Kindly be reminded that due to ISCA copyright issues, proceedings of the Interspeech conferences should not be put online for public access. The complete ISCA electronic archive, including over 100 proceedings of Interspeech (and former Eurospeech and ICSLP) conferences and ISCAworkshops is available to all members.   Thank you for your attention. 

Prof  Helen Meng,  ISCA publications.

We are concerned about these emails and want to alert you to this. We also want to protect our community as well as helping you to recognize fraudulent and/or phishing emails.

The fraudulent e-mail messages currently in circulation, generally contain "Manuscript Submission" or "Call for Papers" in the subject line and are typically sent using e-mail accounts supported by Gmail, Hotmail or by other free e-mail providers. Typically, the body of these messages contain a "Call for Papers," requesting that authors submit scholarly articles via e-mail for publication by Elsevier in various Elsevier journals and other publications.  Ultimately, these fraudulent e-mails involve a request for the victims to send "handling fees" to cover the processing of the article that has been submitted. 

Please be assured that Elsevier, is in no way associated with this fraudulent e-mail campaign.  Elsevier is currently investigating this fraud to identify the persons responsible and to bring them to justice.

In addition, please be advised that Elsevier does not solicit intellectual property from authors in this fashion, and does not utilize Gmail, Hotmail, or any other free third-party e-mail providers in communications with authors and editors. 

If you receive any e-mail messages that appear to be a part of this fraudulent solicitation, DO NOT respond to the message and do not open any attachments contained in the message.  Rather, please forward the message to Elsevier's Fraud Department at emailabuse@elsevier.com. We will use the information included in the message to aid in our investigation. If you know of someone who has received this message, please pass along the above information and ask them also to forward the message to the Elsevier's Fraud Department.

Thank you for your understanding and your cooperation. 

Kind regards,
Author support/communications team

An ISCA Tutorial and Research Workshop on NON-LINEAR SPEECH PROCESSING (NOLISP'09) 25/06/2009 - // DeadLine: 20090315 Vic Catalonia Espagne http://nolisp2009.uvic.cat  After the success of NOLISP'03 held in Le Croisic, NOLISP'05 in Barcelona and NOLISP'07 in Paris, we are pleased to present NOLISP'09 to be held at the University of Vic (Catalonia, Spain) on June 25-27, 2009. The workshop will feature invited lectures by leading researchers as well as contributed talks. The purpose of NOLISP'09 is to present and discuss novel ideas, works and results related to alternative techniques for speech processing, which depart from mainstream approaches.  Prospective authors are invited to submit a 3 to 4 page paper proposal in English, which will be evaluated by the Scientific C! ommittee. Final papers will be due one month after the workshop to be included in the CD-ROM proceedings. Contributions are expected (but not restricted to) the following areas:  Non-linear approximation and estimation Non-linear oscillators and predictors Higher-order statistics Independent component analysis Nearest neighbours Neural networks Decision trees Non-parametric models Dynamics of non-linear systems Fractal methods Chaos modelling Non-linear differential equations  All fields of speech processing are targeted by the workshop, namely: Speech production, speech analysis and modelling, speech coding, speech synthesis, speech recognition, speaker identification/verification, speech enhancement/separation, speech perception, etc.  ADDITIONAL INFORMATION  Proceedings will be published in Springer-Verlag's Lecture Notes Series in Computer Science (LNCS). LNCS is published, in parallel to the printed books, in full-text electronic form. All contributions should  be original, and must not have been previously published, nor be under review for presentation elsewhere. A special issue of Speech Communication (Elsevier) on “Non-Linear and Non-Conventional Speech Processing” will be also published after the workshop  Detailed instructions for submission to NOLISP'09 and further informations will be available at the conference Web site (http://nolisp2009.uvic.cat).  IMPORTANT DATES: * March 15, 2009 - Submission (full papers) * April 30, 2009 - Notification of acceptance * September 30, 2009 - Final (revised) paper

September 6-10, 2009, Brighton, UK,
Conference Website
Chairman: Prof. Roger Moore, University of Sheffield.

INTERSPEECH 2009 will take place 6th-10th September 2009 in Brighton, UK (http://www.interspeech2009.org/).  The theme of IS09 will be 'Speech and Intelligence' and, as well as the usual programme of scientific sessions, we are also organising a series of events related to the theme.  For example, IS09 will be hosting the 2009 annual contest for the 'Loebner Prize' (http://loebner.net/Prizef/loebner-prize.html) and, for the first time, we are planning to mount a speech-based version of the competition.  Also, in order to raise awareness of speech science and technology in the general public, we are hoping to be able to put together an exhibition of speech-related demonstrations/displays.  We envisage that these will provide hands-on interaction with our current technologies, and will serve to promote our field to the wider population.  Before committing significant resource to organising these events, we would like to have some idea of the level of interest/support in the community. So, we would very much like to ask the following simple questions ...    Q1.  Would you be interested in entering a system for a speech-based version of the Loebner competition?: YES/NO    Q2.  Would you be interested to provide an exhibit for a public display?: YES/NO  ... if you have answered YES to either of these questions, please reply to Simon Worgan (simon.worgan@gmail.com) who will collate the responses and contact you with further information.  Best wishes  Roger K. Moore General Chair: INTERSPEECH 2009

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

 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

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

Opened positions/internships at Microsoft: Norwegian, Finnish, Italian, French, English, Polish Linguists (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 full-time temporary language experts in the following languages:

-          Norwegian language (Bokmal variety)

-          Finnish

-          Italian

-          English (UK)

-          French (France)

-          Polish

The contracts range from 2-6 months and the scope of them is to work in speech technology related development projects. The successful candidate should have the following requirements:

·         Be a native or near native speaker (for each of the required language)

·         Have a university degree in Linguistics or related field

·         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 immediately

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: Opened until filled.

Title:      Assistant Professor Position at EURECOM
            in Multimedia content analysis and processing
Department: Multimedia Communications
URL:        http://www.eurecom.fr/research/
Start date: September 2008

Description:
Research in the Department currently focuses on several aspects of the processing of Multimedia Content:
• Video analysis and information filtering,
• Image Processing with application to 3D Face Cloning, watermarking and
biometrics,
• Speech and sound processing.
The new faculty is expected to undertake research in close collaboration with the existing activities and to participate in the teaching program for Master students. Extensions of the current research areas are encouraged.

Requirements:
The candidates must have a Ph.D. in computer science or electrical engineering with a solid background in signal processing and statistical analysis. The ideal candidate will have an established record of conducting research activities at the international level, and a proven record of collaboration with academic and industrial partners in national and European programs or equivalent. Excellence in research is a constant requirement for EURECOM. A strong commitment to excellence in research and teaching is essential.

Applications:
Send, by email, a letter of motivation, a resume including a list of publications, the names of 3 references and a copy of the three most important publications.

Contact:        Prof. Bernard Merialdo
Postal address: 2229 route des Crêtes
                BP 193
                06904 Sophia Antipolis cedex
                France
Email address:  Bernard.Merialdo@eurecom.fr
Web address:    http://www.eurecom.fr/main/institute/job.fr.htm
Phone:          +33/0 4 93 00 81 29
Fax:            +33/0 4 93 00 82 00

Located in the heart of Sophia Antipolis technology park, EURECOM is a graduate school and a Research center in Communication Systems, founded in 1991 by TELECOM ParisTech (Ecole Nationale Supérieure des Télécommunications) and EPFL (Swiss federal institute of Lausanne) in a consortium form, combining academic and industrial partners. Teaching and research activities of EURECOM focus on three areas: networking and security, mobile communications and multimedia communications. EURECOM has a strong international scope and strategy. 

Dans le cadre du projet RAPSODIS, l'IRISA recrute un post-doc pour une période de 12 mois. Le sujet porte sur l'intégration de connaissances syntaxiques et sémantiques dans un système de transcription automatique de la parole (voir les détails en fin de message). Le poste est à pourvoir d'ici la fin de l'année. Les personnes intéressés sont invités à contacter Guillaume Gravier (guig@irisa.fr) et/ou Pascale Sébillot (pascale.sebillot@irisa.fr).

