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 

Thanks  to Ebru's efforts the online grant application system has been modified accordingly to Alan Black's suggestions and is now reachable for all isca-students.org registered users from the main menu and through the page:

http://www.isca-students.org/grants
 
You might want to link it from the relevant pages on IS08 and ISCA's websites.

Use it when registering for Interspeech 2008 (Brisbane)!

We remain at your disposal for any further need.

Marco A. Piccolino-Boniforti
ISCA-SAC web coordinator

map55@cam.ac.uk

AISV-SIG

Italian Speech Communication Association

Regional Activities

Annual Report 2005-2008

Conference, Workshops and Summer Schools

organized or sponsored by AISV-SIG during the 2005-2008 period

The purpose AISV Italian Speech Communication Regional ISCA Special Interest Group is that of promoting, in the scientific, technical, normative, industrial, social, professional and didactic fields, the study of Speech Sciences in Italy. In particular, the Association is turned to all the subjects differently involved in the study of Phonetics (Acustic/Articulatory), of the Speech Signal Processing and of the Automatic Treatment of the Language (Trattamento Automatico del Linguaggio - TAL) in which are grouped all those disciplines that take care of the voice man-machine interaction and of the human language understanding. The purpose of AISV-SIG is also that of maintaining a direct connection with ISCA with the aim of more effectively valuing the studies and researches on these fields by the Italian scientific community in the European and International framework.

AISV-SIG pursues its own purposes in particular:

• promoting and favouring the studies and researches on Speech Sciences in all their different fields;
• facing the problems relating to the definition, the instruction and the spreading of Speech Sciences and to their institutional positioning;
• promoting and favouring the spreading and the deepening of the know ledges acquainted in Speech Sciences through the cure of publications, the organization of conferences, the organization of course, schools and masters, the attribution of scholarships, ecc.;
• promoting and favouring scientific and technical exchanges of information and relationships of collaboration between the associates;
• collaborating with Italian and international Organizations responsible of financing the scientific search in this field;
• favouring and promoting relationships with other Italian and international Associations or Organizations whose purposes are coherent with the own ones for the realization of common aims.

ACTIVITIES

2005

AISV 2005 Summer School:

1^st AISV Summer School

"Modelli linguistici e tecnologici per l'analisi e la gestione di corpora vocali"

“Linguistic and Technological Models for the Analysis and Management of Speech Corpora”

Castello Orsini,  Soriano nel Cimino

10-14 October 2005

http://www2.pd.istc.cnr.it/AISVScuolaEstiva2005/

The 1^st AISV Summer School was indeed a big success...... we had more than 20 new and very interested young attendees which learned the basis of Linguistic and Technological Methodologies for the Analysis and Management of Speech Corpora. Various speech analysis software were presented and their functionality illustrated and, among them, the PRAAT software was introduced in various practical exercises.

AISV 2005 - Annual Conference:

2^nd AISV National Conference
"ANALISI PROSODICA" - teorie, modelli e sistemi di annotazione

“Prosodic Analysis - Theory, Models and Annotation Techniques”

Campus di Fisciano - "Aula delle Lauree"

Salerno, 30 November - 2 December 2005
http://www2.pd.istc.cnr.it/AISV2005/

The AISV 2005 2^nd Conference was organized in Salerno and there were more than 80 attendees. During the Conference a very interesting RoundTable on "Modelli di Analisi e Sistemi di etichettatura prosodica a confronto" (Prosodic Analysis and Annotation) was organized with G. Marotta chairman supervision and D. Hirst, M. D'Imperio, M. Contini, Ph. Martin were invited. This was a very promising starting point for confronting Italian and French Regional SIG Group activities!

