctionnelle (IRMf) et en électro-encéphalographie (EEG) devra permettre une description spatiale et temporelle précise des activités cérébrales impliquées lors de la production et la perception des phonèmes du français ainsi que de la connectivité dynamique entre ces régions. Les expériences menées devraient permettre une compréhension approfondie des processus d’analyse et de construction des représentations verbales par la mise en évidence d’une co-structuration et dépendance des régions sensorielles et motrices.

Outre une recherche bibliographique approfondie du corpus de la littérature, multidisciplinaire, dans les domaines de la phonétique et de la phonologie, de la neuropsycholinguistique et des neurosciences cognitives, ce projet comprendra l’élaboration de protocoles expérimentaux en IRMf et EEG, la passation des sujets et le recueil des données, enfin l’analyse statistique des données et leur interprétation.

Le(a) candidat(e) aura de préférence un M2R en neurosciences, sciences cognitives, psychologie cognitive ou neuropsychologie. Le candidat devra être familier avec l’expérimentation comportementale classique ainsi que les tests et analyses statistiques appliqués en psychologie cognitive. La pratique de l’anglais et une première expérience avec les techniques d’IRMf et/ou d’EEG est souhaitable.

Ce travail de thèse sera inscrit dans le contexte d’un projet de recherche grenoblois portant sur la mise en œuvre de nouvelles techniques d’analyses et de modélisation non-linéaire de mesure de la connectivité cérébrale. Il s’appuiera également sur des collaborations nationales et internationales, notamment avec le Centre de Recherche sur le Langage, l’Esprit et le Cerveau de l’Université McGill (Canada). Le financement est pour une période de trois ans, à compter d’Octobre 2008.

Contact pour cette annonce:

Marc Sato

GIPSA-Lab, UMR CNRS 5216

Université Stendhal

BP 25 - 38040 Grenoble cedex 9

Tel: (+33) (0)476 827 784

Fax: (+33) (0)476 824 335

E-mail: marc.sato [at] gipsa-lab.inpg.fr

La date limite pour envoyer un CV détaillé est fixée au 18 juin. Dès que possible, envoyez également les notes finales et classement au M2R, et éventuellement une lettre de recommandation.

Thèse : Caractérisation et l'identification automatique de dialectes These DeadLine: 10/06/2008 jerome.farinas@irit.fr http://www.irit.fr/recherches/SAMOVA/these-caracterisation-et-lidentification-automatique-de-dialectes.html Description du sujet Le recherche en traitement automatique de la parole s'intéresse de plus en plus au traitement de grandes collections de données, dans des conditions de parole spontanée et conversationnelle. Les performances sont dépendantes de toutes les variabilités de la parole. Une de ces variabilités concerne l'appartenance dialectale du locuteur, qui induit de la variabilité tant au niveau de la prononciation phonétique, mais également au niveau de la morphologie des mots et de la prosodie. Nous proposons de réaliser un sujet de recherche sur la caractérisation automatique dialectale des locuteurs, en vue de diriger l'adaptation des systèmes de reconnaissance de la parole : la sélection de modèles acoustiques et prosodiques adaptées permettront d'améliorer des performances dans des conditions de reconnaissance indépendante du locuteur. La réalisation d'un tel système devra s'appuyer sur les avancées récentes en identification de la langue au niveau de la modélisation acoustique par exp! loration de réseaux de phonèmes et proposer une modélisation fine basée sur la micro et macro prosodie. Les bases de données disponibles au sein du projet sur la phonologie du français contemporain (http://www.projet-pfc.net/) permettront de disposer d'un large éventail de données sur les variances de prononciation. Le système final sera évalué lors des campagnes internationales organisées par le NIST sur la vérification de la langue, qui prennent maintenant en compte les variances dialectales (mandarin, anglais, espagnol et hindi) : http://www.nist.gov/speech/tests/lre/. English version The research in automatic speech processing is increasingly concerned at the treatment of large data collections, with spontaneous and conversational speech. The variability of speech alter the general performances. The dialect of the speaker is of these variability sources, who leads in alterations in terms of both the phonetic pronunciation, but also in terms of the morphology of words and prosody. We propose to conduct a research on the automatic characterization of the dialects, in order to adapt automatic speech recognition systems: by selection of acoustic and prosodic models suited to improve performance in speaker independent recognition conditions. The realization of such a system should be based on recent advances in the identification of the language in the exploration of phonemes modeling lattices and propose a fine modelling based on micro and macroprosody. The databases available within the PFC project (http://www.projet-pfc.net/) will provide a wide range of d! ata on variances pronunciation. The final system will be evaluated during international campaigns conducted by the NIST on language verification, which now take into account the dialect identification (Mandarin, English, Spanish and Hindi): http://www.nist.gov/speech/tests/lre/. Connaissances et compétences requises * compétences en informatique (en particulier traitement automatique de la parole) * compétences en linguistique (phonologie, prosodie) Contact Un financement sera attribué aux meilleurs candidats de thèse de l'Ecole Doctorale, il faut donc nous contacter avant le 10 juin pour pouvoir participer à ce classement.

Thèse de l’école doctorale MITT, Université Paul Sabatier Toulouse III

DeadLine: 10/06/2008

Contacts: jerome.farinas@irit.fr

http://www.irit.fr/-Equipe-SAMoVA-

DESCRIPTION DU SUJET :

Le recherche en traitement automatique de la parole s'intéresse de plus en plus au traitement de grandes collections de données, dans des conditions de parole spontanée et conversationnelle. Les performances sont dépendantes de toutes les variabilités de la parole. Une de ces variabilités concerne l'appartenance dialectale du locuteur, qui induit de la variabilité tant au niveau de la prononciation phonétique, mais également au niveau de la morphologie des mots et de la prosodie. Nous proposons de réaliser un sujet de recherche sur la caractérisation automatique dialectale des locuteurs, en vue de diriger l'adaptation des systèmes de reconnaissance de la parole : la sélection de modèles acoustiques et prosodiques adaptées permettront d'améliorer des performances dans des conditions de reconnaissance indépendante du locuteur. La réalisation d'un tel système devra s'appuyer sur les avancées récentes en identification de la langue au niveau de la modélisation acoustique par exploration de réseaux de phonèmes et proposer une modélisation fine basée sur la micro et macro prosodie. Les bases de données disponibles au sein du projet sur la phonologie du français contemporain (http://www.projet-pfc.net/) permettront de disposer d'un large éventail de données sur les variances de prononciation. Le système final sera évalué lors des campagnes internationales organisées par le NIST sur la vérification de la langue, qui prennent maintenant en compte les variances dialectales (mandarin, anglais, espagnol et hindi) : http://www.nist.gov/speech/tests/lre/.

ENGLISH VERSION:

The research in automatic speech processing is increasingly concerned at the treatment of large data collections, with spontaneous and conversational speech. The variability of speech alter the general performances. The dialect of the speaker is of these variability sources, who leads in alterations in terms of both the phonetic pronunciation, but also in terms of the morphology of words and prosody. We propose to conduct a research on the automatic characterization of the dialects, in order to adapt automatic speech recognition systems: by selection of acoustic and prosodic models suited to improve performance in speaker independent recognition conditions. The realization of such a system should be based on recent advances in the identification of the language in the exploration of phonemes modeling lattices and propose a fine modelling based on micro and macroprosody. The databases available within the PFC project (http://www.projet-pfc.net/) will provide a wide range of data on variances pronunciation. The final system will be evaluated during international campaigns conducted by the NIST on language verification, which now take into account the dialect identification (Mandarin, English, Spanish and Hindi): http://www.nist.gov/speech/tests/lre/.

CONNAISSANCES ET COMPETENCES REQUISES

• compétences en informatique (en particulier traitement automatique de la parole)

• compétences en linguistique (phonologie, prosodie)

Cambridge University: Research Position in Speech Synthesis and Recognition / Machine Translation


A position exists for a Research Associate to work on the EMIME ("Efficient multilingual interaction in mobile environment") project. This project is funded by the European Commission within the FP7 programme. The project aims to develop a mobile device that performs personalized speech-to-speech translation such that a user's spoken input in one language is used to produce spoken output in another language, while continuing to sound like the user's voice. We will build on recent developments in speech synthesis using hidden Markov models, which is the same technology used for automatic speech recognition. Using a common statistical modelling framework for automatic speech recognition and speech synthesis will enable the use of common techniques for adaptation and multilinguality. The project objectives are to

1. Personalise speech processing systems by learning individual characteristics of a user's speech and reproducing them in synthesised speech.
2. Introduce a cross-lingual capability such that personal characteristics can be reproduced in a second language not spoken by the user.
3. Develop and better understand the mathematical and theoretical relationship between speech recognition and synthesis.
4. Eliminate the need for human intervention in the process of cross-lingual personalisation.
5. Evaluate our research against state-of-the art techniques and in a practical mobile application.
See the EMIME website for more information: http://www.emime.org/

This is an opportunity to work in a research group with a world-leading reputation in speech recognition and statistical machine translation research. There are excellent opportunities for publications, travel and conference visits. The group has outstanding research facilities. For suitably qualified candidates there may also be the chance to contribute to the MPhil in Computer Speech, Text and Internet Technology (http://mi.eng.cam.ac.uk/cstit/).

The successful candidate must have a very good first degree in a relevant discipline and preferably have a higher degree as well as experience in acoustic modeling for speech synthesis and/or recognition. Expertise in one or more of the following technical areas is also a distinct advantage:
- speech recognition with the HTK toolkit (http://htk.eng.cam.ac.uk)
- speech synthesis with the HTS HMM-based Speech Synthesis System (http://hts.sp.nitech.ac.jp)
- weighted finite state transducers for speech and language processing
The project focus is acoustic modeling but experience in statistical machine translation is also an advantage.

The cover sheet for applications, PD18 is available from http://www.admin.cam.ac.uk/offices/personnel/forms/pd18/ Part I and Part III only, should be sent, with a letter and CV to Dr Bill Byrne, Department of Engineering, Trumpington Street, Cambridge, CB2 1PZ, (Fax +44 01223 332662, email wjb31@cam.ac.uk).
Quote Reference: NA03547, Closing Date: 30 June 2008

The University values diversity and is committed to equality of opportunity.

Head of NLP :

 We are now looking for a very experienced Computational Linguist

 to lead our efforts in the natural language processing area. This

 is a hugely challenging, but also a very rewarding role; the

 opportunities for applying linguistic techniques are virtually

 limitless and even small improvements in the algorithms for

 detecting and correcting potential conversion errors translate

 into serious cost savings for the company. This is a senior position

 leading a team and having the autonomy to build a strategic way

 forward for this department.

 Experience Needed:

 Grammars and parsing for spontaneous speech.

 Statistical methods.

 At least basic programming ability (shell scripts, Perl, awk).

 Spell-checkers, grammar checkers, auto correcting tools and predictive typing.

 Experience with Automatic Speech Recognition technology.

 Probabilistic Language Modelling

 Phonetics

 Multi-lingual

info@voxid.co.uk 

Nuance is the leading provider of speech and imaging solutions for businesses and consumers around the world.  Every day, millions of users and thousands of businesses experience Nuance by calling directory assistance, requesting account information, dictating patient records, telling a navigation system their destination, controlling their mobile phone or digitally reproducing documents that can be shared and searched.  With more than 2000 employees worldwide, we are committed to make the user experience more enjoyable by transforming the way people interact with information and how they create, share and use documents. Making each of those experiences productive and compelling is what Nuance is about. To strengthen our international team we are currently looking for a

Junior Research Engineer for Embedded Automatic Speech Recognition

Work Environment

·          You will work in the Embedded ASR research and production team in Merelbeke, Belgium or Aachen, Germany, working with state-of -he-art speech technology, tools and runtime software. Both Gent and Aachen are nice, historical European university cities.

