We are back from another memorable Interspeech event in Brisbane,
Australia. Our thanks go to Denis Burnham and his whole team, for all
their 4-year work that led to this event. The venue was great, the
social events wonderful, the coffee/tea breaks were too good, in short,
everything contributed to a great and fruitful interaction between all
participants. From the scientific point of view, we had another top
quality conference, complemented by a very good selection of keynote
speakers and tutorials. We also had a very active student participation
in the round tables.
The ISCA activities during the past year were briefly presented in the
opening ceremony, where we honored Prof. Hiroya Fujisaki with a very
special medal, and we also honored the first 12 ISCA Fellows: Rolf
Carlson, Paul Dalsgaard, Hiroya Fijisaki, Sadaoki Furui, Björn
Granström, Julia Hirschberg, Frederick Jelinek, Roger Moore, Mari
Ostendorf, Louis Pols, Steve Young, and Victor Zue.
The detailed report was presented at the General Assembly - see the
slides in our website at http://www.isca-speech.org/assembly.html
The expansion of ISCA's activities in virtually all areas is the
motivation for expanding the next Board from 11 (+2 ex-officio members)
to 14 members in the upcoming elections in 2009. Even then, to
effectively execute our expanded set of programs, we need to solicit
volunteers to aid us with various functions, including international
affairs, grants, the management of our website, selection of best
student papers, liaison with other professional organizations, liaison
with industry, etc.
We believe that this growing voluntary involvement is particularly
important with regards to enhancing the transparency of ISCA's
organization and functions, facilitating the processes of nomination and
election to the board, as well as smoothing the handover from board
members who are completing their terms of service to new board members
who are starting their terms.
If you are interested in volunteering your time to help ISCA best
provide a rich set of services to its members, please contact us at:
president[at]isca-speech<dot>org
In the closing ceremony, we also honored the winners of the best SPEECH
Communication paper (J.P. Barker, M.P. Cooke, and D.P.W. Ellis), and the
winners of the best student papers at Interspeech 2008 (Georg Heigold,
Mitchell McLaren, and Yen-Liang Shue).
Finally, we also honored the two SAC Board members who, due to their
approaching graduation, will leave the Board during the next year. Tiago
Falk and Ebru Arisoy will leave the Board next year and, as a token of
our gratitude, they were given the first examples of the ISCA complete
archive on DVD (thanks to Wolfgang Hess), featuring more than 100
events. The SAC is looking for volunteers to replace them, so please
contact volunteer[at]isca-students[dot]org if you're interested in
joining them.
If you missed Interspeech 2008 in Brisbane, you also missed the long
sugar candies labeled Interspeech 2009 that were distributed there.
Please make plans to come to the UK in September 2009, where you can
participate in the Loebner prize, and taste the "Brighton rocks" again.
More info in www.interspeeech2009.org
I hope to meet you all very soon at another ISCA event.
Isabel Trancoso
President of ISCA board 
 

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.

         Dear Prof. Dr. Wellekens,

As you may already be aware, fraudulent e-mail solicitations for scholarly papers have been circulating which claim to originate from Elsevier, Inc. and are directed to prospective authors and editors. 

We are concerned about these emails and want to alert our community 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, Inc. 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,

Gail Rodney 
G.Rodney@elsevier.com

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

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

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

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

Nagoya Institute of Technology is seeking a researcher for a

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

EMIME ("Efficient multilingual interaction in mobile environment")

involving Nagoya Institute of Technology and other five European

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

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

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

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

position provides opportunities to collaborate with other researchers

in a variety of national and international projects. The competitive

salary is calculated according to qualifications based on NIT scales.

The candidate should have a strong background in speech signal

processing and some experience with speech synthesis and recognition.

Desired skills include familiarity with latest spectrum of technology

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

For more information, please contact Keiichi Tokuda

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

About us

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

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

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

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

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

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

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

outstanding, continuous contribution of developing high-performance,

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

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

continuous speech recognition engine "Julius"

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

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

involved in numerous national and international collaborative

projects. SPL also has close partnerships with many industrial

companies, in order to transfer its research into commercial

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

Funai, Asahi-Kasei, ATR.

Project summary of EMIME

The EMIME project will help to overcome the language barrier by

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

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

to produce spoken output in another language, while continuing to

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

cross-lingual spoken communication is an important, but little

explored, topic. It is essential for providing more natural

interaction and making the computing device a less obtrusive element

when assisting human-human interactions.