SENIOR STAFF OPENING AT THE ICSI SPEECH GROUP

The International Computer Science Institute (ICSI) invites applications for a senior staff position in its speech research group. Successful applicants must have significant post-PhD experience, and a world-class research reputation.  Candidates must also have demonstrated ability to grow and manage a strong research effort. A successful track record with obtaining funding for the chosen area is essential. 

The ICSI Speech Group (including its predecessor, the ICSI Realization Group) has been a source of novel approaches to speech processing since 1988. ICSI’s speech group is well known for its efforts in speech recognition (particularly in neural network approaches and novel forms of feature extraction), as well as in speaker recognition, diarization, and speech understanding. It has close ties to research efforts in machine translation on the Berkeley campus, and to the STAR lab at SRI for the complete range of its research. It also works closely with several European labs, particularly IDIAP in Switzerland and to the University of Edinburgh.

ICSI is an independent not-for-profit Institute located a few blocks from the Berkeley campus of the University of California. It is closely affiliated with the University, and particularly with the Electrical Engineering and Computer Science (EECS) Department. Students, faculty, and administrative colleagues from the University all play a key role at the Institute. In addition to its Speech Group, areas of current strength in the Institute include: Artificial Intelligence (primarily in natural language), Internet research (primarily in architecture and internet security), and Algorithms (primarily associated with problems in bioinformatics and networking). We also have new activities in Computer Vision and Computer Architecture. See http://www.icsi.berkeley.edu to learn more about ICSI.

Applications should include a cover letter, a vita, the names of at least 3 references (with both postal and email addresses), and a research statement. Applications should be sent by email to speechjob@icsi.berkeley.edu and by postal mail to

Nelson Morgan (re Senior Search)

ICSI

1947 Center Street, Suite 600

Berkeley, CA 94704

ICSI is an Affirmative Action/Equal Opportunity Employer. Applications from women and minorities are especially encouraged. Hiring is contingent on eligibility to work in the United States.

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 & 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 

Call for Papers

Special issue of Eurasip journal on advanced signal processing

Analysis and Signal Processing of Oesophagial and Pathological Voices

Speech is the most important means of communication among humans. Speech, however, is not limited only to the process of communication, but is also very important for transferring emotions, expressing our personality, and reflecting situations of stress. Modern lifestyles have increased the risk of experiencing some kind of voice alteration. It is estimated that around 19% of the population suffer or have suffered from dysphonic voicing. This motivates new and objective ways to evaluate speech, its quality, and its connection with other phenomena.

Speech research to date has favored areas such as synthesis, recognition, and speaker verification. The last few years have witnessed the emerging topic of processing and evaluation of disordered speech. Acoustic analysis is a noninvasive technique providing an efficient tool for the objective diagnosis, the screening of voice diseases, the objective determination of vocal function alterations, and the evaluation of surgical treatment and rehabilitation. Its application extends beyond medicine, and now includes forensic analysis as well as voice quality control for voice professionals. Acoustic analysis may also be seen as complementary to other methods of evaluation based on the direct observation of the vocal folds using videoendoscopy.

This special issue aims to foster an interdisciplinary forum for presenting new work in the analysis and modeling of voice signals and videoendoscopic images, with applications in pathological and oesophageal voices.

Topics of interest include (but are not limited to):

o Automatic detection of voice disorders
o Automatic assessment and classification of voice quality
o New strategies for the parameterization and modeling of normal and pathological voices (biomechanical-based parameters, chaos modeling, etc.)
o Databases of vocal disorders
o Inverse filtering
o Signal processing for remote diagnosis
o Speech enhancement for pathological and oesophageal voices
o Objective parameters extraction from vocal fold images using videolaryngoscopy, videokymography, fMRI, and other emerging techniques
o Multimodal analysis of disordered speech
o Robust pitch extraction algorithms for pathological and oesophageal voices

Since speech communication is fundamental to human interaction, we are moving towards a new scenario where speech is gaining greater importance in our daily lives, and many common speech disorders and dysfunctions would be treated using computer-based or physical prosthetics.

Authors should follow the EURASIP Journal on Advances in Signal Processing manuscript format described at 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 tentative timetable:

Manuscript Due                 November 1, 2008
First Round of Reviews         February 1, 2009
Publication Date               May 1, 2009

Guest Editors

o Juan I. Godino-Llorente, Department of Circuits and Systems Engineering, Polytechnic University of Madrid (UPM), Ctra. de Valencia, 28031 Madrid, Spain; igodino@ics.upm.es
o Pedro Gómez-Vilda, Department of Computer Science and Engineering, Polytechnic University of Madrid (UPM), Boadilla del Monte, 28660 Madrid, Spain; pedro@pino.datsi.fi.upm.es
o Tan Lee, Department of Electronic Engineering, The Chinese University of Hong Kong, Shatin, New Territories, Hong Kong; tanlee@ee.cuhk.edu.hk 

Special Issue on "Emergent Artificial Intelligence Approaches for Pattern Recognition in Speech and Language Processing"

      Computer Speech and Language, Elsevier       Deadline for paper submission: September 26, 2008.  http://speechlab.ifsc.usp.br/call/csl.pdf                        =

Special Issue of IEEE Transactions on Audio, Speech and Language Processing **SIGNAL MODELS AND REPRESENTATION OF MUSICAL AND ENVIRONMENTAL SOUNDS**  http://www.ewh.ieee.org/soc/sps/tap http://www.ewh.ieee.org/soc/sps/tap/sp_issue/audioCFP.pdf * *-- Submission deadline: 15 December, 2008 --  Notification of acceptance: 15 June, 2009 Final manuscript due: 1st July, 2009 Tentative publication date: 1st September, 2009   Guest editors Dr. Bertrand David (Telecom ParisTech, France)  bertrand.david@telecom-paristech.fr Dr. Laurent Daudet (UPMC University Paris 06, France) daudet@lam.jussieu.fr Dr. Masataka Goto (National Institute of Advanced Industrial Science and  Technology, Japan) m.goto@aist.go.jp Dr. Paris Smaragdis (Adobe Systems, Inc, USA) paris@adobe.com ------------------------------------------------------------------------------    The non-stationary nature, the richness of the spectra and the mixing of  diverse sources are common characteristics shared by musical and  environmental audio scenes. It leads to specific challenges of audio  processing tasks such as information retrieval, source separation,  analysis-transformation-synthesis and coding. When seeking to extract  information from musical or environmental audio signals, the  time-varying waveform or spectrum are often further analysed and  decomposed into sound elements. Two aims of this decomposition can be  identified, which are sometimes antagonist: to be together adapted to  the particular properties of the signal and to the targeted application.  This special issue is focused on how the choices of a low level  representation (typically a time-frequency distribution with or without  probabilistic framework, with or without perceptual considerations), a  source model or a decomposition technique may influence the overall  performance.  Specific topics of interest include but are not limited to: * factorizations of time-frequency distribution * sparse representations * Bayesian frameworks * parametric modeling * subspace-based methods for audio signals * representations based on instrument or/and environmental sources  signal models * sinusoidal modeling of non-stationary spectra (sinusoids, noise,  transients)  Typical applications considered are (non exclusively): * source separation/recognition * mid or high level features extraction (metrics, onsets, pitches, …) * sound effects * audio coding * information retrieval * audio scene structuring, analysis or segmentation * ...  B. David, L.Daudet, P. Smaragdis, M. Goto.

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

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

The Association for Computational Linguistics invites proposals for
workshops to be held in conjunction with one of the three flagship
conferences sponsored in 2009 by the Association for Computational
Linguistics: ACL-IJCNLP 2009, EACL 2009, and NAACL HLT 2009.  We solicit
proposals on any topic of interest to the ACL community. Workshops will
be held at one of the following conference venues:

EACL 2009 is the annual meeting of the European chapter of the ACL. The
conference will be held in Athens, Greece, March 30-April 3 2009;
workshops March 30-31.

NAACL HLT 2009 is the annual meeting of the North American chapter of
the ACL.  It continues the inclusive tradition of encompassing relevant
work from the natural language processing, speech and information
retrieval communities.  The conference will be held in Boulder,
Colorado, USA, from May 31-June 5 2009; workshops will be held June 4-5.

