ISCA Scientific Achievement Medal for 2008 It is with great pleasure that I announce the ISCA Medalist for 2008 - Hiroya Fujisaki. Prof. Fujisaki has contributed to the speech research community in so many aspects, in speech analysis, synthesis and prosody, that it will be a very hard task for me to summarize his long list of achievements. He is also the founder of the ICSLP series of conferences which, being now fully integrated as one of ISCA's yearly conferences, will have its 10th anniversary this year.

ISCA announces with great pleasure that the venue for
Interspeech 2011 will be FLORENCE.

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

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

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

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

Eva Hajicova - Membership Services 

 Emmanuelle Foxonet - ISCA Secretariat  for the ISCA board 

INTERSPEECH 2008 incorporating SST 08

September 22-26, 2008

Brisbane Convention & Exhibition Centre

Brisbane, Australia

http://www.interspeech2008.org/

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

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

including (but not limited to):

             Human Speech Production, Perception and Communication; 

             Speech and Language Technology; 

             Spoken Language Systems; and 

            Applications, Resources, Standardisation and Evaluation

• In addition, a number of Special Sessions on selected topics have been organised and we invite you to submit for these also (see website for a complete list).

  Interspeech 2008 has two types of submission formats: Full 4-page Papers and

   Short 1-page Papers. Prospective authors are invited to submit papers in either

   format via the conference website by 7 April 2008. 

   

  Important Dates 

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

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

  Early Registration Deadline: Monday, 7 July 2008, 3pm GMT 

  Tutorial Day: Monday, 22 September 2008 

  Main conference: 23-26 September 2008 

   For more information please visit the website http://www.interspeech2008.org

   

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

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

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

NEW Deadline for sending one page abstract = JUNE 9TH


Ten years after our colleague Christian Benoît departed, the mark that
he left is still very vivid in the international community. There will
soon be several occasions to honour his memory: during the next
Interspeech conference (Christian was secretary of the ESCA, future
ISCA, for a long time, the association is a French association of the
type described in the 1901 law and its official headquarters are still
in Grenoble), as well as during the next AVSP workshop (workshop of
which he was one of the creators). The Christian Benoît Association was
created in 1999 and regularly awards young researchers the "Christian
Benoît prize" to promote their research (the 4^th prize was awarded to
the phonetician Susanne Fuchs in 2007). The Christian Benoît association
http://www.icp.inpg.fr/ICP/_communication.fr.html#prixcb), along with
ICP, now Speech and Cognition Department of Gipsa-lab
(http://www.gipsa-lab.inpg.fr <http://www.gipsa-lab.inpg.fr/>), are
organizing a workshop/summer school to Christian Benoît’s memory, in the
line of his innovative and enthusiastic research style and aiming at
exploring the topic of "Speech and Face to Face Communication" in a
pluridisciplinary perspective: neuroscience, cognitive psychology,
phonetics, linguistics and computer modelling. The workshop "Speech and
Face to Face Communication" will be organized around 11 invited
conferences. All researchers from the field are invited to participate
through a call for papers and students will be encouraged to widely
attend the workshop and present their work.

Website: http://www.icp.inpg.fr/~dohen/face2face/

Deadline for sending one page abstracts: June 9th (see Call for Papers
<http://ww.icp.inpg.fr/%7Edohen/face2face/CallForPapers.html>)

You can subscribe to the Christian Benoît Association by sending 15
euros (active member; 45 euros or more, benefactors) to Pascal Perrier,
secretary of the association: Pascal.Perrier@gipsa-lab.inpg.fr
<mailto:Pascal.Perrier@gipsa-lab.inpg.fr>.