2006

AISV 2006 Summer School:

2^nd AISV Summer School

"Dalla fonetica alla fonologia sperimentale: teoria e metodi di analisi"

“From Phonetics to Experimental Phonology: Theory and Analysis Methods”

Soriano nel Cimino

25-29 September 2006

http://www2.pd.istc.cnr.it/AISVScuolaEstiva2006/

The 2^nd AISV Summer School was again a big success with more than 30 very interested attendees. The School was focused on an advanced introduction on experimental phonetics (acoustic and articulatory) and on “Articulatory Phonology” and, more specifically, it was organized with different lessons on:

Introduction to Acoustic Phonetics

·    analysis hardware/software devices and measurement techniques

·    open source programs for speech analysis (Praat, WaveSurfer, Snack);

·    use of acoustic analysis for phonetic descriptions (i.e. vowels, vocalic systems,…);

·    acoustic analysis and pathologies;

·    acoustic analysis of phonetic development;

Introduction to Articulatory Phonetics

·    analysis hardware/software devices and measurement techniques

·    elettropalatografy (EPG), elettroglottografy, EMMA, ELITE, ultrasound-based devices, fMRI.

Introduction to Articulatory Phonology and to “Task Dynamics”

AISV 2006 - Annual Conference:

3^rd AISV National Conference

"Scienze Vocali e del Linguaggio - Metodologie di Valutazione e Risorse Linguistiche"

“Speech and Language Sciences – Evaluation Metodologies and Linguistic Resources”

ITC-irst (Auditorium) - Pantè di Povo TRENTO

29-30 November - 1 December 2006

http://aisv2006.itc.it/

The AISV 2006 3^rd Conference was organized in Trento at ITC-irst (now FBK) with more than 100 attendees. During the Conference a very interesting Round Table on “Speech and Language Sciences – Evaluation Metodologies and Linguistic Resources” was organized with Bernardo Magnini, (ITC-irst, Trento) chairman supervision and Francesco Cutugno (Università "Federico II", Napoli), John Garofolo (NIST - National Institute of Standards and Technology, USA) and Carol Peters (Istituto di Scienza e Tecnologie dell'Informazione, ISTI-CNR, Pisa)  were invited.

2007

AISV 2007 Summer School:

3^rd AISV Summer School/Professional Course

"La voce in ambito Forense - trascrizione, comparazione, manomissione, speaker profile ecc.”

“Forensic Phonetics and Speech – Speaker Profile, Speaker Identification, Transcriptions,…

 Soriano nel Cimino

17-21 September 2007

http://www2.pd.istc.cnr.it/AISVScuolaEstiva2007/

The 3^rd AISV Summer School could be considered also as a professional course on Forensic Phonetics. There were 27 very interested attendees who improve their knowledge on linguistics, psycholinguistics, phonetics, acoustics, statistics and probability theory applied to forensic phonetics. In particular the content of the school was regarding:

·          Speaker identification

·          Speaker profile

·          Utterances analysis

·          Voice line-up

·          Tape authentication

·          Transcription of official recording

·          Speech enhancement

·          Audio signal (identification)

·          Voice changing/morphing

·          Audio signal analysis

AISV 2007 - Annual Conference:

4^th AISV National Conference

"La Fonetica Sperimentale: Metodi e Applicazioni”

“Experimental Phonetics: Methods and Applications”

Campus UniCal - Arcavacata di Rende (CS)

3-5 December 2007

http://www.linguistica.unical.it/aisv2007/

The AISV 2006 4^th Conference was organized  in Arcavacata di Rende at the Unical Campus.The main theme of the Conference was that of promoting Experimental Phonetics underlining especially the interdisciplinary nature of ifs methodological approach. Various Italian experts in the field gathered together confronting their ideas and results. A very interesting Round Table on “Experimental Phonetics: Methods and Applications” was organized with Emanuela Magno Caldognetto, (ISTC CNR, Padova) chairman supervision and Sergio Canazza Targon (Udine University), Carlos Delgado Romero (Voice Lab – Madrid Police), Nicola Lombardo (ORL Clinic, Magna Grecia University - Catanzaro), Pierluigi Salza (Loquendo S.p.A. - Torino) and Valentina Valentini (Rome University - La Sapienza) were invited.