·          You will work in an international company and cooperate with people and research teams on various locations across the globe. You may occasionally be asked to travel.

·          You will work  with our natural language understanding and dialogue research teams as well support our professional services teams.

·          You will work on the development of cutting edge speech recognition products for automotive platforms and mobile devices. You will help the engine cope with multi-lingual speech in various noise conditions, and this while respecting strong limitations on the usage of memory and processing power.

Key Responsibilities

·          Design, implementation, evaluation, optimization and testing of new algorithms and tools, with a strong focus on speech signal processing and acoustic modeling in adverse, noisy environments.

·          Activities are targeted at the creation of commercial products for resource limited platforms.

·          Focus on creating efficient production and development processes to bring the technology to marketable products in a wide range of languages.

·          Occasional application of the developed algorithms and tools for producing systems for a specific language.

·          Specification and follow-up of projects to make the system work with third party components, such as beam formers, echo cancellers or content data providers.

Your Profile

• You have a University degree in engineering, mathematics or physics.
• A PhD degree in speech processing or equivalent relevant experience is a strong asset.
• Experience in speech recognition research, especially acoustic modeling or signal processing, is required.
• Experience in speech processing, machine learning techniques or statistical modeling is required.
• Knowledge about small platforms and experience in developing software for them is a plus.
• Strong software skills are required, especially C/C++ and a scripting language like Perl or Python in a Linux/Unix environment. Knowledge of Matlab is a plus.
• Additional background in computational linguistics is a plus.
• You are a team player, willing to take initiative, and are goal oriented.
• You have a strong desire to make things “really work” in practice, on hardware platforms with limited memory and processing power.
• You are fluent in English and at least one other language, and you can write high quality English documentation. 

Interested?

Please send your CV to Deanna Roe at deanna.roe@nuance.com. If you have any questions, please contact her at +44 207 922 5757.

We are looking forward to receiving your application!  deanna.roe@nuance.com. If you have any questions, please contact her at +44 207 922 5757.

In order to strengthen our Embedded ASR Research team, we are looking for a:

       SOFTWARE ENGINEER SPEECH DIALOGUE TOOLS

As part of our team, you will be creating solutions for voice user interfaces for embedded applications on mobile and automotive platforms.

OVERVIEW:

- You will work in Nuance's Embedded ASR (automatic speech recognition) research and development team, developing technology, tools, and run-time software to enable our customers to develop and test embedded speech applications. Together with our team of speech and language experts, you will work on natural language dialogue systems for our customers in the Automotive and Mobile sector.

- You will work on fascinating technology that has now reached the maturity to enable new generations of powerful and natural user interfaces. Your code is crucial to the research in speech and language technology that defines the state of the art in this field It is equally important for the products that you will find in the market, in speech-enabled cars, navigation devices, and cell phones.

- You will work in a large international software company that is the leading provider of speech and imaging solutions for businesses and consumers around the world. You will cooperate with people on various locations including in Europe, America and Asia. You may occasionally be asked to travel.

RESPONSIBILITIES:

- You will work on the development of tools and solutions for cutting edge speech and language understanding technologies for automotive and mobile devices.

- You will work on enhancing various aspects of our advanced natural language dialogue system, such as the layer of connected applications, the configuration setup, inter-module communication, etc.

- In particular, you will be responsible for the design, implementation, evaluation, optimization and testing, and documentation of tools such as GUI and XML applications that are used to develop, configure, and fine-tune advanced dialogue systems.

QUALIFICATIONS:

- You have a university degree in computer science, engineering, mathematics, physics, computational linguistics, or a related field.

- You have very strong software and programming skills, especially in C/C++, ideally also for embedded applications.

- You have experience with Python or other scripting languages.

- GUI programming experience is an asset.

The following skills are a plus:

- Understanding of communication protocols

- Understanding of databases

- A background in (computational) linguistics, dialogue systems, speech processing, grammars, and parsing techniques, statistics, pattern recognition, and machine learning, especially as related to natural language processing, dialogue, and representation of information

- Understanding of computational agents and related frameworks (such as OAA).

- You can work both as a team player and as goal-oriented independent software engineer.

- You can work in a multi-national team and communicate effectively with people of different cultures.

- You have a strong desire to make things really work in practice, on hardware platforms with limited memory and processing power.

- You are fluent in English and you can write high quality documentation.

- Knowledge of other languages is a plus.

CONTACT:

Please send your applications, including cover letter, CV, and related documents (maximum 5MB total for all documents, please) to

Benjamin Campued        Benjamin.Campued@nuance.com

Please make sure to document to us your excellent software engineering skills.

ABOUT US:

Nuance is the leading provider of speech and imaging solutions for businesses and consumers around the world.  Every day, millions of users and thousands of businesses experience Nuance by calling directory assistance, requesting account information, dictating patient records, telling a navigation system their destination, or digitally reproducing documents that can be shared and searched.  With more than 3500 employees worldwide, we are committed to make the user experience more enjoyable by transforming the way people interact with information and how they create, share and use documents. Making each of those experiences productive and compelling is what Nuance is about.

Nuance is the leading provider of speech and imaging solutions for businesses and consumers around the world.  Every day, millions of users and thousands of businesses experience Nuance by calling directory assistance, requesting account information, dictating patient records, telling a navigation system their destination, controlling their mobile phone or digitally reproducing documents that can be shared and searched.  With more than 2000 employees worldwide, we are committed to make the user experience more enjoyable by transforming the way people interact with information and how they create, share and use documents. Making each of those experiences productive and compelling is what Nuance is about. To strengthen our international team we are currently looking for a

Speech Scientist for Embedded Automatic Speech Recognition

Work Environment

·          You will work in the Embedded ASR research and production team in Merelbeke, Belgium or Aachen, Germany, working with state-of-the-art speech technology, tools and runtime software. Both Gent and Aachen are nice, historical European university cities.

·          You will work in an international company and cooperate with people on various locations, from the USA up to Japan. You may occasionally be asked to travel.

·          You will work on the localization and production of language variants for our cutting edge speech recognition products targeted at automotive platforms and mobile devices. You will help the engine cope with multi-lingual speech in various noise conditions.

·          Initially, you will work on the production of language variants of our acoustic models, later extending your knowledge towards production of statistical language models and natural language dialogue systems.

Key Responsibilities

·          Training of  acoustic models or statistical language models for new languages.

·          Localizing natural language dialogue systems towards a specific market.

·          Contributing to the improvement, design, implementation, evaluation, optimization and testing of new algorithms, tools and processes.

·          Supporting our professional services teams to contribute to customer project success.

·          Assisting senior team members in research tasks.

Your Profile

• You have a University degree in linguistics, engineering, mathematics or physics.
• A PhD or similar experience in a relevant field is a plus.
• Experience in acoustic modeling, NLU or statistical language modeling is recommended.
• Additional background in computational linguistics is a plus.
• Working in Windows and Linux environments comes naturally to you. Experience with computing farms and grid software is welcome.
• You are knowledgable about small, embedded platforms and requirements of software applications designed for them.
• Good software skills are required, especially scripting language like Perl or Python in a Linux/Unix environment, and knowledge of C/C++.
• Experience in speech processing or machine learning techniques is an asset.
• You are a team player, willing to take initiative, and are goal oriented.
• You have a strong sense of precision and quality in your daily job.
• You are fluent in English and you can write high quality documentation. 
• You illustrate your interest in languages by speaking at least two other languages.

Interested?

Please send your CV to Deanna Roe at deanna.roe@nuance.com. If you have any questions, please contact her at +44 207 922 5757.

We are looking forward to receiving your application! 

Nuance is the leading provider of speech and imaging solutions for businesses and consumers around the world.  Every day, millions of users and thousands of businesses experience Nuance by calling directory assistance, requesting account information, dictating patient records, telling a navigation system their destination, controlling their mobile phone or digitally reproducing documents that can be shared and searched.  With more than 2000 employees worldwide, we are committed to make the user experience more enjoyable by transforming the way people interact with information and how they create, share and use documents. Making each of those experiences productive and compelling is what Nuance is about. To strengthen our international team we are currently looking for a

Senior Research Engineer for Embedded Automatic Speech Recognition

Work Environment

·          You will work in the Embedded ASR research and production team in Merelbeke, Belgium or Aachen, Germany, working with state-of -he-art speech technology, tools and runtime software. Both Gent and Aachen are nice, historical European university cities.

·          You will work in an international company and cooperate with people and research teams on various locations across the globe. You may occasionally be asked to travel.

·          You will work  with our natural language understanding and dialogue research teams as well support our professional services teams.

·          You will work on the development of cutting edge speech recognition products for automotive platforms and mobile devices. You will help the engine cope with multi-lingual speech in various noise conditions, and this while respecting strong limitations on the usage of memory and processing power.

Key Responsibilities

·          Design, implementation, evaluation, optimization and testing of new algorithms and tools, with a strong focus on speech signal processing and acoustic modeling in adverse, noisy environments.

·          Activities are targeted at the creation of commercial products for resource limited platforms.

·          Focus on creating efficient production and development processes to bring the technology to marketable products in a wide range of languages.

·          Occasional application of the developed algorithms and tools for producing systems for a specific language.

·          Specification and follow-up of projects to make the system work with third party components, such as beam formers, echo cancellers or content data providers.

Your Profile

• You have a University degree in engineering, mathematics or physics.
• A PhD degree in speech processing or equivalent relevant experience is a strong asset.
• Experience in speech recognition research, especially acoustic modeling or signal processing, is required.
• Experience in speech processing, machine learning techniques or statistical modeling is required.
• Knowledge about small platforms and experience in developing software for them is a plus.
• Strong software skills are required, especially C/C++ and a scripting language like Perl or Python in a Linux/Unix environment. Knowledge of Matlab is a plus.
• Additional background in computational linguistics is a plus.
• You are a team player, willing to take initiative, and are goal oriented.
• You have a strong desire to make things “really work” in practice, on hardware platforms with limited memory and processing power.
• You are fluent in English and at least one other language, and you can write high quality English documentation. 

Interested?

Please send your CV to Deanna Roe at deanna.roe@nuance.com. If you have any questions, please contact her at +44 207 922 5757.

We are looking forward to receiving your application! 

Title: jr. Speech Scientist

Location: Aachen, Germany

Type: Permanent

Job:   

Overview:

Nuance is the leading provider of speech and imaging solutions for businesses and consumers around the World. Our technologies, applications and services make the user experience more compelling by transforming the way people interact with information and how they create, share and use documents. Every day, millions of users and thousands of businesses, experience Nuance by calling directory assistance, getting account information, dictating patient records, telling a navigation system their destination, or digitally reproducing documents that can be shared and searched. Making each of those experiences productive and compelling is what Nuance is all about.

Responsibilities:

Nuance is seeking a jr. Speech Scientist who possesses a solid background in natural language technology and computational linguistics.

Candidates should enjoy working in a fast-paced, collaborative atmosphere that applies speech science in a commercial, result driven and customer oriented setting.