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

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

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

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

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

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

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

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

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

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

La production de la parole
Author: Alain Marchal, Universite d'Aix en Provence, France
Publisher: Hermes Lavoisier
Year: 2007

Speech enhancement-Theory and Practice
Author: Philipos C. Loizou, University of Texas, Dallas, USA
Publisher: CRC Press
Year:2007

Speech and Language Engineering
Editor: Martin Rajman
Publisher: EPFL Press, distributed by CRC Press
Year: 2007

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

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

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

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

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

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

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

Membership Mailbag - 'Penn' Treebanks and Recent Directions in English Treebanking

The LDC Membership Office responds to over 4000 emailed queries a year, and, over time, we've noticed that some questions tend to crop up with regularity.  To address the questions that you, our data users, have asked, we'd like to introduce our new Membership Mailbag series of newsletter articles.  This month we will look into the differences between the 'Penn' Treebanks and review recent directions in English treebanking.

Treebank-2 and Treebank-3 both contain 1 million words of Wall Street Journal (WSJ) text  and a small sample of ATIS-3 data that have been annotated using a Treebank II annotation-style, plus a part-of-speech tagged version of the Brown corpus.  Treebank-3 is considered a super-set of Treebank-2.  That is, if you are undecided between Treebank-2 and -3, in most instances, the best choice would be Treebank-3. Treebank-3 corrects known technical errors in Treebank-2 plus it contains Switchboard data which has been tagged, dysfluency-annotated, and a small portion of the Brown corpus which has been parsed in the Treebank II annotation-style.   

Note, however, that there are a few items missing from Treebank-3 that are found in Treebank-2.  Treebank 3 does not contain the complete parsed Brown corpus.  Treebank-2 contains the complete parsed Brown corpus done in the older Treebank I annotation-style; this is not contained in Treebank-3. Also, Treebank-3 does not include the tgrep software for extracting data, but tgrep and a newer version, tgrep2, are freely available online.  Finally, Treebank-3 does not contain the raw Wall Street Journal (WSJ) text, but organizations can obtain this by request.

Much recent treebanking has focused on languages other than English, but English treebanking efforts did not come to an end with the release of Treebank-3.  Ongoing work uses an updated Treebank II annotation-style and consists of two types of annotation; straight treebanking and treebanking in combination with another kind of annotation.  Straight treebank annotation can be found in corpora such as English Chinese Translation Treebank v 1.0 and English-Arabic Treebank v 1.0.  In these corpora, the Chinese or Arabic source texts have been translated into English, then POS-tagged and treebanked, thus making them suitable for machine translation work as well.  Additional translation treebanks are planned for release and will feature cleaner translation and contain substantially more data. 

Corpora which combine treebanking with another type of annotation include the English Conversational Telephone Speech Treebank with Structural Metadata, to be released later this year.  This treebank is annotated for structural metadata including fillers, disfluencies and sentence/semantic units, and also tagged for syntactic structure, and so, evaluates the impact of metadata extraction (MDE) on parsing information.  While these newer releases are smaller than the Penn Treebanks, the improved Treebank II annotation-style has a very high rate of inter-annotator agreement..  Additionally, the source texts are more varied in both domain and style than the WSJ texts that constitute the bulk of Penn Treebank.

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

New Publications

(1) Chinese Proposition Bank 2.0 (CPB2.0) is a continuation of the Chinese Proposition Bank project, which aims to create a corpus of Chinese text annotated with information about basic semantic propositions. Chinese Proposition Bank 1.0 consists of predicate-argument annotation on 250,000 words from Chinese Treebank 5.0. Chinese Proposition Bank 2.0 adds predicate-argument annotation on 500,000 words from Chinese Treebank 6.0. The data sources include newswire from Xinhua News Agency, articles from Sinorama Magazine, news from the website of the Hong Kong Special Administrative Region and transcripts from various Chinese broadcast news programs.

This release contains the predicate-argument annotation of 81,009 verb instances (11,171 unique verbs) and 14,525 noun instances (1,421 unique nouns). The annotation of nouns is limited to nominalizations that have a corresponding verb. The general annotation guidelines and the lexical guidelines (called frame files) for each verbal and nominal predicate are included in this release.  Chinese Proposition Bank 2.0 is distributed via web download.