2008 (what's up)

AISV 2008 Summer School:

4^th AISV Summer School

"Archivi di corpora vocali - conservazione, catalogazione, restauro audio e fruizione dei documenti sonori”

“Speech Corpora – maintenance, preservation and restoration of audio data”

Soriano nel Cimino

08-12 September 2008

http://www2.pd.istc.cnr.it/AISVScuolaEstiva2008/

AISV 2008-2009 - Annual Conference:

5^th AISV National Conference

"La dimensione temporale del parlato"

"The Temporal dimension of Speech"

January-February 2009

Zurich University

http://www2.pd.istc.cnr.it/AISV2008/

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 

Full registration details and fees are under “Registration” ( www.interspeech2008.org/registration )

 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: 15 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. 

September 6-10, 2009, 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."

Interspeech 2011

Palazzo dei Congressi,  Italy, August 27-31, 2011.

Organizing committee

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).

More information www.interspeech2011.org

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.

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!

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!

The experience speaks for itself™

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!

The experience speaks for itself™

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

Opened positions/internships at Microsoft: Danish Linguist (M/F)

MLDC – Microsoft Language Development Center, a branch of the Microsoft Product Group that develops Speech Recognition and Synthesis Technologies, situated in Porto Salvo, Portugal (http://www.microsoft.com/portugal/mldc), is seeking a full-time temporary language expert in the Danish language, for a 3-4 month contract, to work in speech technology related development projects. The successful candidate should have the following requirements:

·         Be native or near native Danish speaker

·         Have a university degree in Linguistics or related field (preferably in Danish Linguistics)

·         Have an advanced level of English

·         Have some experience in working with Speech Technology/Natural Language Processing/Linguistics, either in academia or in industry

·         Have some computational ability – no programming is required, but he/she should be comfortable working with MS Windows and MS Office tools

·         Have team work experience

·         Willing to work in Porto Salvo (near Lisbon) for the duration of the contract

·         Willing to start in September 2008

To apply, please submit your resume and a brief statement describing your experience and abilities to Daniela Braga: i-dbraga@microsoft.com

We will only consider electronic submissions. 

Opened positions/internships at Microsoft: Catalan Linguist (M/F)

MLDC – Microsoft Language Development Center, a branch of the Microsoft Product Group that develops Speech Recognition and Synthesis Technologies, situated in Porto Salvo, Portugal (http://www.microsoft.com/portugal/mldc), is seeking a full-time temporary language expert in the Catalan language, for a 3-4 month contract, to work in speech technology related development projects. The successful candidate should have the following requirements:

·         Be native or near native Catalan speaker

·         Have a university degree in Linguistics or related field (preferably in Catalan Linguistics)

·         Have an advanced level of English

·         Have some experience in working with Speech Technology/Natural Language Processing/Linguistics, either in academia or in industry

·         Have some computational ability – no programming is required, but he/she should be comfortable working with MS Windows and MS Office tools

·         Have team work experience

·         Willing to work in Porto Salvo (near Lisbon) for the duration of the contract

·         Willing to start in September 2008

To apply, please submit your resume and a brief statement describing your experience and abilities to Daniela Braga: i-dbraga@microsoft.com

We will only consider electronic submissions. 

titre du sujet : Synthèse vocale de haute qualité

Introduction
=========

Ces dernières années ont vu l'émergence de systèmes de synthèse de la parole construits autour de base de données de parole de taille importante qui correspondent le plus souvent à quelques heures d'enregistrement de parole. A des degrés divers, ces systèmes considèrent qu'il est possible de produire une parole de qualité en allant chercher des fragments de sons dans une base de données enregistrée au préalable par un locuteur. Ce type d'approche pousse à l'extrême l'hypothèse fonctionnelle des systèmes fondés sur la concaténation d'unités acoustiques. Avec une base de données suffisamment importante, il doit être possible de couvrir statistiquement les cas les plus fréquents de coarticulation sonore.

Des systèmes récents comme Festival (Black 1995), CHATR (Campbell 1996), Whistler (Huang 1996), XIMERA (Toda 2006), IBM \citethese{Eid06}, prouvent que cette approche méthodologique permet de construire des systèmes de synthèse de très bonne qualité.