ACL-IJCNLP 2009 combines the 47th Annual Meeting of the Association for
Computational Linguistics (ACL 2009) with the 4th International Joint
Conference on Natural Language Processing (IJCNLP).  The conference will
be held in Singapore, August 2-7 2009; workshops will be held August 6-7.


    SUBMISSION INFORMATION

In a departure from previous years, ACL-IJCNLP, EACL and NAACL HLT will
coordinate the submission and reviewing of workshop proposals for all
three ACL 2009 conferences.

Proposals for workshops should contain:

    * A title and brief (2-page max) description of the workshop topic
      and content.
    * The desired workshop length (one or two days), and an estimate
      of the audience size.
    * The names, postal addresses, phone numbers, and email addresses
      of the organizers, with one-paragraph statements of their
      research interests and areas of expertise.
    * A budget.
    * A list of potential members of the program committee, with an
      indication of which members have already agreed.
    * A description of any shared tasks associated with the workshop.
    * A description of special requirements for technical needs.
    * A venue preference specification.

The venue preference specification should list the venues at which the
organizers would be willing to present the workshop (EACL, NAACL HLT, or
ACL-IJCNLP).  A proposal may specify one, two, or three acceptable
workshop venues; if more than one venue is acceptable, the venues should
be preference-ordered.  There will be a single workshop committee,
coordinated by the three sets of workshop chairs.  This single committee
will review the quality of the workshop proposals.  Once the reviews are
complete, the workshop chairs will work together to assign workshops to
each of the three conferences, taking into account the location
preferences given by the proposers.

The ACL has a set of policies on workshops. You can find general
information on policies regarding attendance, publication, financing,
and sponsorship, as well as on financial support of SIG workshops, at
the following URL:
http://www.cis.udel.edu/~carberry/ACL/index-policies.html

Please submit proposals by electronic mail no later than September 1
2008, to acl09-workshops at acl09-workshops@uni-konstanz.de with the
subject line: "ACL 2009 WORKSHOP PROPOSAL."


    PRACTICAL ARRANGEMENTS

Notification of acceptance of workshop proposals will occur no later
than September 23, 2008.  Since the three ACL conferences will occur at
different times, the timescales for the submission and reviewing of
workshop papers, and the preparation of camera-ready copies, will be
different for the three conferences. Suggested timescales for each of
the conferences are given below.

ALL CONFERENCES
Sep 1, 2008     Workshop proposal deadline
Sep 23, 2008    Notification of acceptance of workshops

EACL 2009
Sep 30, 2008    Call for papers issued by this date
Dec 12, 2008    Deadline for paper submission
Jan 23, 2009    Notification of acceptance of papers
Feb  6, 2009    Camera-ready copies due
Mar 30-31, 2009 EACL 2009 workshops

NAACL HLT 2009
Dec 10, 2008    Call for papers issued by this date
Mar 6, 2009     Deadline for paper submissions
Mar 30, 2009    Notification of paper acceptances
Apr 12, 2009    Camera-ready copies due
June 4-5, 2009  NAACL HLT 2009 workshops

ACL-IJCNLP 2009
Feb 6, 2009     Call for papers issued issued by this date
May 1, 2009     Deadline for paper submissions
Jun 1, 2009     Notification of acceptances
Jun 14, 2009    Camera-ready copies due
Aug 6-7, 2009   ACL-IJCNLP 2009 Workshops

Workshop Co-Chairs:

    * Miriam Butt, EACL, University of Konstanz
    * Stephen Clark, EACL, Oxford University
    * Nizar Habash, NAACL HLT, Columbia University
    * Mark Hasegawa-Johnson, NAACL HLT, University of Illinois at
Urbana-Champaign
    * Jimmy Lin, ACL-IJCNLP, University of Maryland
    * Yuji Matumoto, ACL-IJCNLP, Nara Institute of Science and Technology

For inquiries, send email to: acl09-workshops at
acl09-workshops@uni-konstanz.de. 

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

Jeudi 13 novembre 2008, 13h30 - Séminaire externe
========================================
Annie RIALLAND
Laboratoire de Phonétique et Phonologie
Paris

Zones linguistiques en Afrique : arguments phonologiques et prosodiques

Une hypothèse centrale des travaux que nous menons actuellement en collaboration avec G. N. Clements (Clements & Rialland 2008) est que le continent africain peut être divisé en six grandes zones linguistiques, chacune d'entre elles étant définie par un nombre de propriétés phonologiques typologiquement peu fréquentes. Une des zones les plus remarquables est la ceinture soudanique recouvrant la vaste savane qui traverse l'Afrique sub-saharienne de l'océan Atlantique aux hauts plateaux éthiopiens et qui coïncide en grande partie avec la “core area” reconnue par Greenberg (1959).
Nous avons cartographié un certain nombre de propriétés (sons, prosodies) présentes dans des langues africaines. Ces propriétés sont soit segmentales (occlusives labio-vélaires, voyelles nasales, consonnes implosives et clicks), soit prosodiques (systèmes tonals de trois tons et plus, prosodie de question “relâchée”). Nous avons sélectionné en priorité des traits bien documentés dans un grand nombre de langues, qu'on trouve dans des langues génétiquement distantes parlées dans une même aire et qui apparaissent moins fréquemment hors de l'aire en question et hors de l'Afrique de façon générale. Les analyses proposées sont étayées par des tableaux statistiques et/ou des cartes.
Dans cette présentation, nous accorderons une attention particulière aux faits prosodiques. Nous considérerons d’abord la répartition géographique des systèmes tonals à trois tons et plus, qui ont des distributions aréales impliquant le plus souvent plusieurs familles de langues, ainsi que les mécanismes d’émergence des systèmes à quatre et cinq tons dans les zones de langues à trois tons.
Nous avons pu également dégager une distribution aréale de certaines intonations de question. Une typologie des intonations de question oui/non en Afrique, fondée sur une base de données de 130 langues, sera présentée (Rialland 2007). Un faisceau de marqueurs intonatifs (contour mélodique descendant, allongement vocalique, voyelle basse finale et terminaison soufflée, produite par une ouverture progressive de la glotte) apparaissant soit individuellement soit en combinaison dans de nombreuses langues de la zone soudanique retiendra notre attention. Nous faisons l’hypothèse d’une origine commune pour ces formes d’intonation: une intonation de question descendante « relâchée », à l’opposé de l’intonation de question montante prototypique bien connue dans les langues indo-européennes.
Nous présenterons les études phonétiques que nous avons menées sur ces marqueurs de question, en particulier dans des langues voltaïques (moba, ncam, dagara, kabiye), utilisant des méthodes d’investigation physiologiques (mesure de débit d’air, EGG, fibroscopie) pour caractériser les qualités de voix et les mécanismes laryngaux en jeu.
Des cartes montreront la distribution géographique de ces intonations et de leurs variations à l’intérieur de la zone soudanique. Elles se trouvent surtout dans les langues du phylum niger-congo, en particulier au coeur même de cette zone c'est-à-dire dans les familles voltaïque, kwa et kru mais sont aussi présentes dans quelques langues du phylum nilo-saharien, en particulier dans les langues Soudaniques centrales et dans la famille tchadique du phylum afro-asiatique (Rialland, à paraître).

Références :
Clements , George N. & Rialland Annie. 2008. Africa as a phonological area. In Bernd Heine & Nurse Derek, eds, A Linguistic Geography of Africa. Cambridge University Press, pp. 36-85
Rialland Annie. À paraître. The African “lax”question prosody. Lingua
Rialland Annie. 2007. Question prosody : an African perspective, in Tones and Tunes: Studies in Word and Sentence Prosody, C. Gussenhoven and T. Riad (eds.), Mouton de Gruyter, Berlin, 2007, pp. 35-62

Salle de réunion du Département Parole et Cognition (B314)
3ème étage Bâtiment B ENSE3
961 rue de la Houille Blanche
Domaine Universitaire

Attention le Séminaire externe prévu le 20 novembre est reporté au 12 février 2009
========================================
Thierry NAZZI
Laboratoire Psychologie de la Perception
Paris

Titre à préciser

Résumé à préciser

Salle de réunion du Département Parole et Cognition (B314)
3ème étage Bâtiment B ENSE3
961 rue de la Houille Blanche
Domaine Universitaire

Vendredi 28 novembre 2008, matin –
Séminaire commun avec la formation Industries De la Langue
========================================
Cédric GENDROT
Laboratoire de Phonétique et de Phonologie de Paris

Contributions du traitement automatique de la parole à l’analyse phonétique des voyelles.