2008 Subscription Members will automatically receive two copies of this corpus on disc. 2008 Standard Members may request a copy as part of their 16 free membership corpora. Nonmembers may license this data for US$850.

*

(2)  Hindi WordNet was developed by researchers at the Center for Indian Language Technology, Computer Science and Engineering Department, IIT Bombay.  Wordnets are systems for analyzing the different lexical and semantic relations between words. Specifically, a wordnet is a word sense network in which words are grouped into semantically equivalent units called synsets. Each synset represents a lexical concept, and synsets are linked to each other by semantic relations (between synsets) and lexical relations (between words). Similar in design to the Princeton Wordnet for English, Hindi Wordnet incorporates additional features to capture the complexities of Hindi. This release of Hindi Wordnet consists of 56,928 unique words and 26,208 synsets.

Additional information about the development of Hindi Wordnet is available at the Hindi WordNet web site.

Hindi WordNet contains nouns, verbs, adjectives and adverbs. Each entry consists of the following elements:

1.      Synset: a set of synonymous words. The words in the synset are arranged according to the frequency of usage.

2.      Gloss: the concept. It consists of two parts:

Text definition: explains the concept denoted by the synset. 

Example sentence: gives the usage of the words in the sentence.

3.      Position in Ontology: An ontology is a hierarchical organization of concepts, or more specifically, a categorization of entities and actions. A separate ontological hierarchy exists for each syntactic category (noun, verb, adjective adverb). Each synset is mapped into some place in the ontology..

This release of Hindi WordNet is made available as a complete Java application along with an API to facilitate further development.  Hindi WordNet is distributed via web download. 

2008 Subscription Members will automatically receive two copies of this corpus on disc, provided that they have submitted a signed copy of the User License Agreement for Hindi WordNet (LDC2008L02).  2008 Standard Members may request a copy as part of their 16 free membership corpora. Nonmembers may license this data for US$300.

*

(3) West Point Brazilian Portuguese Speech is a database of digital recordings of spoken Brazilian Portuguese designed and collected by staff and faculty of the Department of Foreign Languages (DFL) and Center for Technology Enhanced Language Learning (CTELL) to develop acoustic models for speech recognition systems. The U.S. government uses such systems to provide speech-recognition enhanced language learning course ware to government linguists and students enrolled in various government language programs.

The data in this corpus was collected in March 1999 in Brasilia, Brazil using informants from a Brazilian military academy. The corpus consists of read speech from 60 female and 68 male native and non-native speakers.  The speech was elicited from a prompt script containing 296 sentences and phrases typically used in language learning situations.

The speech was collected using four laptop computers running MS Windows. Three of the computers recorded with a 16 bit data size and sampling rate of 22050 Hz, the other laptop recorded with an 8 bit data size at a sampling rate of 11025 Hz. The recording script presented a visual display of the sentence to be recorded. The informant pressed a key and spoke the sentence. The recording was played back for review, allowing the utterance to be re-recorded. West Point Brazilian Portuguese Speech is distributed on one DVD-ROM.

2008 Subscription Members will automatically receive two copies of this corpus. 2008 Standard Members may request a copy as part of their 16 free membership corpora. Nonmembers may license this data for US$500. l

The QAst organizers are pleased to announce the release of the development dataset for
the CLEF-QA 2008 track "Question Answering on Speech Transcripts" (QAst).
We take this opportunity to launch a first call for participation in
this evaluation exercise.

QAst is a CLEF-QA track that aims at providing an evaluation framework
for QA technology on speech transcripts, both manual and automatic.
A detailed description of this track is available at:
http://www.lsi.upc.edu/~qast <http://www.lsi.upc.edu/~qast>

It is the second evaluation for the QAst track.
Last year (QAst 2007), factual questions had been generated for two
distinct corpora (in English language only). This year, in addition to
factual questions,
some definition questions are generated, and five corpora covering three
different languages are used (3 corpora in English, 1 in Spanish and 1
in French).