En suivant cette méthodologie, les modèles ne sont plus utilisés pour produire des valeurs de paramètres qui serviront à la génération d'un signal de parole. Ils sont en revanche utilisés pour rechercher dans la base d'exemples sonores un extrait de parole qui sera le plus proche possible des paramètres modélisés et conformes à une élocution humaine. Concernant la problématique de recherche d'une séquence d'unités acoustiques, différentes solutions sont possibles. Les plus connues appliquent des solutions de recherche de meilleurs chemins (Sagisaka 1992) (Hunt 1996) en proposant une hypothèse de programmation dynmique. D'autres travaux (Donovan 1995) ont défini des modèles acoustiques permettant de guider le choix d'une séquence d'unités.

L'enjeu du procédé de sélection est double, (Iwahashi 1992). Il s'agit d'une part de trouver une correspondance entre une sous-séquence de la chaîne phonémique à synthétiser et un exemplaire plausible dans le corpus de référence. On parle alors d'une \emph{discrimination par critères de cible}, (Hune 1996). Une correspondance à la cible ne suffit pas puisque cette décision est prise unité par unité. Il faut un mécanisme supplémentaire garantissant que l'enchaînement du séquencement proposé réponde à des critères de continuité acoustique (de nature segmentale ou supra-segmentale). On parle dans ce cas de critères de concaténation. La difficulté du problème réside dans le fait que les deux critères sont combinés. Le choix d'une sous-séquence en correspondance avec une unité du corpus dépend de son contexte passé (contexte de la séquence à gauche) et à venir (contexte à droite). Il s'agit encore une fois d'un problème de nature combinatoire qui peut formellement être posé comme un problème de recherche d'un meilleur chemin dans un graphe.

La grande majorité des systèmes de synthèse appliquent un algorithme de Viterbi. Cet algorithme, efficace en complexité spatiale et temporelle, tire sa justification du fait que l'expression du coût global d'une séquence d'unités s'écrit, par hypothèse, sous la forme d'une suite récurrente additive. Cette justification est largement partagée par l'ensemble de la communauté pour ce qui est de l'expression des coûts de concaténation et des coûts de proximité à la cible. En revanche pour ce qui concerne la prise en compte de coûts de nature prosodique, une mise en forme recurrente est plus délicate et difficilement justifiable puisque ces phénomènes ont lieu à l'échelle du groupe intonatif et de la phrase.

Nous considérons qu'il est possible de dépasser la qualité des systèmes de synthèse actuels par la prise en compte de critère prosodiques lors de la recherche de la séquence optimale des unités. Tenir compte de ces critières proposidiques n'est pas une chose simple, puiqu'il faut définir de nouveaux modèles de description des coûts acoustiques et prosodiques d'une séquence. Ces nouvelles techniques de sélection devraient être acapables de proposer des voix avec d'une part plus de relief ou d'expressivité tout en maintenant une très bonne qualité sonore.