Dans cette présentation, les résultats de publications effectuées en collaboration avec Martine Adda-Decker, Audrey Bürki et Cécile Fougeron seront développés. Les données de plusieurs dizaines d'heures de parole journalistique du français et de 7 autres langues (allemand, anglais américain, arabe, espagnol, italien, mandarin, portugais européen) ont été analysées automatiquement. Ces données sont principalement issues du système de transcription automatique du LIMSI. Les méthodes d’investigation sont ici fondamentalement inversées en comparaison des études réalisées sur des corpus linguistiques construits ad-hoc, puis segmentés et analysés manuellement. Trois questions méthodologiques se posent en premier lieu :

1- Les frontières des différents phonèmes ainsi segmentés automatiquement sont-elles fiables par rapport à une segmentation humaine ? Une segmentation humaine est-elle naturellement irréprochable ?
2- Les mesures acoustiques automatiques (f0, formants, amplitude) effectuées sont-elles fiables pour des analyses phonétiques ? Les mesures erronées ne fournissent-elles pas des indications utiles ?
3- Les différentes sources de variation dans un corpus de parole continue doivent être prises en compte dans les analyses (par exemple la quantité très importante de mots outils qui diffèrent des mots lexicaux, la réalisation des /y/ est influencée par la forte présence de "tu" dans certains corpus d’interaction, le /œ/ est fréquemment suivi d'un /r/ en français ce qui modifie sa réalisation, etc.). Ces sources de variations sont également riches en informations puisqu'elles font partie des caractéristiques de la langue.

Après ces précautions méthodologiques, différentes études phonétiques réalisées sur ces données conséquentes ont permis de revisiter et de re-préciser des théories sur la réalisation des voyelles en français et différentes langues, habituellement réalisées sur une petite quantité de données et de locuteurs.

4- Comment prédire la réalisation spectrale des voyelles en fonction de leur durée et de l’influence du contexte consonantique ? Les analyses précédemment effectuées dans la littérature se contentent généralement d’étudier des contextes consonantiques symétriques, dont certains sont très peu fréquents en parole.
5- Des interactions importantes entre segmental et suprasegmental sont mises en évidence pour tous les phonèmes : réalisation du phonème en fonction de sa position au sein de la syllabe, du mot, du groupe intonatif...
6- Certains phonèmes sont considérés comme plus résistants à la coarticulation et donc moins variables acoustiquement (/i/ et /y/ par opposition à /a/ par exemple). Nous montrons que certaines mesures souvent négligées mettent en évidence des variations importantes pour ces phonèmes (mouvements du troisième formant de /i/ et /y/ pour l’exemple précédent).
7- Les théories phonologiques prédisent une réalisation acoustique similaire entre schwa et /œ/ en français. Nos études ont pu mettre en valeur des réalisations spectrales distinctes et différentes stratégies pour les locuteurs analysés.
8- Comparaisons inter-langues : confrontation à nos données de théories sur la prédiction des systèmes vocaliques (la théorie quantique, la théorie de la dispersion adaptative et la théorie de la focalisation-dispersion).
9- Discussion sur l'utilité de ces résultats en Reconnaissance Automatique de la Parole et en synthèse de la parole, et sur les limites du traitement automatique pour l’analyse phonétique.

Amphi 9
Université Stendhal
Domaine Universitaire – avenue Centrale

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
  
  
  
  
  

Consonant Challenge for Indian Languages

Detection and Recognition of Consonants in Indian Language Speech Data

 Call for Participation (linked to SLT 2008)

In order to promote speech technology research in Indian Languages and to better understand any specific issues related to speech recognition of these languages and the possible means to address them, we are pleased to announce a Consonant Challenge in Indian Languages. The task involves detection of consonants (in CV, VC, CVC and VCV positions) in a surprise language. Training data is provided in 6 Indian languages, namely, Assamese, Bengali, Hindi, Marathi, Tamil and Telugu to all registered participants.

Based on the recognition results received by the organizers and evaluated by the program committee, the highest two accuracy results will be awarded a cash prize of USD 500 and USD 250 respectively.

The results will be presented in a special session at SLT 08 in Goa, India.

Background

Consonant detection in speech by a machine based on purely spectral features is always problematic due to a number of reasons like the unvoiced (no-energy) portions of stop consonants that can be confused with real silence, the high energy fricative noise that maybe confused with environmental or additive noise, and the vowel like spectrum of the liquids, the nasals and the semi-vowels that make them hard to distinguish from vowels. This problem is further compounded in Indian languages where the number of consonants can go from around 23 (in Tamil) to almost 40 (in Hindi-Urdu).  For example, acoustic phonetic features like voice and aspiration form a four way contrast in many Indian language stop and affricate consonants. Further, stop consonants occur for at least four, that is, labial, dental, retroflex, and velar, place of articulation (this can go to 5 or 6 for some languages like Malayalam and Hindi-Urdu). Though all Indian Languages come from four major language families (Indo-Aryan, Dravidian, Austronesian and Tibeto-Burman, with the majority from the former two), the languages have co-existed for a long time to have borrowed and shared features even at the phonetic level. For example, the borrowing of retroflex sounds from Dravidian to Indo-European and of aspiration as a feature of stops the other way around.

From a Speech Recognition perspective, a deeper understanding of how consonants are detected and recognized can not only help us better understand how to model these sounds (ref. difference between human and computer consonant recognition) but also, in the specific case of Indian languages, open up research issues  into model adaptation from one language to another (related)language. This might allow researchers to explore ways and means to scale from one language to another where resources in terms of training data are limited

Register

* To register for the CCIL, please mail the organizing chairs by 25th August 2008.

* Data will be released to the registered participants ONLY

* All participants for the CCIL will have to register for the main SLT08 workshop.

Important Dates

* Registration for CCIL : 25th August 2008

* Training Data release to registered participants: 29th August 2008

* Test Data in surprise language made available:  8th September 2008

* Recognition results and paper submission: 29th September 2008

* Results Announced: 20th October 2008

* Camera-ready paper submission: 3rd November 2008

Contact

Please mail the organizing chairs to register for the challenge at:

ramkiag@ee.iisc.ernet.in

kalikab@microsoft.com

http://ragashri.ee.iisc.ernet.in/ILCC

 2008 International Symposium on

         Chinese Spoken Language Processing (ISCSLP 2008)

                          December 16 - 19, 2008

                              Kunming, China

                       http://www.iscslp2008.org

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

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

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

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

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

The Keynote speakers of ISCSLP2008 is as following:

Qiang Huo

Microsoft Research Asia, Beijing, China

Research Area

Automatic speech & speaker recognition and related multidisciplinary research topics

Chinese character recognition

Biometric authentication

Document analysis and recognition

Machine learning, etc.

Shigeki Sagayama

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

Research Area

Speech and spoken language processing

Signal processing

Music signal/Information processing

Hand-written character recognition

Multimedia Information processing, etc.

Vincent Vanhoucke

Google, USA

Research Area

Software engineering

Text recognition

Speech recognition

Image processing

Face recognition, etc.

Yuqing Gao

IBM T. J. Watson Research Center, USA

Research Area

Speech recognition

understanding and Translation Research

The large vocabulary continuous speech dictation system

Speech-to-speech translation research, etc.

Hideki Kawahara

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

Research Area

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

Signal processing models of hearing and neural networks

Interaction between speech perception and production, etc.

Yu Hu

Research Director of iFLYTEK, Hefei, China

Research Area

Speech pronunciation evaluation

Speech pronunciation defect detection, etc

Paper Submission

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

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

Each submission will be reviewed by two or more reviewers.