As a jr. Speech Scientist in the Embedded Professional Services group, you will work on speech recognition grammars, statistical language models, prompts and custom voice development for leading edge automotive applications across the world, covering a broad range of activities in all project phases, including the design, development, and optimization of the system.

Representative duties include:

• Develop rule based grammars, train statistical language models for speech recognition and natural language understanding in commercial products in a variety of languages, according to UI Design specifications
• Identify or gather suitable text for training language models and custom voices
• Design, develop, and test semantic classifier rules and models
• Develop custom voices for use with Nuance’s leading text to speech products
• Direct voice talents for prompt recordings
• Organize and conduct usability tests
• Localization of speech resources for embedded speech applications
• Optimize accuracy of applications by analyzing performance and tuning statistical language models, grammars, and pronunciations within CPU and memory constraints of embedded platforms
• Contribute to the generation and presentation of client-facing reports

Qualifications:

• University degree in computational linguistics or Software design or similar degree
• Strong analytical and problem solving skills and ability to troubleshoot issues
• Good judgment and quick-thinking
• Strong programming skills, preferably Perl or Python
• Excellent written and verbal communications skills
• Ability to scope work taking technical, business and time-frame constraints into consideration
• Ability and willingness to travel abroad
• Works well independently and collaboratively in team settings in fast-paced environment
• Mastering Office applications

Beneficial Skills

• Additional language skills, eg. French, German, Spanish or other
• Strong programming skills in either Perl, Python, C, VB
• Speech recognition knowledge
• Pattern recognition, linguistics, signal processing, or acoustics knowledge

IEEE Transactions on Multimedia            
 Special Issue on Integration of Context and Content for Multimedia Management
=====================================================================

Guest Editors:

Alan Hanjalic, Delft University of Technology, The Netherlands
Alejandro Jaimes, IDIAP Research Institute, Switzerland
Jiebo Luo, Kodak Research Laboratories, USA
        Qi Tian, University of Texas at San Antonio, USA

---------------------------------------------------
URL: http://www.cs.utsa.edu/~qitian/cfp-TMM-SI.htm
---------------------------------------------------
Important dates:

Manuscript Submission Deadline:       April 1, 2008
        Notification of Acceptance/Rejection: July 1, 2008
        Final Manuscript Due to IEEE:         September 1, 2008
        Expected Publication Date:            January 2009

---------------------
Submission Procedure
---------------------
Submissions should follow the guidelines set out by IEEE Transaction on Multimedia.
Prospective authors should submit high quality, original manuscripts that have not
appeared, nor are under consideration, in any other journals.

-------
Summary
-------
Lower cost hardware and growing communications infrastructure (e.g., Web, cell phones,
etc.) have led to an explosion in the availability of ubiquitous devices to produce,
store, view and exchange multimedia (images, videos, music, text). Almost everyone is
a producer and a consumer of multimedia in a world in which, for the first time,
tremendous amount of contextual information is being automatically recorded by the
various devices we use (e.g., cell ID for the mobile phone location, GPS integrated in
a digital camera, camera parameters, time information, and identity of the producer).

In recent years, researchers have started making progress in effectively integrating
context and content for multimedia mining and management. Integration of content and
context is crucial to human-human communication and human understanding of multimedia:
without context it is difficult for a human to recognize various objects, and we
become easily confused if the audio-visual signals we perceive are mismatched. For the
same reasons, integration of content and context is likely to enable  (semi)automatic
content analysis and indexing methods to become more powerful in managing multimedia
data. It can help narrow part of the semantic and sensory gap that is difficult or
even impossible to bridge using approaches that do not explicitly consider context for
(semi)automatic content-based analysis and indexing.

The goal of this special issue is to collect cutting-edge research work in integrating
content and context to make multimedia content management more effective. The special
issue will unravel the problems generally underlying these integration efforts,
elaborate on the true potential of contextual information to enrich the content
management tools and algorithms, discuss the dilemma of generic versus narrow-scope
solutions that may result from "too much" contextual information, and provide us
vision and insight from leading experts and practitioners on how to best approach the
integration of context and content. The special issue will also present the state of
the art in context and content-based models, algorithms, and applications for
multimedia management.

-----
Scope
-----

The scope of this special issue is to cover all aspects of context and content for
multimedia management.

Topics of interest include (but are not limited to):
        - Contextual metadata extraction
        - Models for temporal context, spatial context, imaging context (e.g., camera
          metadata), social and cultural context and so on
- Web context for online multimedia annotation, browsing, sharing and reuse
- Context tagging systems, e.g., geotagging, voice annotation
- Context-aware inference algorithms
        - Context-aware multi-modal fusion systems (text, document, image, video,
          metadata, etc.)
- Models for combining contextual and content information
        - Context-aware interfaces
- Context-aware collaboration
- Social networks in multimedia indexing
- Novel methods to support and enhance social interaction, including
          innovative ideas integrating context in social, affective computing, and
          experience capture.
- Applications in security, biometrics, medicine, education, personal
          media management, and the arts, among others
- Context-aware mobile media technology and applications
- Context for browsing and navigating large media collections
- Tools for culture-specific content creation, management, and analysis

------------
Organization
------------
Next to the standard open call for papers, we will also invite a limited number of
papers, which will be written by prominent authors and authorities in the field
covered by this Special Issue. While the papers collected through the open call are
expected to sample the research efforts currently invested within the community on
effectively combining contextual and content information for optimal analysis,
indexing and retrieval of multimedia data, the invited papers will be selected to
highlight the main problems and approaches generally underlying these efforts.

All papers will be reviewed by at least 3 independent reviewers. Invited papers will
be solicited first through white papers to ensure the quality and relevance to the
special issue. The accepted invited papers will be reviewed by the guest editors and
expect to account for about one fourth of the papers in the special issue.

---------
Contacts
---------
Please address all correspondences regarding this special issue to the Guest Editors
Dr. Alan Hanjalic (A.Hanjalic@ewi.tudelft.nl), Dr. Alejandro Jaimes
(alex.jaimes@idiap.ch), Dr. Jiebo Luo (jiebo.luo@kodak.com), and Dr. Qi Tian
(qitian@cs.utsa.edu).
-------------------------------------------------------------------------------------

Guest Editors:
Alan Hanjalic, Alejandro Jaimes, Jiebo Luo, and Qi Tian

*CALL FOR PAPERS*

Special Issue of Speech Communication

on

*Spoken Language Technology for Education*



*Guest-editors:*

Maxine Eskenazi, Associate Teaching Professor, Carnegie Mellon University

Abeer Alwan, Professor, University of California at Los Angeles

Helmer Strik, Assistant Professor, University of Nijmegen

 

Language technologies have evolved to the stage where they are reliable
enough, if their strong and weak points are properly dealt with, to be
used for education. The creation of an application for education
presents several challenges: making the language technology sufficiently
reliable (and thus advancing our knowledge in the language
technologies), creating an application that actually enables students to
learn, and engaging the student. Papers in this special issue should
deal with several of these issues. Although language learning is the
primary target of research at present, papers on the use of language
technologies for other education applications are encouraged. The scope
of acceptable topic interests includes but is not limited to:

 

- Use of speech technology for education

- Use of spoken language dialogue for education

- Applications using speech and natural language processing for education

- Intelligent tutoring systems using speech and natural language

- Pedagogical issues in using speech and natural language technologies
for education

- Assessment of tutoring software

- Assessment of student performance

 

*Tentative schedule for paper submissions, review, and revision**: ** *

Deadline for submissions: June 1, 2008.

Deadline for decisions and feedback from reviewers and editors: August
31, 2008.

Deadline for revisions of papers: November 31, 2008.

 

*Submission instructions:*

Authors should consult the "Guide for Authors", available online, at
http://www.elsevier.com/locate/specom for information about the
preparation of their manuscripts. Authors, please submit your paper via
_http://ees.elsevier.com/specom_, choosing *Spoken Language Tech. *as
the Article Type, and  Dr. Gauvain as the handling E-i-C.

Consonant Challenge:
  Human-machine comparisons of consonant recognition in quiet and noise

                   Interspeech, 22-26 September 2008
                         Brisbane, Australia

* Update:
All information concerning the native listener experiments and baseline 
recognisers
including their results can now be found and downloaded from the Consonant 
Challenge website:
http://www.odettes.dds.nl/challenge_IS08/

* Deadline for submissions:
The deadline and paper submission guidelines for full paper submission (4 
pages) is April
7th, 2008. Paper submission is done exclusively via the Interspeech 2008 
conference website.
Participants of this Challenge are asked to indicate the correct Special 
Session during
submission. More information on the Interspeech conference can be found 
here: http://
www.interspeech2008.org/

* Topic of the Consonant Challenge:
Listeners outperform automatic speech recognition systems at every level 
of speech
recognition, including the very basic level of consonant recognition. What 
is not clear is
where the human advantage originates. Does the fault lie in the acoustic 
representations of
speech or in the recogniser architecture, or in a lack of compatibility 
between the two?
There have been relatively few studies comparing human and automatic 
speech recognition on
the same task, and, of these, overall identification performance is the 
dominant metric.
However, there are many insights which might be gained by carrying out a 
far more detailed
comparison.

The purpose of this Special Session is to make focused human-computer 
comparisons on a task
involving consonant identification in noise, with all participants using 
the same training
and test data. Training and test data and native listener and baseline 
recogniser results
will be provided by the organisers, but participants are encouraged to 
also contribute
listener responses.

* Call for papers:
Contributions are sought in (but not limited to) the following areas:

- Psychological models of human consonant recognition
- Comparisons of front-end ASR representations
- Comparisons of back-end recognisers
- Exemplar vs statistical recognition strategies
- Native/Non-native listener/model comparisons

* Organisers:
Odette Scharenborg (Radboud University Nijmegen, The Netherlands -- 
O.Scharenborg@let.ru.nl)
Martin Cooke (University of Sheffield, UK -- M.Cooke@dcs.shef.ac.uk)

CALL FOR PARTICIPATION

Phonetics & Phonology Laboratories, UMR-7018 CNRS-Univ. Paris III, will have a one-day meeting: Speech Production Workshop: Instrumentation-based Approach, on the day just after ACOUSTICS ’08 at ILPGA, Paris 5e.

This workshop is free of charge. If you would like to join us, please send us the following information by replying to this email so that we will know the approximate number of the participants.

 ----%---

 I will attend the workshop.

 Name:

 Affiliation:

 Email:

 ---%---

SPEECH PRODUCTION WORKSHOP: INSTRUMENTATION-BASED APPROACH

Date: 5th JULY, 2008 (Saturday)

Location: ILPGA 19 rue des Bernardins 75005 Paris

http://lpp.univ-paris3.fr/workshop.html

This workshop aims to introduce recent progresses of instrumentational approach in the fields of speech production research to advance our knowledge of key technologies and recent outcomes. Eleven invited speakers will give their recent development in four sessions.