(Sagisake 1992) : Sagisaka, Y. and Kaiki, N. and Iwahashi, N. and Mimura, K., ATR mu-TALK speech synthesis system, proceedings of the International Conference on Spoken Language Processing (ICSLP'92)", 1992, pp. 483-486.
(Hunt 1996) :  Hunt, A. and Black, A.W., Unit selection in a concatenative speech synthesis system using a large speech database, proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP'96), 1996, pp. 373-376.
(Donovan 1995) :  Donovan, R. and P. Woodland, P., Automatic speech synthesizer parameter estimation using HMMs, proceedings of the IEEE International Conference on Acoustics and Signal Processing (ICASSP'95), 1995, pp. 640-643.
(Iwahashi 1992) : Iwahashi, N. and Kaiki, N. and Sagisaka, Y., Concatenative speech synthesis by minimum distortion criteria, proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP'92), 1992, pp. 65-68.
(Black 1995) :  Alan W. Black and Nick Campbell, Optimizing selection of units from speech databases for concatenative sysnthesis, proceedings of the IEEE International Conference on Acoustics and Signal Processing (ICASSP'95), vol. 1, pp. 581-584.
(Campbell 1996) :  Campbell, N. and Black, A., CHART: A High-definition speech re-sequencing system, in Progress in Speech Synthesis,
eds. van Santen, J. and Sproat, R. and Olive, J. and Hirschberg, J., 1996, pp. 365-381,
(Huang 1996) : Huang, X. and Acero, A. and Adcock, J. and Hon, H.-W. and Goldsmith, J. and Liu, J. and Plumpe, M., Whistler: A trainable text-to-speech system, proceedings of the International Conference on Spoken Language Processing (ICSLP'96), 1996, pp. 2397-2390.
(Toda 2006) :  Tomoki Toda and Hisashi Kawai and Toshio Hirai and Jinfu Ni and Nobuyuki Nishizawa and Junichi Yamagishi and Minoru Tsuzaki and Keiichi Tokuda and Satoshi Nakamura, Developing a test bed of English text-to-speech system XIMERA for the Blizzard challenge 2006,  Blizzard Challenge, 2006.
(Eide 2006) :  Ellen Eide and Raul Fernandez and Ron Hoory and Wael Hamza and Zvi Kons and Michael Picheny and Ariel Sagi and Slava Shechtman and Zhi Wei Shuang, The IBM submission to the 2006 Blizzard text-to-speech challenge, Blizzard Challenge, 2006.

Proposition d'un travail de thèse
======================

Nous proposons de nous intéresser à de nouvelles méthodologies de sélection d'unités acoustiques pour la synthèse de la parole à partir du texte. La proposition de thèse comporte deux volets: un axe de propositions scientifiques permettant de lever certains verrous notamment dans la formulation du coût d'une séquence d'unités, et un axe expérimental par la proposition d'une évolution du système de synthèse du groupe Cordial permettant de mettre en place des évaluations perceptuelles qui permettront de valider ou d'invalider les hypothèses de travail qui auront été choisies. Le travail expérimental sera réalisé sur le français. Nous souhaitons doubler les expérimentations sur l'anglais et participer ainsi au challenge Blizzard qui est une compétition internationale en synthèse de la parole.

Le travail de thèse prendra comme point d'appui la proposition suivante:
  * Mise en place et évaluation d'un premier système reposant sur l'état de l'art actuel en synthèse de la parole par corpus de parole continue. Prise en compte des niveaux acoustiques. Utilisation d'une base de parole expressive,  "chronic",  issue du projet ANR Vivos.
  * Proposition de modèles de sélection de nature prosodique.
  * Propositions algorithmiques, définition d'heuristiques pour une solution acceptable en temps de calcul.
  * Intégration des propositions prosodiques au système de synthèse de référence et évaluation.

Contexte du travail de thèse
===================

L'étudiant sera accueilli au sein de l'équipe Cordial de l'irisa : http://www.irisa.fr/cordial dont les
principaux travaux concernent le traitement de la parole : synthèse, transformation de parole, annotation de corpus.
L'équipe de recherche est hébergée dans les locaux de l'Ecole Nationale Supérieure des Sciences Appliquées et de Technologie,
http://www.enssat.fr, à Lannion. La thèse est financée sur trois ans par une bourse du conseil général des Côtes d'Armor.

__________________________________________________________________________________________________________

Opened positions/internships at Microsoft: Norwegian Linguist (M/F)

MLDC – Microsoft Language Development Center, a branch of the Microsoft Product Group that develops Speech Recognition and Synthesis Technologies, situated in Porto Salvo, Portugal (http://www.microsoft.com/portugal/mldc), is seeking a full-time temporary language expert in the Norwegian language (Bokmal), for a 4-6 month contract, to work in speech technology related development projects. The successful candidate should have the following requirements:

·         Be native or near native Norwegian Bokmal speaker

·         Have a university degree in Linguistics or related field (preferably in Norwegian Linguistics)

·         Have an advanced level of English

·         Have some experience in working with Speech Technology/Natural Language Processing/Linguistics, either in academia or in industry

·         Have some computational ability – no programming is required, but he/she should be comfortable working with MS Windows and MS Office tools

·         Have team work experience

·         Willing to work in Porto Salvo (near Lisbon) for the duration of the contract

·         Willing to start in October 2008

To apply, please submit your resume and a brief statement describing your experience and abilities to Daniela Braga: i-dbraga@microsoft.com

We will only consider electronic submissions.