Important dates:

# 15 June 2008: evaluation set released
# 30 June 2008: submission deadline

The pilot track is organized jointly by the Technical University of
Catalonia (UPC), the Evaluations and Language resources Distribution
Agency (ELDA) and Laboratoire d'Informatique pour la Mécanique et les
Sciences de l'Ingénieur (LIMSI).

If you are interested in participating please send an email to Jordi
Turmo (turmo_AT_lsi.upc.edu) with "QAst" in the subject line.

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

ATT - 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).

Motorola Labs - Center for Human Interaction Research (CHIR) located in Schaumburg Illinois, USA, is offering summer intern positions in 2008 (12 weeks each).

CHIR's mission:

Our research lab develops technologies that provide access to rich communication, media and information services effortless, based on natural, intelligent interaction. Our research aims on systems that adapt automatically and proactively to changing environments, device capabilities and to continually evolving knowledge about the user.

Intern profiles:

1) Acoustic environment/event detection and classification.

Successful candidate will be a PhD student near the end of his/her PhD study and is skilled in signal processing and/or pattern recognition; he/she knows Linux and C/C++ programming. Candidates with knowledge of acoustic environment/event classification are preferred.

2) Speaker adaptation for applications on speech recognition and spoken document retrieval.

The successful candidate must currently be pursuing a Ph.D. degree in EE or CS with complete understanding and hand-on experience on automatic speech recognition related research. Proficiency in Linux/Unix working environment and C/C++ programming. Strong GPA. A strong background in speaker adaptation is highly preferred.

3) Development of voice search-based web applications on a smartphone

We are looking for an intern candidate to help create an "experience" prototype based on our voice search technology. The app will be deployed on a smartphone and demonstrate intuitive and rich interaction with web resources. This intern project is oriented more towards software engineering than research. We target an intern with a master's degree and strong software engineering background. Mastery of C++ and experience with web programming (AJAX and web services) is required. Development experience on Windows CE/Mobile desired.

4) Integrated Voice Search Technology For Mobile Devices.

Candidate should be proficient in information retrieval, pattern recognition and speech recognition. Candidate should program in C++ and script languages such as Python or Perl in Linux environment. Also, he/she should have knowledge on information retrieval or search engines.

We offer competitive compensation, fun-to-work environment and Chicago-style pizza.

If you are interested, please send your resume to:

Dusan Macho, CHIR-Motorola Labs

Email: dusan.macho@motorola.com

Tel: +1-847-576-6762

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 R&D 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 either at Nuance's Office in Aachen, a beautiful, old city right in the heart of Europe with great history and culture, or at Nuance's International Headquarters in Merelbeke, a small town just 5km away from the heart of the vibrant and picturesque city of Ghent, in the Flanders region of Belgium. Both Aachen and Ghent offer some of the most spectacular historic town centers in Europe, and are home to large international universities.

- You will work in an international company and 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 a strong asset.

The following skills are a plus:

- Understanding of communication protocols

- Understanding of databases

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

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

- 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

Deanna Roe                 Deanna.roe@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 3000 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, 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 Professional Services team, based in London, we are currently looking for a

                            Speech Scientist, London, UK

Nuance Professional Services (PS) has designed, developed, and optimized thousands of speech systems across dozens of industries, including directory search, call center automation, applications in telecom, finance, airline, healthcare, and other verticals; applications for video games, mobile dictation, enhanced search services, SMS, and in-car navigation.  Nuance PS applications have automated approximately 7 billion phone conversations for some of the world's most respected companies, including British Airways, Vodafone, Amtrak, Bank of America, BellCanada, Citigroup, General Electric, NTT and Verizon.

The PS organization consists of energetic, motivated, and friendly individuals.  The Speech Scientists in PS are among the best and brightest, with PhDs from universities such as Cambridge (UK), MIT, McGill, Harvard, Penn, CMU, and Georgia Tech, and having worked at research labs such Bell Labs, Motorola Labs, and ATR (Japan), culminating in over 300 years of Speech Science experience and covering well over 20 languages.