[Voice Production]

J. Ohala      (to be announced)

  D. Demolin    Subglottal air pressure measurement

  H. Imagawa   High-speed digital imaging of vocal fold vibration

[Articulation 1]

  A. Marchal    Electropalatography, new development

 M-O. Berger   Multimodality acquisition of articulatory data

 P. Hoole       3D electromagnetic articulography, recent progress

[Articulation 2]

  R. Sock        Digital X-ray cinematography and database

  M. Stone       Magnetic resonance imaging (MRI), new techniques

  S. Masaki      Magnetic resonance imaging (MRI), motion imaging

[Demonstration]

  K. Honda      Noninvasive photoglottographic technique

  H. Takemoto   MRI-based vocal-tract solid models

Organizers

   Jacqueline Vaisisiere (LPP, UMR-7018 CNRS-Univ. Paris III, ATR-CIS)

   Kiyoshi Honda (LPP, UMR-7018 CNRS-Univ. Paris III)

   Shinji Maeda (LPP, UMR-7018 CNRS-Univ. Paris III, & ENST)

lundi 7 juillet, 13h30 - Séminaire externe
==============================================
Jorge Lucero
Department of Mathematics
University of Brasilia
Brésil

Titre et résumé à préciser

Salle de réunion du Département Parole et Cognition (B314)
3ème étage Bâtiment B ENSIEG
961 rue de la Houille Blanche
Domaine Universitaire

SIGIR-08 Workshop

2nd Workshop on Analytics for Noisy Unstructured Text Data
24 July 2008 , Singapore

http://and2008workshop.googlepages.com/

Call for Papers


Workshop Description and Objectives
Noise is an unavoidable fact of life. It can manifest itself at the
earliest stages of processing in the form of degraded inputs that
our systems must be prepared to handle. People are adept when it
comes to pattern recognition tasks involving typeset or handwritten
documents or recorded speech, machines less-so. From the perspective
of down-stream processes that take as their inputs the outputs of
recognition systems, including document analysis and OCR, noise can
be viewed as the errors made by earlier stages of processing, which
are rarely perfect and sometimes quite brittle.
Noisy unstructured text data is also found in informal settings such
as online chat, SMS, email, message board and newsgroup postings,
blogs, wikis and web pages. In addition to the aforementioned
recognition errors, such text may contain spelling errors,
abbreviations, non-standard terminology, missing punctuation,
misleading case information, as well as false starts, repetitions,
and pause-filling sounds such as “um” and “uh” in the case of speech.
By its very nature, noisy text warrants moving beyond traditional
text analytics techniques. Noise introduces challenges that need
special handling, either through new methods or improved versions of
existing ones. We invite you to submit your own unique perspectives
on this important topic.

Topics of Interest (not limited to)
Information Retrieval and Information Extraction on noisy texts
IR-related tasks (classification, clustering, genre recognition,
document summarization, keyword search) on noisy texts
Formal models for noise, characterization and classification of noise
Treatment of noisy data in special application fields
        - Historical Texts
         - Multilingual Texts
         - Blogs
         - Chat logs/SMS
         - Social Network Analysis
         - Patent Search
         - Optical Character Recognition
         - Automated Speech Recognition
         - Machine Translation
Data sets, benchmarks and evaluation techniques for analysis of noisy
texts

Participation
We hope that the workshop will allow researchers working in areas
related to unstructured data analytics, Natural Language Processing,
Information Extraction, Information Retrieval, etc., to focus on the
needs of users extracting useful information from noisy text. The
target audience is a mixture of academia and industry researchers
working with noisy text. We believe this work is of direct relevance
to domains such as call centers, the world-wide web, and government
organizations that need to analyze huge amounts of noisy data.

Important Dates
Paper Submission: May 16th, 2008
Notification of Acceptance: Jun 6th, 2008
Camera-Ready papers due: Jun 20th, 2008
Workshop at SIGIR 2008: Jul 24th, 2008

Submission Requirements
We invite papers up to 8 pages in length in the style specified at
http://and2008workshop.googlepages.com/submission There will also be a
Best Student Paper Award. Papers with a student as the primary
author/presenter will be eligible for this award.

Publication
We are currently in negotiation with a leading publisher for the
proceedings to be available onsite. We have also received tentative
approval for a special issue of a journal for post-workshop
publication of selected papers.

Workshop Chairs
  Daniel Lopresti
Lehigh University
  Shourya Roy
IBM Research, India Research Lab
   Klaus U Schulz
University of Munich
  L. Venkata Subramaniam
IBM Research, India Research Lab

Workshop contacts
* L. V. Subramaniam lvsubram@in.ibm.com
* Shourya Roy rshourya@in.ibm.com


Please visit the workshop website
***** http://and2008workshop.googlepages.com/  *****
for information about participation and submitting papers.

For general information, please visit the SIGIR website
***** http://www.sigir2008.org/ *****

The field of Semantic Computing (SC) brings together those disciplines concerned with connecting the (often vaguely-formulated) intentions  of humans with computational content. This connection can go both ways: retrieving,  using  and  manipulating  existing content according to user's goals ("do what the user means"); and creating, rearranging, and managing content that matches the  author's intentions ("do what the author means").

The content addressed in SC includes, but is not limited to, structured and semi-structured data, multimedia data, text, programs, services and, even, network behaviour. This connection between content and the user is made via (1) Semantic  Analysis, which analyzes content with the goal of converting it to meaning (semantics); (2)Semantic Integration, which  integrates  content and  semantics  from multiple sources; (3)Semantic Applications, which utilize content and semantics to solve  problems;  and  (4)Semantic Interfaces, which attempt to interpret users' intentions expressed in natural language or other communicative forms.

Example areas of SC include (but, again, are not limited to) the following:

ANALYSIS AND UNDERSTANDING OF CONTENT

• Natural-language processing
• Image and video analysis
• Audio and speech analysis
• Analysis of structured and semi-structured data
• Analysis of behavior of software, services, and networks

INTEGRATION OF MULTIPLE SEMANTIC REPRESENTATIONS

• Database schema integration
• Ontology integration
• Interoperability and Service Integration

SEMANTIC INTERFACES

• Natural-Language Interface
• Multimodal Interfaces

APPLICATIONS

• Semantic Web and other search technologies
• Question answering
• Semantic Web services
• Multimedia databases
• Engineering of software, services, and networks based on
• natural-language specifications
• Context-aware networks of sensors, devices, and/or applications

The second IEEE International Conference  on  Semantic  Computing (ICSC2008)  builds on the success of ICSC2007 as an international interdisciplinary forum  for  researchers  and  practitioners  to present research that advances the state of the art and practice of Semantic Computing, as well as identifying  the  emerging  research topics and defining the future of Semantic Computing.  The conference particularly welcomes interdisciplinary research that facilitates the ultimate success of Semantic Computing.

The event is located in Santa Clara, California, the heart of Silicon Valley. The technical program of ICSC2008 includes tutorials, workshops, invited talks, paper presentations, panel discussions, demo sessions, and an industry track. Submissions of high-quality papers  describing  mature results or on-going work are invited.

In addition to Technical Papers, the conference will feature

 * Tutorials  * Workshops * Demo Sessions * Special Sessions * Panels   * Industry Track

SUBMISSIONS

Authors are invited to submit an 8-page technical paper manuscript in double-column IEEE format following the guidelines available on the ICSC2008 web page under "submissions".

The Conference Proceedings will be published by the IEEE Computer Society Press.  Distinguished quality papers presented at the conference will be selected for publications in internationally renowned journals

eNTERFACE'08 the next international summer workshop on multimodal

interfaces will take place at LIMSI, in Orsay (near Paris), France,

during four weeks, August 4th-29th, 2008.

http://enterface08.limsi.fr

Please consider proposing projects and participate to the workshop (see

the Call for Projects proposal on the web site or attached to this mail).

eNTERFACE08 is the next of a series of successful workshops initiated by

SIMILAR, the European Network of Excellence (NoE) on Multimodal

interfaces. eNTERFACE'08 will follow the fruitful path opened by

eNTERFACE05 in Mons, Belgium, continued by eNTERFACE06 in Dubrovnik,

Croatia and eNTERFACE07 in Istambul, Turkey. SIMILAR came to an end in

2007, and the eNTERFACE http://www.enterface.org workshops are now

under the aegis of the OpenInterface http://www.openinterface.org

Foundation.

eNTERFACE'08 Important Dates

. December 17th, 2007: Reception of the complete Project proposal in

the format provided by the Author's kit

. January 10rd, 2008: Notification of project acceptance

. February 1st, 2008: Publication of the Call for Participation

. August 4th -- August 29th, 2008: eNTERFACE 08 Workshop

Christophe d'Alessandro

CNRS-LIMSI, BP 133 - F91403 Orsay France

tel +33 (0) 1 69 85 81 13 / Fax -- 80 88

EUSIPCO-2008 - 16th European Signal Processing Conference - August 25-29, 2008, Lausanne, Switzerland

- http://www.eusipco2008.org/

DEADLINE FOR SUBMISSION: February 8, 2008

The 2008 European Signal Processing Conference (EUSIPCO-2008) is the sixteenth in a series of conferences promoted by EURASIP, the European Association for Signal Processing (www.eurasip.org). This edition will take place in Lausanne, Switzerland, organized by the Swiss Federal Institute of Technology, Lausanne (EPFL).

EUSIPCO-2008 will focus on the key aspects of signal processing theory and applications. Exploration of new avenues and methodologies of signal processing will also be encouraged. Accepted papers will be published in the Proceedings of EUSIPCO-2008. Acceptance will be based on quality, relevance and originality. Proposals for tutorials are also invited.

*** This year will feature some exciting events and novelties: ***

- We are preparing a very attractive tutorial program and for the first time, access to the tutorials will be free to all registered participants! Some famous speakers have already been confirmed, but we also hereby call for new proposals for tutorials.

- We will also have top plenary speakers, including Stéphane Mallat (Polytechnique, France), Jeffrey A. Fessler (The University of Michigan, Ann Arbor, Michigan, USA), Phil Woodland (Cambridge, UK) and Bernhard Schölkopf (Max Planck Institute, Tübingen, Germany).

- The Conference will include 12 very interesting special sessions on some of the hottest topics in signal processing. See http://www.eusipco2008.org/11.html for the complete list of those special sessions.

- The list of 22 area chairs has been confirmed: see details at http://www.eusipco2008.org/7.html

- The social program will also be very exciting, with a welcome reception at the fantastic Olympic Museum in Lausanne, facing the Lake Geneva and the Alps (http://www.olympic.org/uk/passion/museum/index_uk.asp) and with the conference banquet starting with a cruise on the Lake Geneva on an historical boat, followed by a dinner at the Casino of Montreux (http://www.casinodemontreux.ch/).

Therefore I invite you to submit your work to EUSIPCO-2008 by the deadline and to attend the Conference in August in Lausanne.

IMPORTANT DATES:

Submission deadline of full papers (5 pages A4): February 8, 2008

Submission deadline of proposals for tutorials: February 8, 2008

Notification of Acceptance: April 30, 2008

Conference: August 25-29, 2008

More details on how to submit papers and proposals for tutorials can be found on the conference web site http://www.eusipco2008.org/

Eleventh International Conference on TEXT, SPEECH and DIALOGUE (TSD 2008)

Brno, Czech Republic, 8-12 September 2008

http://www.tsdconference.org/

The conference is organized by the Faculty of Informatics, Masaryk

University, Brno, and the Faculty of Applied Sciences, University of

West Bohemia, Pilsen. The conference is supported by International

Speech Communication Association.

Venue: Brno, Czech Republic

TSD SERIES

TSD series evolved as a prime forum for interaction between

researchers in both spoken and written language processing from the

former East Block countries and their Western colleagues. Proceedings

of TSD form a book published by Springer-Verlag in their Lecture Notes

in Artificial Intelligence (LNAI) series.