Deadline for submissions: August 10, 2008 

Opened positions/internships at Microsoft: Finnish Linguist (M/F)

MLDC – Microsoft Language Development Center, a branch of the Microsoft Product Group that develops Speech Recognition and Synthesis Technologies, situated in Porto Salvo, Portugal (http://www.microsoft.com/portugal/mldc), is seeking a full-time temporary language expert in the Finnish language, for a 6 month contract, to work in speech technology related development projects. The successful candidate should have the following requirements:

·         Be native or near native Finnish speaker

·         Have a university degree in Linguistics or related field (preferably in Norwegian Linguistics)

·         Have an advanced level of English (oral and written)

·         Have some experience in working with Speech Technology/Natural Language Processing/Linguistics, either in academia or in industry

·         Have some computational ability – no programming is required, but he/she should be comfortable working with MS Windows and MS Office tools

·         Have team work experience

·         Willing to work in Porto Salvo (near Lisbon) for the duration of the contract

·         Willing to work in a multicultural and multinational team across the globe

·         Willing to start in September 1, 2008

To apply, please submit your resume and a brief statement describing your experience and abilities to Daniela Braga: i-dbraga@microsoft.com

We will only consider electronic submissions.

Deadline for submissions: August 10, 2008

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.

Call for Papers

Recent Advances in Biometric Systems: A Signal Processing Perspective

Biometrics a digital recognition technology that relies on highly distinctive physical and physiological characteristics of an individual is potentially a powerful and reliable method for personal authentication. The increasing importance of biometrics is underscored by the rapidly growing number of educational and research activities devoted to this field; and by a large number of annually organized Conferences and Symposia exclusively devoted to biometrics. Biometrics is a multidisciplinary field with researchers from signal processing, pattern recognition, computer vision, and statistics. Recently, a number of new important directions have been identified for biometric research, including processing and encoding of nonideal data, biometrics at a distance, and data quality assessment. Problems in nonideal biometric data include off-angle, occluded, blurred, and noisy images. Biometrics at a distance is concerned with recognition from video or snapshots of a biometric samples captured from a noncooperative moving individual. The goal of this special issue is to focus on recent advances in signal processing of biometric data that allow improved recognition performance through novel restoration, processing, and encoding; matching techniques capable of dealing with complexity and distortions in data acquired from a distance; recognition from biometric data acquired from unconstrained environments or complex experimental set ups; and the characterization of quality and its relationship with performance.

Topics of interest include, but are not limited to:

Biometric-based recognition under unconstrained presentation and/or complex environment using the following:
    o Face
    o Iris
    o Fingerprint
    o Voice
    o Hand
    o Soft biometrics

Multimodal biometric recognition using nonideal data

Biometric image/signal quality assessment:
    o Face
    o Iris
    o Fingerprint
    o Voice
    o Hand
    o Soft biometrics

Biometric security and privacy
    o Liveness detection
    o Encryption
    o Cancelable biometrics

The special issue will focus both on the development and comparison of novel signal/image processing approaches and on their expanding range of applications.