Come and join Nuance PS and work on the latest technology from one of the prominent speech recognition technology providers, and make a difference in the way the world communicates.

Job Overview

As a Speech Scientist in the Professional Services group, you will work on automated speech recognition applications, covering a broad range of activities in all project phases, including the design, development, and optimization of the system.  You will:

• Work across application development teams to ensure best possible recognition performance in deployed systems
• Identify recognition challenges and assess accuracy feasibility during the design phase,
• Design, develop, and test VoiceXML grammars and create JSPs, Java, and ECMAscript grammars for dynamic contexts
• Optimize accuracy of applications by analyzing performance and tuning statistical language models, pronunciations, and acoustic models, including identifying areas for improvement by running the recognizer offline
• Contribute to the generation and presentation of client-facing reports
• Act as technical lead on more intensive client projects
• Develop methodologies, scripts, procedures that improve efficiency and quality
• Develop tools and enhance algorithms that facilitate deployment and tuning of recognition components
• Act as subject matter domain expert for specific knowledge domains
• Provide input into the design of future product releases

     Required Skills

• MS or PhD in Computer Science, Engineering, Computational Linguistics, Physics, Mathematics, or related field (or equivalent)
• 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
• Works well in a team and in a fast-paced environment

Beneficial Skills

• Strong programming skills in either Perl, Python, Java, C/C++, or Matlab
• Speech recognition knowledge
• Strong pattern recognition, linguistics, signal processing, or acoustics knowledge
• Statistical data analysis
• Experience with XML, VoiceXML, and Wiki
• Ability to mentor or supervise others
• Additional language skills, eg French, Dutch, German, Spanish

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.

Digital publishing AG is one of Europe's leading producers of  interactive software for foreign language training. In our e- learning courses we want to place the emphasis on speaking and  spoken language understanding.  In order to strengthen our Research & Development Team in Munich,  Germany, we are looking for experienced C or C++ programmers with  at least 3 years experience in the design and coding of  sophisticated software systems under Windows.   

We offer   

-a creative working atmosphere in an international team of   software engineers, linguists and editors working on    challenging research projects in speech recognition and    speech dialogue systems  

- participation in all phases of a product life cycle, as we    are interested in the fast transfer of research results    into products.  

- the possibility to participate in international scientific    conferences.   

- a permanent job in the center of Munich.  

- excellent possibilities for development within our fast    growing company.    

- flexible working times, competitive compensation and    arguably the best espresso in Munich.   

We expect  

-several years of practical experience in software    development in C or C++ in a commercial or academic    environment.  

-experience with parallel algorithms and thread    programming.  

-experience with object-oriented design of software    systems.  

-good knowledge of English or German.   

Desirable is  

-experience with optimization of algorithms.  

-experience in statistical speech or language    processing, preferably speech recognition, speech    synthesis, speech dialogue systems or chatbots.  

-experience with Delphi or Turbo Pascal.   

Interested? We look forward to your application:  (preferably by e-mail)   

digital publishing AG  

Freddy Ertl  f.ertl@digitalpublishing.de  

Tumblinger Straße 32  

D-80337 München Germany

Motorola Labs - Center for Human Interaction Research (CHIR) 

located in Schaumburg Illinois, USA, 

is offering summer intern positions in 2008 (12 weeks each). 

CHIR's mission

Our research lab develops technologies that provide access to rich communication, media and 

information services effortless, based on natural, intelligent interaction. Our research 

aims on systems that adapt automatically and proactively to changing environments, device 

capabilities and to continually evolving knowledge about the user.

Intern profiles

1) Acoustic environment/event detection and classification. 

Successful candidate will be a PhD student near the end of his/her PhD study and is skilled 

in signal processing and/or pattern recognition; he/she knows Linux and C/C++ programming. 

Candidates with knowledge of acoustic environment/event classification are preferred. 