We will build on recent developments in speech synthesis using hidden

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

recognition. Using a common statistical modeling framework for

automatic speech recognition and speech synthesis will enable the use

of common techniques for adaptation and multilinguality.

Significant progress will be made towards a unified approach for

speech recognition and speech synthesis: this is a very powerful

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

project, we will explore the use of speaker adaptation across

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

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

characteristics when producing output speech in another language.

Our objectives are to:

1. Personalize speech processing systems by learning individual

characteristics of a user's speech and reproducing them in

synthesized speech.

2. Introduce a cross-lingual capability such that personal

characteristics can be reproduced in a second language not spoken

by the user.

3. Develop and better understand the mathematical and theoretical

relationship between speech recognition and synthesis.

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

cross-lingual personalization.

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

practical mobile application.

Head of NLP :

 We are now looking for a very experienced Computational Linguist

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

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

 opportunities for applying linguistic techniques are virtually

 limitless and even small improvements in the algorithms for

 detecting and correcting potential conversion errors translate

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

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

 forward for this department.

 Experience Needed:

 Grammars and parsing for spontaneous speech.

 Statistical methods.

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

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

 Experience with Automatic Speech Recognition technology.

 Probabilistic Language Modelling

 Phonetics

 Multi-lingual

info@voxid.co.uk 

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

Junior Research Engineer for Embedded Automatic Speech Recognition

Work Environment

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

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

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

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

Key Responsibilities

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

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

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

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

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

Your Profile

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

Interested?

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

We are looking forward to receiving your application!

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

       SOFTWARE ENGINEER SPEECH DIALOGUE TOOLS

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

OVERVIEW:

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

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

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

RESPONSIBILITIES:

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

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

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

QUALIFICATIONS:

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

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

- You have experience with Python or other scripting languages.

- GUI programming experience is an asset.

The following skills are a plus:

- Understanding of communication protocols

- Understanding of databases

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

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

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

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

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

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

- Knowledge of other languages is a plus.

CONTACT:

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

Benjamin Campued       Benjamin.Campued@nuance.com

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

ABOUT US:

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

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

 Speech Scientist for Embedded Automatic Speech Recognition

Work Environment

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

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

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

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

Key Responsibilities

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

·          Localizing natural language dialogue systems towards a specific market.

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

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

·          Assisting senior team members in research tasks.

Your Profile

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

Interested?

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

We are looking forward to receiving your application!

The experience speaks for itself™

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

Senior Research Engineer for Embedded Automatic Speech Recognition

Work Environment

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

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

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

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

Key Responsibilities

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

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

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

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

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

Your Profile

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

Interested?

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

We are looking forward to receiving your application!

The experience speaks for itself™

Title: jr. Speech Scientist

Location: Aachen, Germany

Type: Permanent

Job:  

Overview:

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

Responsibilities:

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

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

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

Representative duties include:

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

Qualifications:

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

Beneficial Skills

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

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

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

·         Be native or near native Danish speaker

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

·         Have an advanced level of English

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

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

·         Have team work experience

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

·         Willing to start in September 2008

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

We will only consider electronic submissions. 

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

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

·         Be native or near native Catalan speaker

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

·         Have an advanced level of English

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

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

·         Have team work experience

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

·         Willing to start in September 2008

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

We will only consider electronic submissions. 

titre du sujet : Synthèse vocale de haute qualité

Introduction
=========

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

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

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

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

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

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

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

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

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

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

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

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

__________________________________________________________________________________________________________

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

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

·         Be native or near native Norwegian Bokmal speaker

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

·         Have an advanced level of English

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

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

·         Have team work experience

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

·         Willing to start in October 2008

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

We will only consider electronic submissions.

Deadline for submissions: August 10, 2008 

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

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

·         Be native or near native Finnish speaker

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

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

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

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

·         Have team work experience

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

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

·         Willing to start in September 1, 2008

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

We will only consider electronic submissions.

Deadline for submissions: August 10, 2008

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.

Within AUDIS 4 universities and 2 industrial companies around Europe offer a joint research and training programme funded by the EU under the FP 7 “People” Programme as a Marie Curie Action (ITN). Over the duration of 24 to 36 months AUDIS will provide opportunities for doctoral and post-doctoral research fellows in digital signal processing for applications in hearing instruments. Cutting edge research projects in the fields of digital signal processing, audiology and psychoacoustics, a wide variety of international training activities and attractive grants will be made available to successful candidates.