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

Schedule

Full paper submission by Jun. 29, 2008

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

Notification of acceptance by   Aug.24, 2008

Camera ready papers by    Sep.03, 2008

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

BIOSIGNALS will be held in Porto (Portugal) on January 14 - 17 2009.  Technically co-sponsored by the IEEE Engineering in Medicine and Biology Society (EMBS) and in cooperation with the Association for Computing Machinery (ACM SIGART) and the Association for the Advancement of Artificial Intelligence (AAAI), BIOSIGNALS brings together top researchers and practitioners in several areas of Biomedical Engineering, from multiple areas of knowledge, including biology, medicine, engineering and other physical sciences, interested in studying and using models and techniques inspired from or applied to biological systems. A diversity of signal types can be found in this area, including image, audio and other biological sources of information. The analysis and use of these signals is a multidisciplinary area including signal processing, pattern recognition and computational intelligence techniques, amongst others. The proceedings will be indexed by several major international indexers, including INSPEC and DBLP. Additionaly, a selection of the best papers of the conference will be published in a book, by Springer-Verlag. Best paper awards will be distributed during the conference. Further details can be found at the BIOSIGNALS conference web site (http://www.biosignals.org) This conference is co-located and part of the Joint Conference on Biomedical Engineering Systems and Technologies (BIOSTEC www.biostec.org). Workshops and special sessions are also invited. If you wish to propose a workshop or a special session, for example based on the results of a specific research project, please contact the secretariat.

Marina Carvalho BIOSIGNALS Secretariat Av. D.Manuel I, 27A 2ºesq. 2910-595 Setúbal, Portugal Tel.: +351 265 520 185 Fax: +44 203 014 5436 Email: secretariat@biosignals.org    Web site: http://www.biosignals.org

IMPORTANT DATES: Regular Paper Submission (EXTENDED): July 21, 2008 Authors Notification: October 9, 2008 Final Paper Submission and Registration: October 23, 2008 in cooperation with: ACM SIGART and AAAI technically co-sponsored: IEEE EMB proceedings indexed by INSPEC and DBLP best papers published by Springer-Verlag 

CONFERENCE TOPIS: - Medical Signal Acquisition, Analysis and Processing - Wearable Sensors and Systems - Real-time Systems - Biometrics - Pattern Recognition - Computational Intelligence - Evolutionary Systems - Neural Networks - Speech Recognition - Acoustic Signal Processing - Time and Frequency Response - Wavelet Transform - Medical Image Detection, Acquisition, Analysis and Processing - Physiological Processes and Bio-signal Modeling, Non-linear dynamics - Bioinformatics - Cybernetics and User Interface Technologies - Electromagnetic fields in biology and medicin

KEYNOTE SPEAKERS: - Edward H. Shortliffe, Arizona State University, United States - Vimla L. Patel, Arizona State University, United States - Pier Luigi Emiliani, Institute of Applied Physics "Nello Carrara" (IFAC) of the Italian National Research Council (CNR), Italy - Maciej Ogorzalek, Jagiellonian University, Poland

WORKSHOP: (Regular Paper Submission: October 17, 2008) - Medical Image Analysis and Description for Diagnosis Systems - MIAD 2009 http://www.biostec.org/MIAD.htm

Early discounted registration for the Voice Search Conference

International Workshop on Pharyngeals & Pharyngealisation Co-organised by the Centre for Research in Linguistics and Language Science (CRiLLS), Newcastle University and Praxiling Centre National de la Recherche Scientifique (CNRS), Université Montpellier III  Date: 26 March - 27 March 2009 Venue: Research Beehive, Newcastle University This is the first call for posters and for participants for this Pharyngeals and Pharyngealisation Workshop.  Background on Pharyngeals and Pharyngealisation Over the last 50 years, the study of pharyngeal and pharyngealised sounds has fascinated many a linguist due to their varied phonological representations, their complex articulation and co-articulation patterns, their late development in the process of language acquisition, and their sociolinguistic and crosslinguistic patterning. Approaches to the study of pharyngeal and pharyngealised sounds have included: a) phonological work looking at identifying consistent and uniform characteristics for this class of sounds b) experimental work looking at auditory, acoustic, and articulatory manifestations of pharyngeal constrictions c) acquisition studies looking at pharyngeals in babies’ early vocalisations regardless of language and later development in languages which contain these sounds d) socioliguistic work identifying cross-linguistic and cross-dialectal differences in the realisation of these sounds as well as the influence of social factors such gender, ethnicity, and social class. For more information please click here.  Aims of the Workshop The purpose of this workshop is to bring together researchers from around the world who have worked on pharyngeal and pharyngealised sounds in order to share expertise in different methodologies and theoretical approaches to the study of these sounds and attempt to answer various outstanding questions regarding:  Language universals: why are pharyngeal sounds present in only 1% of languages surveyed in the UPSID data base when they are present in children’s early vocalisations? How have these sounds evolved in languages that have lost the pharyngeal distinctions? Is there a relationship between a dense consonantal system and the existence of pharyngeal/pharyngealised sounds in a language? Production and perception: what are the acoustic, articulatory, and perceptual correlates of pharyngeal and pharyngealised sounds? What role do visual cues (e.g. lip rounding) play in processing pharyngeal articulations? The sociolinguistic indices of pharyngeal/pharyngealised articulations: How does pharyngealisation manifest itself in different languages/dialects? Is the gender-correlated patterning that has been documented in urban areas in the Arab world with respect to de-emphasis found in other varieties/languages with pharyngeal/pharyngealised articulations? How are pharyngeal articulations affected in language contact situations? Acquisition: at what age are pharyngeal and pharyngealised sounds acquired and what are the developmental manifestations across languages and/or dialects? Invited Speakers Dr John Esling, University of Victoria (Canada) Dr Zeki Hassan, University of Gottenbörg (Sweden) Dr Barry Heselwood, University of Leeds (UK) Prof Asher Laufer, The Hebrew University (Israel Prof Ian Maddieson, University of New Mexico (USA) Dr Slim Ouni, CNRS-Université de Nancy I (France) Dr Yves Laprie, CNRS-Université de Nancy I (France) Dr Rachid Ridouane, LPP CNRS-Université Paris III (France) Dr Kimary Shahin, Qatar University (Qatar) Prof Jim Scobbie, Queen Margaret University (UK) Prof Janet Watson, University of Salford (UK) Dr Chakir Zeroual, Faculté Polydisciplinaire de Taza (Morocco) Scientific Committee Jalal Al-Tamimi (Newcastle University) Enam Al Wer (Essex University) Thomas Baer (University of Cambridge) Jean-Francois Bonnot (Université de Besançon) Nick Clements (LPP CNRS-Université Paris III) Stuart Davis (Indiana University) Susanne Fuchs (Berlin University) John McCarthy (UMASS) Mohamed Embarki (Université Montpellier III) Kenneth de Jong (Indiana University) Ghada Khattab (Newcastle University) Amanda Miller (University of British Columbia) Daniel Recasens (University Autònoma of Barcelona) Harvey Sussman (University of Texas) Nathalie Vallée (Université Stendhal, Grenoble) Organisation of the Workshop The programme consists of 11 invited oral presentations and a poster session.  Workshop Themes We welcome work on pharyngeals and pharyngealisation in any of the following fields:  Universals Phonology Production and perception Modelling Acquisition Sociolinguistic variation To Apply to Present a Poster Abstracts on any of the workshop themes are invited for the poster session. Abstracts should be no longer than two pages including illustrations and references. Please submit your abstract to Crills@ncl.ac.uk by November 30, 2008. Abstracts will be reviewed by the Scientific Committee and applicants will be notified of their acceptance by December the 15th, 2008.  To Apply to Participate Places are available for participants who are not presenting papers but who will participate in discussions. Registration for this event will open in mid-December.  Important dates: Abstract Submission for the poster session: November 30, 2008 Notification of Acceptance: December 15, 2008 Workshop Dates: March 26-27, 2009 Registration: Early deadline (£80 for staff and £40 for students):  January 30, 2009 Late registration (£100 for staff and £50 for students):  January 30- March 25 Registration on the day: £120 for staff and £60 for students Registration for this event will open mid-December so please keep checking the website. Scholarships: Four scholarships of 125 Euros (approximately £100) each will be offered to students travelling from outside the UK and presenting at the poster session. In order to apply for one of the scholarships, please fill in the scholarship form and submit it together with your abstract by the 30th of November deadline. Travel For more information on travelling to and around Newcastle please visit link to www.ncl.ac.uk/travel/info where you can also download maps of the city centre and campus.  Accommodation Coming soon  Organization Committee Jalal Al-Tamimi, CRiLLS, Newcastle University (UK). Jalal.Al-Tamimi@ncl.ac.uk  Mohamed Embarki, Praxiling UMR 5267 CNRS-Montpellier III (France). mohamed.embarki@univ-montp3.fr  Ghada Khattab, CRiLLS, Newcastle University (UK). Ghada.Khattab@ncl.ac.uk  Hussain Kriba, CRiLLS, Newcastle University (UK). Hussin.Kriba@ncl.ac.uk