2) Speaker adaptation for applications on speech recognition and spoken document retrieval

The successful candidate must currently be pursuing a Ph.D. degree in EE or CS with complete 

understanding and hand-on experience on automatic speech recognition related research. Proficiency 

in Linux/Unix working environment and C/C++ programming. Strong GPA. A strong background in speaker 

adaptation is highly preferred.

3) Development of voice search-based web applications on a smartphone 

We are looking for an intern candidate to help create an "experience" prototype based on our 

voice search technology. The app will be deployed on a smartphone and demonstrate intuitive and 

rich interaction with web resources. This intern project is oriented more towards software engineering 

than research. We target an intern with a master's degree and strong software engineering background. 

Mastery of C++ and experience with web programming (AJAX and web services) is required. 

Development experience on Windows CE/Mobile desired.

4) Integrated Voice Search Technology For Mobile Devices

Candidate should be proficient in information retrieval, pattern recognition and speech recognition. 

Candidate should program in C++ and script languages such as Python or Perl in Linux environment. 

Also, he/she should have knowledge on information retrieval or search engines.

We offer competitive compensation, fun-to-work environment and Chicago-style pizza.

If you are interested, please send your resume to:

Dusan Macho, CHIR-Motorola Labs

Email: dusan [dot] macho [at] motorola [dot] com

Tel: +1-847-576-6762

Training Generative Bayesian Networks with Missing Data

Bayesian Methods for Generalization in Reinforcement Learning

Carnegie Mellon University: Language Technologies Institute
Speech Faculty Position

The Language Technologies Institute (LTI), a department in the School of Computer Science at Carnegie Mellon University invites applications for a tenure or research track faculty position, starting on or around August, 2008. We are particularly interested in candidates at the Assistant Professor level for tenure track or research track, and specializing in the area of Speech Recognition. Applicants should have a Ph.D. in Computer Science, or a closely related subject.

Preference will be given to applicants with a strong focus on new aspects of speech recognition such as finite state models, active learning, discriminative training, adaptation techniques.

The LTI offers two existing speech recognition engines, JANUS and SPHINX, which are integrated into a wide range of speech applications including speech-to-speech translation and spoken dialog systems.

The LTI is the largest department of its kind with more than 20 faculty and 100 graduate students covering all areas of language technologies, including speech, translation, natural language processing, information retrieval, text mining, dialog, and aspects of computational biology. The LTI is part of Carnegie Mellon's School of Computer Science, which has hundreds of faculty and students in a wide variety of areas, from theoretical computer science and machine learning to robotics, language technologies, and human-computerinteraction.

Please follow the instructions for faculty applications to the School of Computer Science, explicitly mentioning LTI, at: http://www.cs.cmu.edu/~scsdean/FacultyPage/scshiringad08.html, and also notify the head of the LTI search committee by email, Alan W Black (awb@cs.cmu.edu) or Tanja Schultz (tanja@cs.cmu.edu) so that we will be looking for your application. Electronic submissions are greatly preferred but if you wish to apply on paper, please send two copies of your application materials, to the School of Computer Science
               
1. Language Technologies Faculty Search Committee
   School of Computer Science
   Carnegie Mellon University
   5000 Forbes Avenue
   Pittsburgh, PA 15213-3891

               
Each application should include curriculum vitae, statement of research and teaching interests, copies of 1-3 representative papers, and the names and email addresses of three or more individuals who you have asked to provide letters of reference. Applicants should arrange for reference letters to be sent directly to the Faculty Search Committee (hard copy or email), to arrive before March 31, 2008. Letters will not be requested directly by the Search Committee. All applications should indicate citizenship and, in the case of non-US citizens, describe current visa status.
               