TOPICS

Topics of the conference will include (but are not limited to):

text corpora and tagging

transcription problems in spoken corpora

sense disambiguation

links between text and speech oriented systems

parsing issues

parsing problems in spoken texts

multi-lingual issues

multi-lingual dialogue systems

information retrieval and information extraction

text/topic summarization

machine translation

semantic networks and ontologies

semantic web

speech modeling

speech segmentation

speech recognition

search in speech for IR and IE

text-to-speech synthesis

dialogue systems

development of dialogue strategies

prosody in dialogues

emotions and personality modeling

user modeling

knowledge representation in relation to dialogue systems

assistive technologies based on speech and dialogue

applied systems and software

facial animation

visual speech synthesis

Papers on processing of languages other than English are strongly

encouraged.

PROGRAM COMMITTEE

Frederick Jelinek, USA (general chair)

Hynek Hermansky, Switzerland (executive chair)

FORMAT OF THE CONFERENCE

The conference program will include presentation of invited papers,

oral presentations, and a poster/demonstration sessions. Papers will

be presented in plenary or topic oriented sessions.

Social events including a trip in the vicinity of Brno will allow

for additional informal interactions.

CONFERENCE PROGRAM

The conference program will include oral presentations and

poster/demonstration sessions with sufficient time for discussions of

the issues raised.

IMPORTANT DATES

March 15 2008 ............ Submission of abstract

March 22 2008 ............ Submission of full papers

May 15 2008 .............. Notification of acceptance

May 31 2008 .............. Final papers (camera ready) and registration

July 23 2008 ............. Submission of demonstration abstracts

July 30 2008 ............. Notification of acceptance for

demonstrations sent to the authors

September 8-12 2008 ...... Conference date

The contributions to the conference will be published in proceedings

that will be made available to participants at the time of the

conference.

OFFICIAL LANGUAGE

of the conference will be English.

ADDRESS

All correspondence regarding the conference should be

addressed to

Dana Hlavackova, TSD 2008

Faculty of Informatics, Masaryk University

Botanicka 68a, 602 00 Brno, Czech Republic

phone: +420-5-49 49 33 29

fax: +420-5-49 49 18 20

email: tsd2008@tsdconference.org

LOCATION

Brno is the the second largest city in the Czech Republic with a

population of almost 400.000 and is the country's judiciary and

trade-fair center. Brno is the capital of Moravia, which is in the

south-east part of the Czech Republic. It had been a Royal City since

1347 and with its six universities it forms a cultural center of the

region.

Brno can be reached easily by direct flights from London, Moscow, Barcelona

and Prague and by trains or buses from Prague (200 km) or Vienna (130 km).

Call for papers
      Third Workshop on Speech in Mobile and Pervasive Environments
                  (in conjunction with ACM MobileHCI '08)
                        Amsterdam, The Netherlands
                             September 2, 2008
                      http://research.ihost.com/SiMPE



In the past, voice-based applications have been accessed using
unintelligent telephone devices through Voice Browsers that reside on the
server. The proliferation of pervasive devices and the increase in their
processing capabilities, client-side speech processing has been emerging as
a viable alternative. In SiMPE 2008, the third in the series, we will
continue to explore the various possibilities and issues that arise while
enabling speech processing on resource-constrained, possibly mobile
devices.



Topics of Interest:

All areas that enable, optimise or enhance Speech in mobile and pervasive
environments and devices. Possible areas include, but are not restricted
to:
      * Robust Speech Recognition in Noisy and Resource-constrained
Environments
      * Memory/Energy Efficient Algorithms
      * Multimodal User Interfaces for Mobile Devices
      * Protocols and Standards for Speech Applications
      * Distributed Speech Processing
      * Mobile Application Adaptation and Learning
      * Prototypical System Architectures
      * User Modelling
      * HCI issues in SiMPE applications
      * Design and cultural issues in SiMPE
      * Speech interfaces/applications for Developing Regions



Submissions:

We seek original, unpublished papers in the following three categories: (a)
Position papers that describe novel ideas that can lead to interesting
research directions, (b) Early results or work-in-progress that has
significant promise, or, (c) Full papers. Papers should be of 4-8 pages in
length in the MobileHCI publication format. The LaTeX and Microsoft Word
templates are available at the workshop website. All submissions should be
in the PDF format and should be submitted electronically through the
workshop submission web site,
http://www.easychair.org/conferences/?conf=simpe08. Since the submission
deadlines are dependent on the MobileHCI conference, we will not be able to
grant any extensions in any circumstances.

For any comments regarding submissions and participation, contact:
simpe08@easychair.org



Key Dates:

 * Paper Submission Deadline: May 05, 2008 (11:59 PM CET)
 * Notification of Acceptance: May 19, 2008
 * Early Registration Deadline: June 02, 2008
 * Workshop: September 02, 2008.



Organising Committee:

Amit A. Nanavati, IBM India Research Laboratory.
Nitendra Rajput, IBM India Research Laboratory.
Alexander I. Rudnicky, Carnegie Mellon University.
Markku Turunen, University of Tampere, Finland.



Programme Committee:

Lou Boves, University of Nijmegen, The Netherlands
Matt Jones, Swansea University, UK
Yoon Kim, Novauris Technologies, USA
Lars Bo Larsen, Aalborg University, Denmark
Gary Marsden, University of Cape Town, South Africa
Michael McTear, University of Ulster, Ireland
Shrikanth S. Narayanan, University of Southern California, USA
Tim Paek, Microsoft, USA
David Pearce, Motorola, UK.
Mike Phillips, Vlingo, USA
Markku Turunen, University of Tampere, Finland
Yaxin Zhang, Motorola, China
(More to be updated)



Websites:

 * SiMPE Workshop: http://research.ihost.com/SiMPE
 * ACM MobileHCI '08: http://www.mobilehci.org/
 * SiMPE 2007: http://research.ihost.com/SiMPE/2007
 * SiMPE 2006: http://research.ihost.com/SiMPE/2006

10-13 September 2008, Zadar, Croatia

http://www.elmar-zadar.org/

TECHNICAL CO-SPONSORS

IEEE Region 8

EURASIP - European Assoc. Signal, Speech and Image Processing

IEEE Croatia Section

IEEE Croatia Section Chapter of the Signal Processing Society

IEEE Croatia Section Joint Chapter of the AP/MTT Societies

TOPICS

--> Image and Video Processing

--> Multimedia Communications

--> Speech and Audio Processing

--> Wireless Commununications

--> Telecommunications

--> Antennas and Propagation

--> e-Learning and m-Learning

--> Navigation Systems

--> Ship Electronic Systems

--> Power Electronics and Automation

--> Naval Architecture

--> Sea Ecology

--> Special Session Proposals - A special session consist

of 5-6 papers which should present a unifying theme

from a diversity of viewpoints; deadline for proposals

is February 04, 2008.

KEYNOTE TALKS

* Professor Sanjit K. Mitra, University of Southern California, Los Angeles, California, USA:

Image Processing using Quadratic Volterra Filters

* Univ.Prof.Dr.techn. Markus Rupp, Vienna University

of Technology, AUSTRIA:

Testbeds and Rapid Prototyping in Wireless Systems

* Professor Paul Cross, University College London, UK:

GNSS Data Modeling: The Key to Increasing Safety and

Legally Critical Applications of GNSS

* Dr.-Ing. Malte Kob, RWTH Aachen University, GERMANY:

The Role of Resonators in the Generation of Voice

Signals

SPECIAL SESSIONS

SS1: "VISNET II - Networked Audiovisual Systems"

Organizer: Dr. Marta Mrak, I-lab, Centre for Communication

Systems Research, University of Surrey, UNITED KINGDOM

Contact: http://www.ee.surrey.ac.uk/CCSR/profiles?s_id=3D3937

SS2: "Computer Vision in Art"

Organizer: Asst.Prof. Peter Peer and Dr. Borut Batagelj,

University of Ljubljana, Faculty of Computer and Information

Science, Computer Vision Laboratory, SLOVENIA

Contact: http://www.lrv.fri.uni-lj.si/~peterp/ or

http://www.fri.uni-lj.si/en/personnel/298/oseba.html

SUBMISSION

Papers accepted by two reviewers will be published in

symposium proceedings available at the symposium and

abstracted/indexed in the INSPEC and IEEExplore database.

More info is available here: http://www.elmar-zadar.org/

IMPORTANT: Web-based (online) paper submission of papers in

PDF format is required for all authors. No e-mail, fax, or

postal submissions will be accepted. Authors should prepare

their papers according to ELMAR-2008 paper sample, convert

them to PDF based on IEEE requirements, and submit them using

web-based submission system by March 03, 2008.

SCHEDULE OF IMPORTANT DATES

Deadline for submission of full papers: March 03, 2008

Notification of acceptance mailed out by: April 21, 2008

Submission of (final) camera-ready papers : May 05, 2008

Preliminary program available online by: May 12, 2008

Registration forms and payment deadline: May 19, 2008

Accommodation deadline: June 02, 2008

GENERAL CO-CHAIRS

Ive Mustac, Tankerska plovidba, Zadar, Croatia

Branka Zovko-Cihlar, University of Zagreb, Croatia

PROGRAM CHAIR

Mislav Grgic, University of Zagreb, Croatia

CONTACT INFORMATION

Assoc.Prof. Mislav Grgic, Ph.D.

FER, Unska 3/XII

HR-10000 Zagreb

CROATIA

Telephone: + 385 1 6129 851=20

Fax: + 385 1 6129 568=20

E-mail: elmar2008 (_) fer.hr

For further information please visit:

http://www.elmar-zadar.org/

 Ecole Recherche Multimodale d'Information
techniques & sciences
http://glotin.univ-tln.fr/ERMITES

Spécial Apprentissage Automatique pour la RI

du 24 au 26 septembre 2008
à la Presqu'île de Giens - Var.

Soutenue par le CNRS, l'UMR LSIS,
l'Association Francophone de la Commmunication Parlée (AFCP),
et l'Univ. du Sud Toulon-Var.


ERMITES 2008 est centrée sur l'apprentissage automatique pour la recherche d'information multimodale, en s'appuyant sur les campagnes d'évaluation dont Technolangue (parole), Technovision et CLEF, NIST, TREC dont la plupart des orateurs sont des acteurs. ERMITES 2008 présente les bases communes entre ces systèmes, et jete des ponts entre les différentes disciplines sollicitées. Cette dizaine de spécialistes d'analyses conjointes de textes, images, sons ou vidéos intervient sur 3 jours, avec discussions et démonstrations ouvertes. L'un des objectifs d'ERMITES, via ces exposés théoriques et empiriques, est de guider des chercheurs à concevoir des systèmes RI multimodaux incontournables de part la diffusion de plus en plus anarchique de l'information. L'originalité d'ERMITES est de mettre l'accent sur les analyses jointes de diverses modalités, démontrant l'intérêt de sortir d'un pré-carré spécifique.

ERMITES se tient sur le superbe VVF de La Badine - Presqu'île de Giens-Var, (accès TGV Toulon) - Connections internet assurées.