Call for Papers  3rd INTERNATIONAL CONFERENCE ON LANGUAGE AND AUTOMATA THEORY AND APPLICATIONS (LATA 2009)  Tarragona, Spain, April 2-8, 2009  http://grammars.grlmc.com/LATA2009/  *********************************************************************  AIMS:  LATA is a yearly conference in theoretical computer science and its applications. As linked to the International PhD School in Formal Languages and Applications that was developed at the host institute in the period 2002-2006, LATA 2009 will reserve significant room for young scholars at the beginning of their career. It will aim at attracting contributions from both classical theory fields and application areas (bioinformatics, systems biology, language technology, artificial intelligence, etc.).  SCOPE:  Topics of either theoretical or applied interest include, but are not limited to:  - algebraic language theory - algorithms on automata and words - automata and logic - automata for system analysis and programme verification - automata, concurrency and Petri nets - biomolecular nanotechnology - cellular automata - circuits and networks - combinatorics on words - computability - computational, descriptional, communication and parameterized complexity - data and image compression - decidability questions on words and languages - digital libraries - DNA and other models of bio-inspired computing - document engineering - extended automata - foundations of finite state technology - fuzzy and rough languages - grammars (Chomsky hierarchy, contextual, multidimensional, unification, categorial, etc.) - grammars and automata architectures - grammatical inference and algorithmic learning - graphs and graph transformation - language varieties and semigroups - language-based cryptography - language-theoretic foundations of natural language processing, artificial intelligence and artificial life - mathematical evolutionary genomics - parsing - patterns and codes - power series - quantum, chemical and optical computing - regulated rewriting - string and combinatorial issues in computational biology and bioinformatics - symbolic dynamics - symbolic neural networks - term rewriting - text algorithms - text retrieval, pattern matching and pattern recognition - transducers - trees, tree languages and tree machines - weighted machines  STRUCTURE:  LATA 2009 will consist of:  - 3 invited talks (to be announced in the second call for papers) - 2 invited tutorials (to be announced in the second call for papers) - refereed contributions - open sessions for discussion in specific subfields or on professional issues (if requested by the participants)  PROGRAMME COMMITTEE:  Parosh Abdulla (Uppsala) Stefania Bandini (Milano) Stephen Bloom (Hoboken) John Brzozowski (Waterloo) Maxime Crochemore (London) Juergen Dassow (Magdeburg) Michael Domaratzki (Winnipeg) Henning Fernau (Trier) Rusins Freivalds (Riga) Vesa Halava (Turku) Juraj Hromkovic (Zurich) Lucian Ilie (London, Canada) Kazuo Iwama (Kyoto) Aravind Joshi (Philadelphia) Juhani Karhumaki (Turku) Jarkko Kari (Turku) Claude Kirchner (Bordeaux) Maciej Koutny (Newcastle) Kamala Krithivasan (Chennai) Martin Kutrib (Giessen) Andrzej Lingas (Lund) Aldo de Luca (Napoli) Rupak Majumdar (Los Angeles) Carlos Martin-Vide (Tarragona & Brussels, chair) Joachim Niehren (Villeneuve d'Ascq) Antonio Restivo (Palermo) Joerg Rothe (Duesseldorf) Wojciech Rytter (Warsaw) Philippe Schnoebelen (Cachan) Thomas Schwentick (Dortmund) Helmut Seidl (Muenchen) Alan Selman (Buffalo) Jeffrey Shallit (Waterloo) Frank Stephan (Singapore)  ORGANIZING COMMITTEE:  Madalina Barbaiani Gemma Bel-Enguix Cristina Bibire Adrian-Horia Dediu Szilard-Zsolt Fazekas Alexander Krassovitskiy Guangwu Liu Carlos Martin-Vide (chair) Robert Mercas Catalin-Ionut Tirnauca Bianca Truthe Sherzod Turaev Florentina-Lilica Voicu  SUBMISSIONS:  Authors are invited to submit papers presenting original and unpublished research. Papers should not exceed 12 single-spaced pages and should be formatted according to the standard format for Springer Verlag's LNCS series (see http://www.springer.com/computer/lncs/lncs+authors?SGWID=0-40209-0-0-0). Submissions have to be uploaded at:  http://www.easychair.org/conferences/?conf=lata2009  PUBLICATION:  A volume of proceedings published by Springer in the LNCS series will be available by the time of the conference. A refereed volume of extended versions of selected papers will be published after it as a special issue of a major journal. (This was Information and Computation for LATA 2007 and LATA 2008.)  REGISTRATION:  The period for registration will be open since September 1, 2008 to April 2, 2009. The registration form can be found at the website of the conference: http://grammars.grlmc.com/LATA2009/  Early registration fees: 450 euros Early registration fees (PhD students): 225 euros Registration fees: 540 euros Registration fees (PhD students): 270 euros  At least one author per paper should register. Papers that do not have a registered author by December 31, 2008 will be excluded from the proceedings.  Fees comprise free access to all sessions, one copy of the proceedings volume, and coffee breaks. For the participation in the full-day excursion and conference lunch on Sunday April 5, the amount of 70 euros is to be added to the fees above: accompanying persons are welcome at the same rate.  PAYMENT:  Early registration fees must be paid by bank transfer before December 31, 2008 to the conference account at Open Bank (Plaza Manuel Gomez Moreno 2, 28020 Madrid, Spain): IBAN: ES1300730100510403506598 - Swift code: OPENESMMXXX (account holder: LATA 2009 – Carlos Martin-Vide).  (Non-early) registration fees can be paid either by bank transfer to the same account or in cash on site.  Besides paying the registration fees, it is required to fill in the registration form at the website of the conference. A receipt for the payment will be provided on site.  FUNDING:  Up to 20 grants covering partial-board accommodation will be available for nonlocal PhD students. To apply, candidates must e-mail their CV together with a copy of the document proving their present status as a PhD student.  IMPORTANT DATES:  Paper submission: October 22, 2008 Notification of paper acceptance or rejection: December 10, 2008 Application for funding (PhD students): December 15, 2008 Notification of funding acceptance or rejection: December 19, 2008 Final version of the paper for the proceedings: December 24, 2008 Early registration: December 31, 2008 Starting of the conference: April 2, 2009 Submission to the journal special issue: June 22, 2009  FURTHER INFORMATION:  carlos.martin@urv.cat  ADDRESS:  LATA 2009 Research Group on Mathematical Linguistics Rovira i Virgili University Plaza Imperial Tarraco, 1 43005 Tarragona, Spain Phone: +34-977-559543 Fax: +34-977-559597

IEEE International Conference on Acoustics, Speech, and Signal Processing

http://icassp09.com

Sponsored by IEEE Signal Processing Society

April 19 - 24, 2009

Taipei International Convention Center

Taipei, Taiwan, R.O.C.

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

Audio and electroacoustics

Bio imaging and signal processing

Design and implementation of signal processing systems

Image and multidimensional signal processing

Industry technology tracks

Information forensics and security

Machine learning for signal processing

Multimedia signal processing

Sensor array and multichannel systems

Signal processing education

Signal processing for communications

Signal processing theory and methods

Speech and language processing

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

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

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

General Chair

Lin-shan, Lee

National Taiwan University

General Vice-Chair

Iee-Ray Wei

Chunghwa Telecom Co.,Ltd.