Applications and reference letters may be submitted via email (word or .pdf format) to lti-faculty-search@cs.cmu.edu
 

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 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 immediately (April 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

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 Swedish language, for a 1 month contract, to work in speech technology related development projects. The successful candidate should have the following requirements:

·        Be native or near native Swedish speaker

·         Have a university degree in Linguistics or related field (preferably in Swedish 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 May 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

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 Dutch language, for a 1 month contract, to work in speech technology related development projects. The successful candidate should have the following requirements:

·        Be native or near native Dutch speaker

·         Have a university degree in Linguistics or related field (preferably in Dutch 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 May 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.

POSTDOCTORAL FELLOWSHIP OPENING AT ICSI

The International Computer Science Institute (ICSI) invites applications for a Postdoctoral Fellow position in spoken language
processing. The Fellow will be working with Dilek Hakkani-Tur, along with other PhD student and international colleagues, in the area of information distillation. Some experience with machine learning for text categorization is required, along with strong capabilities in speech and language processing in general.

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. See
http://www.icsi.berkeley.edu to learn more about ICSI.

The ICSI Speech Group has been a source of novel approaches to speech and language processing since 1988. It is primarily known for its work in speech recognition, although it has housed major projects in speaker recognition,
metadata extraction, and language understanding in the last few years. The effort in information distillation will draw upon lessons learned in our previous work for language understanding.

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

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

Title: Solutions for non-linear acoustic echo.

Background:

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

Task:

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

About the Company:

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

Required skills:

-        Master degree

-        Strong background in signal processing.

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

-        Programming: Matlab, C.

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

-        Language: English

Length of the PhD: 3 years

Place: Infineon Technologies France, Sophia-Antipolis

Contact:

Christophe Beaugeant

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

E-mail : christophe.beaugeant@infineon.com

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

Ph.D. position

in the field of

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

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

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

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

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

Applicants are expected to have:

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

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

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

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

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

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

Ph.D. position

in the field of

Multimodal Dialog Systems

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

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

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

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

Applicants are expected to have:

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

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

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

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

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

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

Offre d’Allocation de Recherche – Rentrée Universitaire 2008

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

Marc Sato, Chargé de Recherche CNRS

Jean-Luc Schwartz, Directeur de Recherche CNRS

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

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

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

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

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

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

Contact pour cette annonce:

Marc Sato

GIPSA-Lab, UMR CNRS 5216

Université Stendhal

BP 25 - 38040 Grenoble cedex 9

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

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

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

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

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

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

DeadLine: 10/06/2008

Contacts: jerome.farinas@irit.fr

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

DESCRIPTION DU SUJET :

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

ENGLISH VERSION:

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

CONNAISSANCES ET COMPETENCES REQUISES

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

• compétences en linguistique (phonologie, prosodie)

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


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

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

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

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

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

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

Head of NLP :

 We are now looking for a very experienced Computational Linguist

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

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

 opportunities for applying linguistic techniques are virtually

 limitless and even small improvements in the algorithms for

 detecting and correcting potential conversion errors translate

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

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

 forward for this department.

 Experience Needed:

 Grammars and parsing for spontaneous speech.

 Statistical methods.

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

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

 Experience with Automatic Speech Recognition technology.

 Probabilistic Language Modelling

 Phonetics

 Multi-lingual

info@voxid.co.uk 

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

A SPECIAL ISSUE OF THE JOURNAL TAL (Traitement Automatique des Langues)

Guest Editors: Bernard Laks and Noël Nguyen


There are long-established connections between research on the sound shape of language and natural language processing (NLP), for which one of the main driving forces has been the design of automatic speech synthesis and recognition systems. Over the last few years, these connections have been made yet stronger, under the influence of several factors. A first line of convergence relates to the shared collection and exploitation of the considerable resources that are now available to us in the domain of spoken language. These resources have come to play a major role both for phonologists and phoneticians, who endeavor to subject their theoretical hypotheses to empirical tests using large speech corpora, and for NLP specialists, whose interest in spoken language is increasing. While these resources were first based on audio recordings of read speech, they have been progressively extended to bi- or multimodal data and to spontaneous speech in conversational interaction. Such changes are raising theoretical and methodological issues that both phonologists/phoneticians and NLP specialists have begun to address.