** INTERVANTS et RESUMES par thèmes:

GRAVIER - CR CNRS IRISA http://www.irisa.fr/metiss
"Transcription automatique de la parole /
Analyse de documents oraux"
* On présentera les fondements de base du traitement automatique de la parole, dans le cadre de l'analyse de la parole contenue dans des données multimédia. Après une présentation des différents constituants d'un système de transcription automatique de la parole, on évoquera l'évaluation, les niveaux de performances que l'on peut attendre de tels systèmes et les difficultés liées à la diversité des documents.
* Suite à l'exposé sur la transcription automatique de la parole, on présentera des travaux sur le traitement automatique des langues appliqué à des transcriptions automatiques dans le but de tendre à une analyse sémantique de documents contenant de la parole. On évoquera ainsi tour à tour l'analyse morphosyntaxique, la segmentation thématique, l'extraction de mots clés à l'aide de méthodes classiques de recherche d'information ainsi que la détection des entités nommés. On mettra en évidence les adaptations nécessaires des outils de traitement automatique des langues pour prendre en compte les spécificités des transcriptions automatiques.


BESACIER L.- MC LIG http://www.liglab.fr
"Reconnaissance de la parole et traduction automatique
pour l'interaction et le traitement de contenus multilingues"
Un des enjeux dans le domaine de l'interaction est le multilinguisme pour les communications entre humains ou entre l'homme et la machine. A ce titre, je présenterai un aperçu de l'état actuel des technologies de reconnaissance automatique de la parole multilingue et de traduction automatique probabiliste, qui ont aussi un potentiel intéressant pour le traitement de contenus audio. Des exemples de projets académiques et industriels récents sur ce thème (IBM MASTOR, projets GALE et TC-STAR) seront également présentés.


FARINAS J. - MC IRIT http://www.irit.fr/recherches/SAMOVA
"Vérification Automatique de la langue /
Structuration automatique de documents AV"
* Structuration automatique de documents audiovisuels : de la recherche d'évènements saillants et de la caractérisation de l'environnement à la structuration du document. Exemple sur la caractérisation de l'environnement sonore à travers le projet ANR EPAC.
* Vérification automatique de la langue : un système automatique de classification de la parole au sein de la plateforme biométrique MISTRAL. Les campagnes d'évaluation NIST seront également abordées dans ce cadre.


MARCEL S.- Chercheur senior IDIAP / EPF Lausanne http://www.idiap.ch
"A tutorial on face detection and recognition:
application to information retrieval"
In this tutorial, we will present state-of-the-art and advanced techniques in face detection and face recognition with a particular emphasis on applications such as information retrieval.


FERTIL - DR CNRS LSIS http://www.lsis.org
"Un exemple de compression supervisée de données visuelles de grande dimension: prédire l'âge de personnes à partir de photos du visage"
Je présenterai une étude qui s'intéresse aux signes du vieillissement et à leur impact sur l'âge apparent, étude réalisée afin de construire un algorithme capable de déterminer l'âge d'individus à partir de leurs photos. Dans un premier temps, sont déterminées et analysées les transformations, anatomiques qui altèrent le visage à partir de l'âge adulte (au-delà de 20 ans). Puis les signes sur lesquels on se base pour prédire l'âge d'une personne sont examinés. En s'appuyant sur les observations précédentes, un modèle prédictif de l'âge est finalement construit et validé. Cette étude a été réalisée à l'aide d'une méthode linéaire de compression de données supervisée, la régression PLS (partial least squares) dont on pourra mesurer la puissance à cette occasion. On présentera aussi une version 'kernelisée' de l'algorithme, à utiliser lorsque les relations entre variable à prédire et variables prédictives sortent du cadre linéaire.


KERMORVANT C. - IR R&D A2IA http://www.a2ia.com/Web_Bao/ACCUEIL-fr.aspx
"Entreprise Content Managment : extraction de données dans les documents numérisés"
Malgré l'usage croissant des documents numériques, les entreprises continuent à devoir traiter des volumes importants de documents papier : chèques, factures, fax, lettres de clients, dossiers, etc. Même si ces documents papier sont numérisés, leur traitement nécessite des techniques complexes : analyse de documents, reconnaissance de caractères (imprimés ou manuscrits), classification, extraction d'informations. Dans cet exposé, je présente un aperçu des différentes techniques mises en œuvre dans les produits proposés par A2iA pour le traitement des documents numérisés ainsi que des exemples d'applications.


MERIALDO B.- Pr. Eurecom Sophia http://www.eurecom.fr
"RI et indexation dans TRECVID"
Cette présentation fera le point des techniques récentes d'indexation multimédia, en particulier concernant la vidéo numérique. On s'intéressera également aux problèmes d'évaluation, et à la description des campagnes d'évaluation TrecVideo.


QUENOT G.- CR CNRS LIG http://clips.imag.fr/mrim/
"Apprentissage actif et RI dans TRECVID"
La plupart des méthodes d'indexation par le contenu des images et des vidéos fonctionnent par apprentissage supervisé. La performance des systèmes dépend de la qualité des algorithmes d'apprentissage et de classification mais aussi de la quantité et de la qualité des annotations disponibles, lesquelles sont coûteuses à obtenir à cause de l'intervention hunaine qu'elle nécessitent. L'apprentissage actif consiste à utiliser un système de classification pour sélectionner les échantillons les plus informatifs pour l'entraînement de ce même système.
Ce cours comprend deux parties. L'introduction décrit les principes, l'histoire et les principales applications de l'apprentissage actif. Puis nous donnons une analyse détaillée d'une application de l'apprentissage actif à l'annotation de corpus et à l'indexation de concepts dans les vidéos dans le cadre de TRECVID.


QUAFAFOU M.- Pr. LSIS http://www.lsis.org
"Web Multimedia Mining"
La démocratisation du web et des moyens d'acquisition, de stockage et de la diffusion de données multimédia fait émerger un univers global riche et complexe. Ce monde constitué de données multimédia distribuées sur le web est en perpétuelle évolution. Ce gisement de données hétérogène, dynamique et inconsistant par nature offre de nouvelles opportunités différentes de celles du web mining et multimédia mining. Le but de cette présentation est d'explorer ces nouveaux challenges notamment suivant la perspective de l'apprentissage automatique.


LE MAITRE - Pr. LSIS http://www.lsis.org
"Indexation de page web par rapport à leur contenu et à leur rendu visuel"
Les concepteurs de page web organisent les informations qu'elles contiennent de façon à faciliter leur consultation par les utilisateurs. Une page web peut être vue comme un ensemble de blocs contenant des informations multimédia (texte, image, vidéo). L'apparence visuelle d'un bloc (fonte, couleur de fond...) et sa position dans la page fournit une information sur son importance. De plus, un bloc peut apporter de l'information à un autre bloc (voisin, englobant, etc.). Par exemple, le texte entourant une image ou la référençant peut être utilisé pour indexer cette image. Un autre avantage de la prise en compte du découpage d'une page en blocs est la possibilité de localiser les réponses à une requête : les blocs les plus similaires sont retournés plutôt que les pages dans leur totalité. La précision et l'exhaustivité des réponses à une requête à des pages web pourraient donc être significativement améliorées en prenant en compte le rendu visuel de ces pages en plus de leur contenu sémantique. Dans cet exposé seront présentés : les principales techniques de segmentation d'une page web à partir le leur arbre DOM, les techniques d'évaluation de l'importance d'un bloc dans une page et le modèle d'indexation d'une page web conçu dans le cadre d'un travail de recherche mené au sein de l'équipe INCOD du LSIS. Les premiers résultats de l'application de ce modèle à l'interrogation de journaux électroniques seront aussi présentés.


****************
** INSCRIPTION :
****************
Pré-inscription par e-mail à

glotin@univ-tln.fr (sujet= ERMITES08)

Les premiers inscrits seront prioritaires (limités à 32).
Paiement ensuite par chèque, bon de Commande ou CB à l'AFCP.

* 3 BOURSES de 150 euros sont offertes par l'AFCP *
en faire la demande lors de votre inscription.

** Tarifs 2008:
(incluant 2 nuits, 7 repas, 2 petits dej., 6 pauses boissons / cafés, actes,...):

En chambre avec 2 lits simples séparés :
Doctorant, Postdoc, Master = 260 euros.
Autres = 390 euros.

En chambre avec 1 lit simple :
Doctorant, Postdoc, Master = 300 euros.
Autres = 420 euros.

Formule avec seulement repas et actes (sans nuit ni pt. dej.) : 150 euros.

===
Les organisateurs
H. Glotin & J. Le Maitre

2008 IEEE International Workshop on MACHINE LEARNING FOR SIGNAL PROCESSING
(Formerly the IEEE Workshop on Neural Networks for Signal Processing)

October 16-19, 2008 Cancun, Mexico
Fiesta Americana Condesa Cancun, www.fiestamericana.com

Deadlines:
Submission of full paper:                     May 5, 2008
Notification of acceptance:                     June 16, 2008
Camera-ready paper and author registration:     June 23, 2008
Advance registration before:                    July 1, 2008

http://mlsp2008.conwiz.dk/

The workshop will feature keynote addresses, technical presentations, special
sessions and tutorials organized in two themes that will be included in the
registration. Tutorials will take place on the afternoon of 16 October, and
the workshop will begin on 17 October. The two themes for MLSP 2008 are
Cognitive Sensing and Kernel Methods for Nonlinear Signal Processing. Papers
are solicited for, but not limited to, the following areas:

Algorithms and Architectures:
Artificial neural networks, kernel methods, committee models, Gaussian
processes, independent component analysis, advanced (adaptive, nonlinear)
signal processing, (hidden) Markov models, Bayesian modeling, parameter
estimation, generalization, optimization, design algorithms.

Applications:
Speech processing, image processing (computer vision, OCR) medical imaging,
multimodal interactions, multi-channel processing, intelligent multimedia and
web processing, robotics, sonar and radar, biomedical engineering, financial
analysis, time series prediction, blind source separation, data fusion, data
mining, adaptive filtering, communications, sensors, system identification,
and other signal processing and pattern recognition applications.

Implementations:
Parallel and distributed implementation, hardware design, and other general
implementation technologies.

For the fourth consecutive year, a Data Analysis and Signal Processing
Competition is being organized in conjunction with the workshop. The goal of
the competition is to advance the current state-of-the-art in theoretical and
practical aspects of signal processing domains. The problems are selected to
reflect current trends, evaluate existing approaches on common benchmarks, and
identify critical new areas of research. Previous competitions produced novel
and effective approaches to challenging problems, advancing the mission of the
MLSP community. A description of the competition, the submissions, and the
results, will be included in a paper which will be published in the
proceedings. Winners will be announced and awards given at the workshop.

Selected papers from MLSP 2008 will be considered for a special issue of The
Journal of Signal Processing Systems for Signal, Image, and Video Technology,
to appear in 2009. The MLSP technical committee may invite one or more winners
of the data analysis and signal processing competition to submit a paper
describing their methodology to the special issue.

Paper Submission Procedure
Prospective authors are invited to submit a double column paper of up to six
pages using the electronic submission procedure at http://mlsp2008.conwiz.dk.
Accepted papers will be published on a CDROM to be distributed at the
workshop.

MLSP'2007 webpage: http://mlsp2008.conwiz.dk/

MLSP 2008 ORGANIZING COMMITTEE:

General Chair
Jose Principe

Program Chair
Deniz Erdogmus

Technical Chair
Tulay Adali

Publicity Chairs
Ignacio Santamaria
Marc Van Hulle

Publication Chair
Jan Larsen

Data Competition
Ken Hild
Vince Calhoun

Local Arrangements
Juan Azuela

Le GIPSA-Lab et le LPP organisent une école thématique CNRS pluridisciplinaire autour du thème de la nasalité :

*"Dynamique de la nasalité"*

du 15 au 19 Septembre 2008 sur l'île de Porquerolles.