Positions for PhDs and Postdocs are offered at the following partner labs:

Ruhr-Universität Bochum / Germany: Prof. R. Martin (rainer.martin@rub.de)

Katholieke Universiteit Leuven / Belgium: Prof. J. Wouters (jan.wouters@med.kuleuven.be)

University of Southampton / United Kingdom: Prof. M. Lutman (mel@isvr.soton.ac.uk)

Kungliga Tekniska Högskolan / Sweden: Prof. A. Leijon (arne.leijon@ee.kth.se)

Siemens Medical Solutions / Germany: Dr. H. Puder (henning.puder@siemens.com)

Cochlear Benelux NV / Belgium: Dr. B. van Dijk (bvandijk@cochlear.be)

For further information on research projects available, application details and eligibility please visit the AUDIS web-site www.audis-itn.eu or contact the project coordinator Prof. R. Martin on rainer.martin@rub.de or Isabel Strauss at the coordination office on isabel.strauss@uv.rub.de.

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.

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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.

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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

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

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

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

http://mlsp2008.conwiz.dk/

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

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

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

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

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

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

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

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

MLSP 2008 ORGANIZING COMMITTEE:

General Chair
Jose Principe

Program Chair
Deniz Erdogmus

Technical Chair
Tulay Adali

Publicity Chairs
Ignacio Santamaria
Marc Van Hulle

Publication Chair
Jan Larsen

Data Competition
Ken Hild
Vince Calhoun

Local Arrangements
Juan Azuela

The Tenth International Conference on Multimodal Interfaces (ICMI

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

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

broad field of multimodal interaction and systems. The conference will

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

identify a roadmap for future research and commercial success. ICMI

2008 will feature a main conference with keynote speakers, panel

discussions, technical paper presentations and discussion (single

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

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

ICMI-08 provides excellent conditions for brainstorming and sharing

the latest advances about multimodal interaction and systems in an

inspired setting full of history, mythology and art.

Paper Submission

There are two different submission categories: regular paper and short

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

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

based on suitable delivery of the content.

Demo Submission

Proposals for demonstrations shall be submitted to demo chairs

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

Doctoral Spotlight

Doctoral Student Travel Support and Spotlight Session. Funds are

expected from NSF to support participation of doctoral candidates at

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

thesis work. Students

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

specified above.

Topics of interest include

* Multimodal and Multimedia processing

* Multimodal input and output interfaces

* Multimodal applications

* User Modeling and Adaptation

* Multimodal Architectures, Tools and Standards

* Evaluation of Multimodal Interfaces

*Important Dates:*

Paper submission: May 23, 2008

Author notification July 14, 2008

Camera ready deadline: August 15, 2008

Conference: October 20-22, 2008

Organizing Committee

General Co-Chairs

Vassilis Digalakis, TU Crete, Greece

Alex Potamianos, TU Crete, Greece

Matthew Turk, UC Santa Barbara, USA

Program Co-Chairs

Roberto Pieraccini, SpeechCycle, USA

Jian Wang, Microsoft Research, China

Yuri Ivanov, MERL 

Jeudi 23 octobre 2008, 13h30 - Séminaire externe
========================================
David OSTRY
Department of Psychology
McGill University, Montréal, Canada

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

Jeudi 6 novembre 2008, 13h30 - Séminaire externe
========================================
Yi XU
University College London, Dept of Phonetics & Linguistics

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

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

Jeudi 20 novembre 2008, 13h30 - Séminaire externe
========================================
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

Invited Speakers:

·         Nestor Becerra (Universidad de Santiago, Chile)

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

·         Giussepe Ricardi (University of Trento, Italy)

Next Generation Spoken Language Interfaces

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

Embodied conversational agents  in verbal and non-verbal communication

·         Yannis Stilianou (University of Crete, Grecia)

Voice Conversion: State of the art and Perspectives

Important dates:

·         Full paper submission: July 20, 2008

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

Contact information:

VJTH’2008

Dept. Electronics and Telecommunications

Faculty of Engineering

Alda. Urkijo s/n

48013 Bilbao

Tel.: +34 946 013 969

Fax.: +34 946 014 259

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

• Language recognition

Registration Form

• Speech synthesis

Registration Form

• Automatic text translation

Registration Form

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

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

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

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

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.