Authors should follow the EURASIP Journal on Advances in Signal Processing manuscript format described at the journal site http://www.hindawi.com/journals/asp/. Prospective authors should submit an electronic copy of their complete manuscript through the journal Manuscript Tracking System at http://mts.hindawi.com/, according to the following timetable:

Manuscript Due                 October 1, 2008
First Round of Reviews         January 1, 2009
Publication Date               April 1, 2009

Guest Editors

o Natalia A. Schmid, Lane Department of Computer Science and Electrical Engineering, West Virginia University, Morgantown, WV 26506, USA; natalia.schmid@mail.wvu.edu
o Stephanie Schuckers, Electrical &amp; Computer Engineering, Clarkson University, Potsdam, NY 13699, USA; sschucke@clarkson.edu
o Jonathon Phillips, National Institute of Standard and Technology, Gaithersburg, MD 20899, USA; jonathon@nist.gov
o Kevin Bowyer, University of Notre Dame, Notre Dame, IN 46556, USA; kwb@cse.nd.edu 

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 WORKSHOP PROPOSALS EACL 2009, NAACL HLT 2009, AND ACL-IJCNLP 2009

Joint site:  http://www.eacl2009.gr/conference/callforworkshops

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/

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

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

SUBMISSION OF DEMONSTRATION ABSTRACTS 
 
       Authors are invited to present actual projects, developed software and hardware or interesting material relevant to the topics of the conference. The authors of the demonstrations should provide the abstract not exceeding one page as plain text. The submission must be made using an online form available at the conference www pages.  The accepted demonstrations will be presented during a special Demonstration Session (see the Demo Instructions at www.tsdconference.org). 
 
  Demonstrators can present their contribution with their own notebook with an Internet connection provided by the organisers or the organisers can prepare a PC computer with multimedia support for demonstrators.   
 
    IMPORTANT DATES
 
  July 31 2008 ............. Submission of demonstration abstracts
 
 August 7 2008 ............ Notification of acceptance for                            demonstrations sent to the authors
 
 September 8-12 2008 ...... Conference date 
 
 The demonstration abstracts will not appear in the Proceedings of TSD 2008 but they will be published electronically at the conference website.
 
 
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).

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/

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

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

• Language recognition

Registration Form

• Speech synthesis

Registration Form

• Automatic text translation

Registration Form

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

Call for Papers/Demos
  
  2nd International Symposium on Universal Communication Dec 15 - 16,
  2008 Osaka International Convention Center, Osaka, Japan
  
  
  
  The development of information network systems enables us to
  communicate with people in remote places "anytime and anywhere",
  enriching human knowledge, affection and sensibility. However, there
  are various barriers which stand in our way to using these systems
  freely and flexibly. In order to discuss how to overcome these
  barriers and create a more human-centered communication environment,
  in 2007 the first International Symposium on Universal Communication
  was held in Kyoto, Japan featuring discussions with well-known
  researchers from around the world. Following its success, the second
  International Symposium on Universal Communication will be held in
  Osaka, Japan.
  
  
  
  Topics of interest are as follows, but not limited to
  
  　- Information retrieval and information analysis
  
  　- Information credibility
  
  　- Knowledge processing
  
  　- Language resources
  
  　- Speech recognition and synthesis
  
  　- Machine translation and speech translation
  
  　- Natural language processing
  
  　- Spoken language processing
  
  　- Multilingual information processing
  
  　- Super high-resolution image technology
  
  　- 3D visualization, imaging and display technologies
  
  　- 3D sound processing
  
  　- Virtual reality, mixed reality and augmented reality
  
  　- Multisensory (visual, acoustic, haptic, olfactory, etc) interaction
  
  　- Human factors
  
  　- Human interface and interaction technologies
  
  　- Real-world sensing technologies
  
  
  
  General Chair
  
  Yuichi Matsushima, National Institute of Information and
  Communications Technology (NICT), Japan
  
  
  
  General Vice Chairs
  
  Kazumasa Enami, NICT, Japan
  
  Hiromitsu Wakana, NICT, Japan
  
  
  
  Technical Program Committee
  
  Chair: Satoshi Nakamura, NICT/ATR, Japan Vice Chair: Naomi Inoue,
  NICT/ATR, Japan
  