L’objectif de cette école thématique multidisciplinaire est de *croiser les approches* et les méthodologies utilisées par les *phonologues*, les *phonéticiens* et les *dialectologues* autour d’un phénomène complexe : la nasalité et les processus de nasalisation. De nombreux intervenants présenteront leurs travaux de recherche et ceux de leurs collègues sur les processus de nasalisation en exposant les hypothèses, les formalismes, les méthodologies et les données qu’ils manipulent autour de trois grands axes : *les variétés dialectales*, *la phonologie de terrain* et les *aspects phonétiques* :

- les variétés dialectales : cartographie, phénomènes de nasalisation en Corse, en Sarde, phénomènes de diphtongaison, modélisation linguistique,

- la phonologie de terrain : géométrie des traits, nasalisation et dénasalisation, phonologie de laboratoire,

- la phonétique : instruments et techniques d’observation et d’analyse, phonétique acoustique, modélisation et synthèse

Public concerné : /doctorants, ingénieurs, chercheurs et enseignants-chercheurs dans les domaines de la parole et du langage/

Possibilité d’accueil : /50 participants/

Inscriptions (logement compris) : 300€/450€/830€

Plus d’information sur le site internet :
http://lpp.univ-paris3.fr/productions/conferences/2008/Dynamique_nasalite/

/Possibilité de Bourses AFCP pour les étudiants : Solange.Rossato@imag.fr <mailto:Solange.Rossato@imag.fr>/

COLLOQUE « LANGAGE ET COGNITION »

Ier et 2 octobre 2008

Maison de la Chimie

28, rue Saint-Dominique

75007 PARIS

Organisé par le Réseau Thématique Européen « Langage et Cognition »

Issu de l’ACI COGNITIQUE (1999-2002)

Sous l’égide de la Fondation Maison des Sciences de l’Homme

Adresse du Site : http://www.msh-paris.fr

Adresse du site d’OCTOGONE E.A 4156 :

http://w3.octogone.univ-tlse2.fr/articles.php?lng=fr&pg=20

Coordonnateurs :

Jean-Luc Nespoulous

Université de Toulouse Le Mirail

Institut Universitaire de France

Unité de Recherche Interdisciplinaire OCTOGONE E.A 4156

Laboratoire Jacques-Lordat

Institut des Sciences du Cerveau de Toulouse IFR 96

nespoulo@univ-tlse2.fr

&

Michel Fayol

Université Blaise Pascal, Clermont-Ferrand

Laboratoire de Psychologie Sociale et Cognitive UMR CNRS 6024

Mercredi Ier Octobre

8h30 : Accueil

8h45 : Introduction : Jean-Luc Nespoulous et Michel Fayol

Thème 1 : « Langage et Cognition : de la phrase au discours »

9h :       « From sentence to discourse : spatial framing adverbials as discourse markers »

Michel Charolles & Laure Sarda (Paris)

9h45 : « Effect on understanding of preposed and postposed locative prepositional phrases »

            Saveria Colonna (Paris)

« Effect of locative prepositional phrases on anaphor resolution and lexical desambiguation »

Joël Pynte (Paris)

10h30 : Discussion

10h45 : PAUSE

11h15 : « From Situation Models to Mental Simulations »

            Rolf Zwaan (Rotterdam)

12h : Discussion générale

12h-30 : repas

Thème 2 : « Diversité des Langues et Cognition »

14h30 : « La diversité des langues à travers les phénomènes de classification dans les langues orales et signées : aspects méthodologiques »

Barbara Köpke (Toulouse) & Colette Grinevald (Lyon)

15h15 : « Scripturisation des langues des signes et contraintes liées à la modalité : état des lieux »

            Brigitte Garcia (Paris)

16h : Discussion

16h15 : PAUSE

16h45 : « Formes linguistiques et catégories sémantiques : le cas des hiéroglyphes classificateurs de l’ancienne Egypte »

            Orly Goldwasser (Jerusalem), Colette Grinevald (Lyon) & Danièle Dubois (Paris)

17h30 : Discussion générale

Jeudi 2 Octobre

8h30 : Invited address : « The roles of general cognitive capacities and domain-specific cognitive skills in language comprehension »

            David Caplan (Cambridge) & Gloria Waters (Boston)

Thème 3 : « Langage, Communication, Pragmatique »

9h15 : « Quelles contraintes pragmatiques pour le langage et la communication ? »

            Michèle Guidetti (Toulouse) & Jean-Louis Dessalles (Paris)

10h : « Quel lien entre troubles pragmatiques et déficits cognitifs ? Le cas de la schizophrénie et des lésions cérébrales droites »

Maud Champagne (Montréal)

10h45 : Discussion

11h : PAUSE

11h30 :  « Structure(s) et contexte(s) dans une théorie pragmatique»

            Jef Verschueren (Anvers)

12h15 : Discussion générale

12h-30 : repas

Thème 4 : « Perception & Compréhension »

14h30 : « Speaking in the brain »

            Jean-François Démonet (Toulouse) & Mireille Besson (Marseille)

15h15 : « Abstract representations in speech perception : sounds, words and voices »

James McQueen (Nijmegen)

16h :     Discussion

16h15 : PAUSE

16h45 : « Functional and structural neuroanatomy of language »

            Angela Friederici (Leipzig)

17h30 : Discussion générale

18h : Fin des travaux

The Tenth International Conference on Multimodal Interfaces (ICMI

2008) will take place in Chania, Greece, on October 20-22, 2008. The

main aim of ICMI 2008 is to further scientific research within the

broad field of multimodal interaction and systems. The conference will

focus on major trends and challenges in this area, including help

identify a roadmap for future research and commercial success. ICMI

2008 will feature a main conference with keynote speakers, panel

discussions, technical paper presentations and discussion (single

track), poster sessions, and demonstrations of state-of-the-art

multimodal concepts and systems. Organized on the island of Crete,

ICMI-08 provides excellent conditions for brainstorming and sharing

the latest advances about multimodal interaction and systems in an

inspired setting full of history, mythology and art.

Paper Submission

There are two different submission categories: regular paper and short

paper. The page limit is 8 pages for regular papers and 4 pages for

short papers. The presentation style (oral or poster) will be decided

based on suitable delivery of the content.

Demo Submission

Proposals for demonstrations shall be submitted to demo chairs

electronically. A 1-2 page description of the demonstration is required.

Doctoral Spotlight

Doctoral Student Travel Support and Spotlight Session. Funds are

expected from NSF to support participation of doctoral candidates at

ICMI 2008, and a spotlight session is planned to showcase ongoing

thesis work. Students

interested in travel support can submit a short or long paper as

specified above.

Topics of interest include

* Multimodal and Multimedia processing

* Multimodal input and output interfaces

* Multimodal applications

* User Modeling and Adaptation

* Multimodal Architectures, Tools and Standards

* Evaluation of Multimodal Interfaces

*Important Dates:*

Paper submission: May 23, 2008

Author notification July 14, 2008

Camera ready deadline: August 15, 2008

Conference: October 20-22, 2008

Organizing Committee

General Co-Chairs

Vassilis Digalakis, TU Crete, Greece

Alex Potamianos, TU Crete, Greece

Matthew Turk, UC Santa Barbara, USA

Program Co-Chairs

Roberto Pieraccini, SpeechCycle, USA

Jian Wang, Microsoft Research, China

Yuri Ivanov, MERL 

Invited Speakers:

·         Nestor Becerra (Universidad de Santiago, Chile)

Aplicaciones de las tecnologías del habla en sistemas CALL (Computer Aided Language Training) y CAPT (Computer Aided Pronunciation Training)

·         Giussepe Ricardi (University of Trento, Italy)

Next Generation Spoken Language Interfaces

·         Björn Granstrom (KTH - Royal Institute of Technology, Suecia)

Embodied conversational agents  in verbal and non-verbal communication

·         Yannis Stilianou (University of Crete, Grecia)

Voice Conversion: State of the art and Perspectives

Important dates:

·         Full paper submission: July 20, 2008

• Notification of acceptance: October 1, 2008
• Conference V JTH 2008: November 12-14, 2008

Contact information:

VJTH’2008

Dept. Electronics and Telecommunications

Faculty of Engineering

Alda. Urkijo s/n

48013 Bilbao

Tel.: +34 946 013 969

Fax.: +34 946 014 259

E-mail: 5jth@ehu.es           Web: http://jth2008.ehu.es

EVALUATION CAMPAIGNS                ALBAYZIN-08 System Evaluation Proposal

The Speech Technologies Thematic Network ("Red Temática en Tecnologías del Habla") is a common forum where the researchers on Speech Technologies can work together and share experiences in order to:  Promote Speech Technology research, attracting new young researchers by means of formation courses, student interchange, grants and awards. Get investments from enterprises for Speech Technology research, looking for new applications  that can bring business opportunities. These applications must be shown in demostrators that can attract enterprises' interest. Make progress in creating collaboration ties among the Network members, enforcing the leadership of Spain in the Spanish speech technologies, as well as the co-official languages, such as Catalan, Basque or Galician.

In order to promote new young researchers' Speech Technology investigation, the "Red Temática en Tecnologías del Habla"  organizes a system evaluation proposal, on the next areas:

• Language recognition

Registration Form

• Speech synthesis

Registration Form

• Automatic text translation

Registration Form

http://jth2008.ehu.es/form_ALBAYZIN08_TA_en.pdf

 2008 International Symposium on

         Chinese Spoken Language Processing (ISCSLP 2008)

                          December 16 - 19, 2008

                              Kunming, China

                       http://www.iscslp2008.org

ISCSLP’08 is the flagship conference of ISCA SIG-CSLP (Special Interest Group on Chinese Spoken Language Processing).

ISCSLP'08 will be held during December 16-19, 2008 in Kunming hosted by The University of Science and Technology of China and Yunnan University.

ISCSLP (International Symposium on Chinese Spoken Language Processing) is a conference for scientists, researchers, and practitioners to report and discuss the latest progress in all scientific and technological aspects of Chinese spoken language processing (CSLP). The idea of having a series of regular conferences devoted to CSLP was an outcome of a small-group meeting held in December 1997 in Singapore. The meeting was organized and chaired by Professor Chin-Hui Lee, then worked at Bell Laboratories, USA; and attended by Professors Tai-Yi Huang and Ren-Hua Wang from mainland China, Professors Chorkin Chan and Pak-Chung Ching from Hong Kong, Professor Kim-Teng Lua and Dr. Haizhou Li from Singapore, and Professors Lin-Shan Lee and Hsiao-Chuan Wang from Taiwan. A Steering Committee, being chaired by Professor Chin-Hui Lee and consisting of the abovementioned nine members, was established to oversee the ISCSLP conferences. It was decided that a bi-annual symposium will be organized and hosted initially by research groups from Asia Pacific regions. Since its inception, ISCSLP has become the world's largest and most comprehensive technical conference focused on Chinese spoken language processing and its applications. In ISCSLP 2002, a special interest group was formed as SIG-CSLP of International Speech Communication Association (ISCA). ISCSLP is now an ISCA and IEEE supported event.

We invite your participation in this premier conference, where the language from ancient civilizations embraces modern computing technology. The ISCSLP'08 will feature world-renowned plenary speakers, tutorials, exhibits, and a number of lecture and poster sessions. The concrete version is attached to this mail.