Secretaries General

Tsungnan Lin

National Taiwan University

Fu-Hao Hsing

Chunghwa Telecom Co.,Ltd

Technical Program Chairs

Liang-Gee Chen

National Taiwan University

James R. Glass

Massachusetts Institute of Technology

Technical Program Members

Petar Djuric

Stony Brook University

Joern Ostermann

Leibniz University Hannover

Yoshinori Sagisaka

Waseda University

Plenary Sessions

Soo-Chang Pei (Chair)

National Taiwan University

Hermann Ney (Co-chair)

RWTH Aachen

Special Sessions

Shih-Fu Chang (Chair)

Columbia University

Lee Swindlehurst (Co-chair)

University of California, Irvine

Tutorial Chair

Tsuhan Chen

Carnegie Mellon University

Publications Chair

Homer Chen

National Taiwan University

Publicity Chair

Chin-Teng Lin

National Chiao Tung University

Finance Chair

Hsuan-Jung Su

National Taiwan University

Local Arrangements Chairs

Tzu-Han Huang

Chunghwa Telecom Co.,Ltd.

Chong-Yung Chi

National Tsing Hwa University

Jen-Tzung Chien

National Cheng Kung University

Conference Management

Conference Management Services

3^rd Advanced Voice Function Assessment International Workshop (AVFA2009)

Madrid (Spain), 18^th - 20^th May 2009

http://www.avfa09.upm.es

     This is the first Call for Papers and Posters for the 3rd Advanced Voice Function Assessment International Workshop (AVFA2009) that will be held from May 18^th to 20^th at the Universidad Politécncia de Madrid, Spain.

Motivation

    Speech is the most important means of communication among humans, resulting from a complex interaction among vocal folds vibration at the larynx and voluntary movements of the articulators (i.e., mouth, tongue, velum, jaw, etc.). The function of voice, however, is not limited to speech communication. It also transfers emotions, expresses personality features and reflects situations of stress or pathology. Moreover, it has an aesthetic value in many different professional activities, affecting salesmen, managers, lawyers, singers, actors, etc.

     Although research in speech science has traditionally favoured areas such as synthesis, recognition or speaker verification, the previous facts motivate the current emerging of a new research area related to voice function assessment.

     AVFA2009 aims at fostering interdisciplinary collaboration and interactions among researchers in voice assessment beyond the framework of COST Action 2103, thus reaching the whole scientific community.

Topics

     Topics of interest include, but are not limited to: 

• Automatic detection of voice disorders
• Automatic assessment & rating of voice quality
• New strategies for parameterization and modelling normal and pathological voices (biomechanical-based parameters, chaos modelling, etc.)
• Databases of vocal disorders
• Inverse filtering
• Signal processing for remote diagnosis
• Speech enhancement for pathological & oesophageal voices
• Objective parameters extraction from vocal fold images using videolaryngoscopy, videokymography, fMRI and other emerging techniques
• Multi-modal analysis of disordered speech
• Robust pitch extraction algorithms for pathological & oesophageal voices
• Emotions in speech
• Speaker adaptation
• Voice Physiology and Biomechanics
• Modelling of Voice Production
• Diagnosis and Evaluation Protocols
• Substitution Voices
• Evaluation of Clinical Treatments
• Analysis of Oesophageal Voices

Submission

    Prospective authors are asked to electronically submit preliminary version of full papers with a maximum length of 4 pages, including figures and tables, in English. Preliminary papers should be submitted as pdf documents, fitted to the linked  template, by the 15^th of January. The submitted documents should include the title and authors' names, affiliations and addresses. In addition, the e-mail address and phone number of the corresponding author should be given. 

    Workshop proceedings will be edited both in paper and CD-ROM. Author registration to the conference is required for accepted papers to be included in the proceedings. The best papers presented at the workshop will be eligible for publication in a referred journal.

Best student paper award

Based on the comments given by the reviewers and the presentation at the conference, the organizing committee will give a best student paper award. The awarded author will be nominated at the closing ceremony of AVFA2009.

Schedule

·        Proposal due 15th January 2009

·        Notification of acceptance 15th February 2009

·        Final papers due 28th February 2009

·        Preliminary program 1st May 2009

·        Workshop 18th May – 20th May 2009

Registration and Information

Registration will be handled via the AVFA2009 web site (http://www.avfa09.upm.es). Please contact the secretariat (avfa09@ics.upm.es) for further information.

Program Committee

• Juan Ignacio Godino Llorente, Universidad Politécnica de Madrid, Co-Chair
• Pedro Gómez Vilda, Universidad Politécnica de Madrid, Co-Chair
• Rubén Fraile, Universidad Politécnica de Madrid, Scientific Secretariat
• Bartolomé Scola Yurrita, Gregorio Marañón Hospital 

·         Phillippe H. Dejonckere, University Medical Center Utrecht

·         Yannis Stylianou, University of Crete

7ème Atelier International sur Indexation Multimédia Par le Contenu.
7th International Workshop on Content-Based Multimedia Indexing

Lieu de la manifestation

Following the six successful previous events (Toulouse 1999, Brescia 2001, Rennes 2003, Riga 2005, Bordeaux 2007, London 2008), 2009 International Workshop on Content-Based Multimedia Indexing (CBMI) will be held on June 3-5, 2009 at the picturesque city of Chania, in Crete Island, Greece. It will be organized by Image, Video and Multimedia Laboratory of National Technical University of Athens. CBMI 2009 aims at bringing together the various communities involved in the different aspects of content-based multimedia indexing, such as image processing and information retrieval with current industrial trends and developments. CBMI 2009 is supported by IEEE, EURASIP, University of Athens. The technical program of CBMI 2009 will include presentation of invited plenary talks, special sessions as well as regular sessions with contributed research papers.

Topics of interest include, but are not limited to:

Multimedia indexing and retrieval (image, audio, video, text)
Matching and similarity search
Construction of high level indices
Multimedia content extraction
Identification and tracking of semantic regions in scenes
Multi-modal and cross-modal indexing
Content-based search
Multimedia data mining
Metadata generation, coding and transformation
Large scale multimedia database management
Summarisation, browsing and organization of multimedia content
Presentation and visualization tools
User interaction and relevance feedback
Personalization and content adaptation
Evaluation and metrics

Paper Submission

Prospective authors are invited to submit full papers at the conference web site: http://www.cbmi2009.org/submission. Style files (Latex and Word) will be provided for the convenience of the authors.

Important Dates

Venue

CBMI 2009 will be hosted at KAM - Mediterranean Centre of Architecture, Chania, at the island of Crete, one of the most exciting Greek destinations. KAM was settled by Chania municipality in 1996 and is situated since 2002 at Great Arsenali, the old port of Chania.

SPECOM 2009 - ANNOUNCEMENT AND CALL FOR PAPERS

13-th International Conference "Speech and Computer"
21-25 June 2009
Saint-Petersburg, Russia
http://www.specom.nw.ru

Organized by St. Petersburg Institute for Informatics and Automation of the Russian Academy of Sciences (SPIIRAS)

Dear Colleagues, we are pleased to invite you to the 13-th International Conference on Speech and Computer SPECOM'2009, which will be held in June 21-25, 2009 in St.Petersburg. The global aim of the conference is to discuss state-of-the-art problems and recent achievements in Signal Processing and Human-Computer Interaction related to speech technologies. Main topics of SPECOM’2009 are:
- Signal processing and feature extraction
- Multimodal analysis and synthesis
- Speech recognition and understanding
- Natural language processing
- Spoken dialogue systems
- Speaker and language identification
- Text-to-speech systems
- Speech perception and speech disorders
- Speech and language resources
- Applications for human-computer interaction

Imporatnt Dates:
- Submission of full papers: December 1, 2008
- Notification of acceptance: February 1, 2009
- Submission of final papers: March 1, 2009
- Early registration: March 1, 2009
- Conference dates: June 21-25, 2009

Scientific Committee:
Andrey Ronzhin, Russia (conference chairman)
Niels Ole Bernsen, Denmark
Jean Caelen, France
Christoph Draxler, Germany
Thierry Dutoit, Belgium
Hiroya Fujisaki, Japan
Sadaoki Furui, Japan
Jean-Paul Haton, France
Ruediger Hoffmann, Germany
Dimitri Kanevsky, USA
George Kokkinakis, Greece
Steven Krauwer, Netherlands
Lin-shan Lee, Taiwan
Boris Lobanov, Belarus
Benoit Macq, Belgium
Jury Marchuk, Russia
Roger Moore, UK
Heinrich Niemann, Germany
Rajmund Piotrowski, Russia
Louis Pols, Netherlands
Rodmonga Potapova, Russia
Josef Psutka, Czechia
Lawrence Rabiner, USA
Gerhard Rigoll, Germany
John Rubin, UK
Murat Saraclar, Turkey
Jesus Savage, Mexico
Pavel Skrelin, Russia
Viktor Sorokin, Russia
Yannis Stylianou, Greece
Jean E. Viallet, France
Taras Vintsiuk, Ukraine
Christian Wellekens, France

The invited speakers of SPECOM'2009 are:
- Prof. Walter Kellermann (University of Erlangen-Nuremberg, Germany), lecture "Towards Natural Acoustic Interfaces for Automatic Speech Recognition"
- Prof. Mikko Kurimo (Helsinki University of Technology, Finland), lecture "Unsupervised decomposition of words for speech recognition and retrieval"


Independently of the scientific actions we will provide essential possibilities for acquaintance with cultural and historical valuables of Saint-Petersburg, the conference will be hosted during a unique and wonderful period known as the White Nights.