  - Akio Ando, NHK Science & Technical Research Laboratories, Japan
  
  - Martin S. Banks, UC Berkeley, USA
  
  - Khalid Choukri, ELDA, France
  
  - Marcello Federico, FBK/IRST, Italy
  
  - Sidney Fels, The University of British Columbia, Canada
  
  - Jukka Häkkinen, University of Helsinki, Finland
  
  - Munpyo Hong, Sungkyunkwan University, Korea
  
  - Kentaro Inui, Nara Institute of Science and Technology, Japan
  
  - Hitoshi Isahara, NICT, Japan
  
  - Ken Kaneiwa, NICT, Japan
  
  - Takashi Kawai, Waseda University, Japan
  
  - Yutaka Kidawara, NICT, Japan
  
  - Kyeong Soo Kim, Swansea University, UK
  
  - Hisashi Miyamori, Kyoto Sangyo University, Japan
  
  - Makoto Okui, NICT, Japan
  
  - Tanja Schultz, Carnegie Melon University, USA
  
  - Yasuyuki Sumi, Kyoto University, Japan
  
  - Eiichiro Sumita, NICT/ATR, Japan
  
  - Yoiti Suzuki, Tohoku University, Japan
  
  - Yasuhiro Takaki, Tokyo University of Agriculture and Technology, Japan
  
  - Kazuya Takeda, Nagoya University, Japan
  
  - Kentaro Torisawa, NICT, Japan
  
  - Chiu-yu Tseng, Institute of Linguistics Academia Sinica, Taiwan
  
  - Kiyotaka Uchimoto, NICT, Japan
  
  - Takehito Utsuro, University of Tsukuba, Japan
  
  - Andy Way, Dublin City University, Ireland
  
  - Andrew Woods, Curtin University of Technology, Australia
  
  - Wieslaw Woszczyk, McGill University, Canada
  
  - Xing Xie, Microsoft Research Asia, China
  
  - Tatsuya Yamazaki, NICT, Japan
  
  - Kwon Yongjin, Korea AeroSpace University, Korea
  
  - Daqing Zhang, Institut TELECOM & Management SudParis, France
  
  
  
  Demo Chair
  
  Kazuhiro Kimura, NICT, Japan
  
  
  
  Technical Advisors
  
  Takashi Matsuyama, Kyoto University, Japan Michitaka Hirose, Tokyo
  University, Japan
  
  
  
  Local Arrangement Co-Chairs
  
  Kazuhiro Kimura, NICT,
  
  Yukio Takahashi, NICT
  
  Contact info: isuc2008@khn.nict.go.jp
  
  
  
  Website
  
  http://www.is-uc.org/2008/
  
  
  
  Venue
  
  Osaka International Convention Center
  
  3-51 Nakanoshima 5-chome,Kita-ku,Osaka, Japan
  http://www.gco.co.jp/english/english.html
  
  
  
  Submission Information
  
  All papers must be submitted through the ISUC 2008 homepage at
  http://www.is-uc.org/2008.
  
  
  
  Paper submission:
  
  The extended abstract must be written in English and is limited to 2
  pages in the IEEE 2-column format. This abstract should also indicate
  whether it is an oral paper or a poster, and be submitted in PDF. Once
  accepted, the camera-ready paper shall be 4~8 pages, not exceeding 8
  pages.
  
  More detailed author guidelines are available at
  http://www.is-uc.org/2008/submission.
  
  
  
  All accepted papers will appear in the conference proceedings
  published by the IEEE Computer Society and will be included in the
  IEEE-Xplore and the IEEE Computer Society (CSDL) digital libraries as
  well as indexed through IET INSPEC, EI (Compendex) and Thomson ISI.
  
  
  
  Demo submission:
  
  Please submit a one-page description of your demo in PDF. This
  description must be written in English and should include: An abstract
  of what you will show, Space needed, Facilities needed including power
  supply and Internet access. A specified submission format will be
  available on the ISUC 2008 homepage.
  
  
  
  Important Dates
  
  Papers:
  
  Extended Abstract due          July 25, 2008
  
  Notification of acceptance      August 29, 2008
  
  Camera-ready papers due      September 26, 2008
  
  
  
  Demos:
  
  Submission of description due September 19, 2008
  
  Notification of acceptance      October 10, 2008