In response to popular requests from authors, the paper submission deadline is extended. The new deadline is Jul 29, 2008.

The Keynote speakers of ISCSLP2008 is as following:

Qiang Huo

Microsoft Research Asia, Beijing, China

Research Area

Automatic speech & speaker recognition and related multidisciplinary research topics

Chinese character recognition

Biometric authentication

Document analysis and recognition

Machine learning, etc.

Shigeki Sagayama

Department of Information Physics and Computing Graduate School of Information Science and Technology, The University of Tokyo, Japan

Research Area

Speech and spoken language processing

Signal processing

Music signal/Information processing

Hand-written character recognition

Multimedia Information processing, etc.

Vincent Vanhoucke

Google, USA

Research Area

Software engineering

Text recognition

Speech recognition

Image processing

Face recognition, etc.

Yuqing Gao

IBM T. J. Watson Research Center, USA

Research Area

Speech recognition

understanding and Translation Research

The large vocabulary continuous speech dictation system

Speech-to-speech translation research, etc.

Hideki Kawahara

Design Information Sciences Department, Faculty of Systems Engineering, Wakayama University, Japan

Research Area

Focus on the use of STRAIGHT in research on human speech perception

Signal processing models of hearing and neural networks

Interaction between speech perception and production, etc.

Yu Hu

Research Director of iFLYTEK, Hefei, China

Research Area

Speech pronunciation evaluation

Speech pronunciation defect detection, etc

Paper Submission

Authors are invited to submit original, unpublished work in English.

Papers should be submitted via http://www.iscslp2008.org.

Each submission will be reviewed by two or more reviewers.

At least one author of each paper is required to register.

Schedule

Full paper submission by Jun. 29, 2008

Extended deadline for submission of full papers by Jul 29, 2008

Notification of acceptance by   Aug.24, 2008

Camera ready papers by    Sep.03, 2008

Registration to cover an accepted paper by    Sep.19, 2008

Call for Papers:
Second IEEE Spoken Language Technology Workshop
Goa, India
December 15-18, 2008

The Second IEEE Spoken Language Technology (SLT) workshop will be held from December 15 to December 18, 2008 in Goa, India. The goal of this workshop is to bring both the speech processing and natural language processing communities together to share and present recent advances in various areas of spoken language technology, with the expectation that such a confluence of the researchers from both communities will foster new ideas, collaborations and new research directions in this area. The SLT 2008 workshop is endorsed by both ISCA and ACL organizations and eligible participants can apply for ISCA grants (http://www.isca-speech.org/grants.html).

Spoken language technology is a vibrant research area, with the potential for significant impact on government and industrial applications especially with the diversity and challenges offered by the multilingual business climates of today's world.

The workshop solicits papers on all aspects of spoken language technology:

 o Spoken language understanding
 o Spoken document summarization
 o Machine translation for speech
 o Spoken dialog systems
 o Spoken language generation
 o Spoken document retrieval
 o Human computer Interactions (HCI)
 o Speech data mining
 o Information extraction from speech
 o Question answering from speech
 o Multimodal processing
 o Spoken language based assistive technologies
 o Spoken language systems and applications
 o Spoken language databases and standards

In addition, this year's workshop will feature three special sessions:

 1) Challenges in Asian spoken language processing with special emphasis on Indian languages
 2) Mining human-human conversations: A resource for building efficient human-machine dialogs
 3) Spoken Language on the go: Challenges and Opportunities for spoken language processing on mobile devices

Submissions for the Technical Program
-------------------------------------
The workshop program will consist of tutorials, oral and poster presentations, and panel discussions. Attendance will be limited with priority for those who will present technical papers; registration is required of at least one author for each paper. Submissions are encouraged on any of the topics listed above. The style guide, templates, and submission form will follow the IEEE ICASSP style. Three members of the Scientific Committee will review each paper. The workshop proceedings will be published on a CD-ROM.

Important Dates
---------------
*Camera-ready paper submission deadline: August 8, 2008
Hotel Reservation and Workshop registration opens: August 8, 2008
Paper Acceptance / Rejection: September 15, 2008
Hotel Reservation and Early Registration closes: October 5, 2008
Workshop: December 15-18, 2008*

For more information visit the SLT 2008 website http://slt2008.org or contact the organizing committee at info@slt2008.org <mailto:info@slt2008.org> if you have any questions.

BIOSIGNALS will be held in Porto (Portugal) on January 14 - 17 2009.  Technically co-sponsored by the IEEE Engineering in Medicine and Biology Society (EMBS) and in cooperation with the Association for Computing Machinery (ACM SIGART) and the Association for the Advancement of Artificial Intelligence (AAAI), BIOSIGNALS brings together top researchers and practitioners in several areas of Biomedical Engineering, from multiple areas of knowledge, including biology, medicine, engineering and other physical sciences, interested in studying and using models and techniques inspired from or applied to biological systems. A diversity of signal types can be found in this area, including image, audio and other biological sources of information. The analysis and use of these signals is a multidisciplinary area including signal processing, pattern recognition and computational intelligence techniques, amongst others.  The proceedings will be indexed by several major international indexers, including INSPEC and DBLP. Additionaly, a selection of the best papers of the conference will be published in a book, by Springer-Verlag. Best paper awards will be distributed during the conference.  Further details can be found at the BIOSIGNALS conference web site (http://www.biosignals.org) This conference is co-located and part of the Joint Conference on Biomedical Engineering Systems and Technologies (BIOSTEC www.biostec.org).  Workshops and special sessions are also invited. If you wish to propose a workshop or a special session, for example based on the results of a specific research project, please contact the secretariat.   Best regards, Marina Carvalho BIOSIGNALS Secretariat Av. D.Manuel I, 27A 2ºesq. 2910-595 Setúbal, Portugal Tel.: +351 265 520 185 Fax: +44 203 014 5436 Email: secretariat@biosignals.org    ************************************************************************* web site: http://www.biosignals.org  IMPORTANT DATES:               Regular Paper Submission (EXTENDED): July 21, 2008   Authors Notification: October 9, 2008    Final Paper Submission and Registration: October 23, 2008     in cooperation with: ACM SIGART and AAAI   technically co-sponsored: IEEE EMB      proceedings indexed by INSPEC and DBLP   best papers published by Springer-Verlag  -----------------------------------------------------------------------  CONFERENCE TOPIS:   - Medical Signal Acquisition, Analysis and Processing  - Wearable Sensors and Systems  - Real-time Systems  - Biometrics  - Pattern Recognition  - Computational Intelligence  - Evolutionary Systems  - Neural Networks  - Speech Recognition  - Acoustic Signal Processing  - Time and Frequency Response  - Wavelet Transform  - Medical Image Detection, Acquisition, Analysis and Processing  - Physiological Processes and Bio-signal Modeling, Non-linear dynamics  - Bioinformatics  - Cybernetics and User Interface Technologies  - Electromagnetic fields in biology and medicine  -----------------------------------------------------------------------  KEYNOTE SPEAKERS: 	   - Edward H. Shortliffe, Arizona State University, United States  - Vimla L. Patel, Arizona State University, United States  - Pier Luigi Emiliani, Institute of Applied Physics "Nello Carrara" (IFAC) of the Italian National Research Council (CNR), Italy  - Maciej Ogorzalek, Jagiellonian University, Poland       --------------------WORKSHOP:  (Regular Paper Submission: October 17, 2008)  - Medical Image Analysis and Description for Diagnosis Systems - MIAD 2009     http://www.biostec.org/MIAD.htm

IEEE International Conference on Acoustics, Speech, and Signal Processing

http://icassp09.com

Sponsored by IEEE Signal Processing Society

April 19 - 24, 2009

Taipei International Convention Center

Taipei, Taiwan, R.O.C.

The 34th International Conference on Acoustics, Speech, and Signal Processing (ICASSP) will be held at the Taipei International Convention Center in Taipei, Taiwan, April 19 - 24, 2009. The ICASSP meeting is the world’s largest and most comprehensive technical conference focused on signal processing and its applications. The conference will feature world-class speakers, tutorials, exhibits, and over 50 lecture and poster sessions on:

Audio and electroacoustics

Bio imaging and signal processing

Design and implementation of signal processing systems

Image and multidimensional signal processing

Industry technology tracks

Information forensics and security

Machine learning for signal processing

Multimedia signal processing

Sensor array and multichannel systems

Signal processing education

Signal processing for communications

Signal processing theory and methods

Speech and language processing

Taiwan: The Ideal Travel Destination. Taiwan, also referred to as Formosa – the Portuguese word for "graceful" – is situated on the western edge of the Pacific Ocean off the southeastern coast of mainland Asia, across the Taiwan Strait from Mainland China. To the north lie Okinawa and the main islands of Japan, and to the south is the Philippines. ICASSP 2009 will be held in Taipei, a city that blends traditional culture and cosmopolitan life. As the political, economic, educational, and recreational center of Taiwan, Taipei offers a dazzling array of cultural sights not seen elsewhere, including exquisite food from every corner of China and the world. You and your entire family will be able to fully experience and enjoy this unique city and island. Prepare yourself for the trip of your dreams, as Taiwan has it all: fantastic food, a beautiful ocean, stupendous mountains and lots of sunshine!

Submission of Papers: Prospective authors are invited to submit full-length, four-page papers, including figures and references, to the ICASSP Technical Committee. All ICASSP papers will be handled and reviewed electronically. The ICASSP 2009 website www.icassp09.com will provide you with further details. Please note that the submission dates for papers are strict deadlines.

Tutorial and Special Session Proposals: Tutorials will be held on April 19 and 20, 2009. Brief proposals should be submitted by August 4, 2008, to Tsuhan Chen at tutorials@icassp09.com and must include title, outline, contact information, biography and selected publications for the presenter, a description of the tutorial, and material to be distributed to participants. Special sessions proposals should be submitted by August 4, 2008, to Shih-Fu Chang at specialsessions@icassp09.com and must include a topical title, rationale, session outline, contact information, and a list of invited speakers. Tutorial and special session authors are referred to the ICASSP website for additional information regarding submissions.

General Chair

Lin-shan, Lee

National Taiwan University

General Vice-Chair

Iee-Ray Wei

Chunghwa Telecom Co.,Ltd.

Secretaries General

Tsungnan Lin

National Taiwan University

Fu-Hao Hsing

Chunghwa Telecom Co.,Ltd

Technical Program Chairs

Liang-Gee Chen

National Taiwan University

James R. Glass

Massachusetts Institute of Technology

Technical Program Members

Petar Djuric

Stony Brook University

Joern Ostermann

Leibniz University Hannover

Yoshinori Sagisaka

Waseda University

Plenary Sessions

Soo-Chang Pei (Chair)

National Taiwan University

Hermann Ney (Co-chair)

RWTH Aachen

Special Sessions

Shih-Fu Chang (Chair)

Columbia University

Lee Swindlehurst (Co-chair)

University of California, Irvine

Tutorial Chair

Tsuhan Chen

Carnegie Mellon University

Publications Chair

Homer Chen

National Taiwan University

Publicity Chair

Chin-Teng Lin

National Chiao Tung University

Finance Chair

Hsuan-Jung Su

National Taiwan University

Local Arrangements Chairs

Tzu-Han Huang

Chunghwa Telecom Co.,Ltd.

Chong-Yung Chi

National Tsing Hwa University

Jen-Tzung Chien

National Cheng Kung University

Conference Management

Conference Management Services