Contact Information:
SPECOM'2009 Organizing Committee,
SPIIRAS, 39, 14-th line, St.Petersburg, 199178, RUSSIA
E-mail: specom@iias.spb.su
Web: http://www.specom.nw.ru

at Johns Hopkins University invites one page research proposals for a

NSF-sponsored, Six-week Summer Research Workshop on

Machine Learning for Language Engineering

to be held in Baltimore, MD, USA,

June 22 to July 31, 2009.

CALL FOR PROPOSALS

Deadline: Wednesday, October 15, 2008.

One-page proposals are invited for the 15^th annual NSF sponsored JHU summer workshop.  Proposals should be suitable for a six-week team exploration, and should aim to advance the state of the art in any of the various fields of Human Language Technology (HLT) including speech recognition, machine translation, information retrieval, text summarization and question answering.  This year, proposals in related areas of Machine Intelligence, such as Computer Vision (CV), that share techniques with HLT are also being solicited.  Research topics selected for investigation by teams in previous workshops may serve as good examples for your proposal. (See http://www.clsp.jhu.edu/workshops.)

Proposals on all topics of scientific interest to HLT and technically related areas are encouraged.  Proposals that address one of the following long-term challenges are particularly encouraged.

Ø  ROBUST TECHNOLOGY FOR SPEECH:  Technologies like speech transcription, speaker identification, and language identification share a common weakness: accuracy degrades disproportionately with seemingly small changes in input conditions (microphone, genre, speaker, dialect, etc.), where humans are able to adapt quickly and effectively. The aim is to develop technology whose performance would be minimally degraded by input signal variations.

Ø  KNOWLEDGE DISCOVERY FROM LARGE UNSTRUCTURED TEXT COLLECTIONS: Scaling natural language processing (NLP) technologies—including parsing, information extraction, question answering, and machine translation—to very large collections of unstructured or informal text, and domain adaptation in NLP is of interest.

Ø  VISUAL SCENE INTERPRETATION: New strategies are needed to parse visual scenes or generic (novel) objects, analyzing an image as a set of spatially related components.  Such strategies may integrate global top-down knowledge of scene structure (e.g., generative models) with the kind of rich bottom-up, learned image features that have recently become popular for object detection.  They will support both learning and efficient search for the best analysis.

Ø  UNSUPERVISED AND SEMI-SUPERVISED LEARNING: Novel techniques that do not require extensive quantities of human annotated data to address any of the challenges above could potentially make large strides in machine performance as well as lead to greater robustness to changes in input conditions.  Semi-supervised and unsupervised learning techniques with applications to HLT and CV are therefore of considerable interest.

An independent panel of experts will screen all received proposals for suitability. Results of this screening will be communicated no later than October 22, 2008. Authors passing this initial screening will be invited to Baltimore to present their ideas to a peer-review panel on November 7-9, 2008.  It is expected that the proposals will be revised at this meeting to address any outstanding concerns or new ideas. Two or three research topics and the teams to tackle them will be selected for the 2009 workshop.

We attempt to bring the best researchers to the workshop to collaboratively pursue the selected topics for six weeks.  Authors of successful proposals typically become the team leaders.  Each topic brings together a diverse team of researchers and students.  The senior participants come from academia, industry and government.  Graduate student participants familiar with the field are selected in accordance with their demonstrated performance, usually by the senior researchers. Undergraduate participants, selected through a national search, will be rising seniors who are new to the field and have shown outstanding academic promise.

If you are interested in participating in the 2009 Summer Workshop we ask that you submit a one-page research proposal for consideration, detailing the problem to be addressed.  If your proposal passes the initial screening, we will invite you to join us for the organizational meeting in Baltimore (as our guest) for further discussions aimed at consensus.  If a topic in your area of interest is chosen as one of the two or three to be pursued next summer, we expect you to be available for participation in the six-week workshop. We are not asking for an ironclad commitment at this juncture, just a good faith understanding that if a project in your area of interest is chosen, you will actively pursue it.

Proposals should be submitted via e-mail to clsp@jhu.edu by 4PM EST on Wed, October 15, 2008.

DIAHOLMIA 2009: THE 13TH WORKSHOP ON THE SEMANTICS AND PRAGMATICS OF DIALOGUE

KTH, Stockholm, Sweden, 24-26 June, 2009

The SemDial series of workshops aims to bring together researchers working on the semantics and pragmatics of dialogue in fields such as artificial intelligence, computational linguistics, formal semantics/pragmatics, philosophy, psychology, and neural science. DiaHolmia will be the 13th workshop in the SemDial series, and will be organized at the Department of Speech Music and Hearing, KTH (Royal Institute of Technology). KTH is Scandinavia's largest institution of higher education in technology and is located in central Stockholm (Holmia in Latin).

WEBSITE: www.diaholmia.org

DATES AND DEADLINES:

Full 8-page papers:
Submission due: 22 March 2009
Notification of acceptance: 25 April 2009
Final version due: 7 May 2009

2-page poster or demo descriptions:
Submission due: 25 April 2009
Notification of acceptance: 7 May 2009

DiaHolmia 2009: 24-26 June 2009 (Wednesday-Friday)

SCOPE:

We invite papers on all topics related to the semantics and pragmatics of dialogues, including, but not limited to:

- common ground/mutual belief
- turn-taking and interaction control
- dialogue and discourse structure
- goals, intentions and commitments
- natural language understanding/semantic interpretation
- reference, anaphora and ellipsis
- collaborative and situated dialogue
- multimodal dialogue
- extra- and paralinguistic phenomena
- categorization of dialogue phenomena in corpora
- designing and evaluating dialogue systems
- incremental, context-dependent processing
- reasoning in dialogue systems
- dialogue management

Full papers will be in the usual 8-page, 2-column format. There will also be poster and demo presentations. The selection of posters and demos will be based on 2-page descriptions. Selected descriptions will be included in the proceedings.

Details on the invited speakers, submission format and procedure and local arrangements will be announced at a later date.

PROGRAMME COMMITTEE:

Jan Alexandersson, Srinivas Bangalore, Ellen Gurman Bard, Anton Benz, Johan Bos, Johan Boye, Harry Bunt, Donna Byron, Jean Carletta, Robin Cooper, Paul Dekker, Giuseppe Di Fabbrizio, Raquel Fernández, Claire Gardent, Simon Garrod, Jonathan Ginzburg, Pat Healey, Peter Heeman, Joris Hulstijn, Michael Johnston, Kristiina Jokinen, Arne Jönsson, Alistair Knott, Ivana Kruijff-Korbayova, Staffan Larsson, Oliver Lemon, Ian Lewin, Diane Litman, Susann Luperfoy, Colin Matheson, Nicolas Maudet, Michael McTear, Wolfgang Minker, Philippe Muller, Fabio Pianesi, Martin Pickering, Manfred Pinkal, Paul Piwek, Massimo Poesio, Alexandros Potamianos, Matthew Purver, Manny Rayner, Hannes Rieser, Laurent Romary, Alex Rudnicky, David Schlangen, Stephanie Seneff, Ronnie Smith, Mark Steedman, Amanda Stent, Matthew Stone, David Traum, Marilyn Walker and Mats Wirén

ORGANIZING COMMITTEE:

Jens Edlund
Joakim Gustafson
Anna Hjalmarsson
Gabriel Skantze