Research on spoken language has thus led to the generalized utilization of a large set of tools and methods for automatic data processing and analysis: grapheme-to-phoneme converters, text-to-speech aligners, automatic segmentation of the speech signal into units of various sizes (from acoustic events to conversational turns), morpho-syntactic tagging, etc. Large-scale corpus studies in phonology and phonetics make an ever increasing use of tools that were originally developed by NLP researchers, and which range from electronic dictionaries to full-fledged automatic speech recognition systems. NLP researchers and phonologists/phoneticians also have jointly contributed to developing multi-level speech annotation systems from articulatory/acoustic events to the pragmatic level via prosody and syntax.

In this scientific context, which very much fosters the establishment of cross-disciplinary bridges around spoken language, the knowledge and resources accumulated by phonologists and phoneticians are now being put to use by NLP researchers, whether this is to build up lexical databases from speech corpora, to develop automatic speech recognition systems able to deal with regional variations in the sound pattern of a language, or to design talking-face synthesis systems in man-machine communication.

LIST OF TOPICS

The goal of this special issue will be to offer an overview of the interfaces that are being developed between phonology, phonetics, and NLP. Contributions are therefore invited on the following topics:

. Joint contributions of speech databases to NLP and phonology/phonetics

. Automatic procedures for the large-scale processing of multi-modal databases

. Multi-level annotation systems

. Research in phonology/phonetics and speech and language technologies: synthesis, automatic recognition

. Text-to-speech systems

. NLP and modelisation in phonology/phonetics

Papers may be submitted in English (for non native speakers of French only) or French and will relate to studies conducted on French, English, or other languages. They must conform to the TAL guidelines for authors available at http://www.atala.org/rubrique.php3?id_rubrique=1.

DEADLINES

. 11 February 2008: Reception of contributions
. 11 April 2008: Notification of pre-selection / rejection
. 11 May 2008: Reception of pre-selected articles
. 16 June 2008: Notification of final acceptance
. 30 June 2008: Reception of accepted articles' final versions

This special issue of Traitement Automatique des Langues will appear in autumn 2008.

THE JOURNAL

TAL (Traitement Automatique des Langues / Natural Language Processing, http://www.atala.org/rubrique.php3?id_rubrique=1)is a forty-year old international journal published by ATALA (French Association for Natural Language Processing) with the support of CNRS (French National Center for Scientific Research). It has moved to an electronic mode of publication, with printing on demand. This affects in no way its reviewing and selection process.

SCIENTIFIC COMMITTEE

. Martine Adda-Decker, LIMSI, Orsay
. Roxane Bertrand, LPL, CNRS & Université de Provence
. Philippe Blache, LPL, CNRS & Université de Provence
. Cédric Gendrot, LPP, CNRS & Université de Paris III
. John Goldsmith, University of Chicago
. Guillaume Gravier, Irisa, CNRS/INRIA & Université de Rennes I
. Jonathan Harrington, IPS, University of Munich
. Bernard Laks, MoDyCo, CNRS & Université de Paris X
. Lori Lamel, LIMSI, Orsay
. Noël Nguyen, LPL, CNRS & Université de Provence
. François Pellegrino, DDL, CNRS & Université de Lyon II
. François Poiré, University of Western Ontario
. Yvan Rose, Memorial University of Newfoundland
. Tobias Scheer, BCL, CNRS & Université de Nice
. Atanas Tchobanov, MoDyCo, CNRS & Université de Paris X
. Jacqueline Vaissière, LPP, CNRS & Université de Paris III
. Nathalie Vallée, DPC-GIPSA, CNRS & Université de Grenoble III

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

Guest Editors:

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

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

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

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

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

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

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

-----
Scope
-----

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

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

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

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

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

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

*CALL FOR PAPERS*

Special Issue of Speech Communication

on

*Spoken Language Technology for Education*



*Guest-editors:*

Maxine Eskenazi, Associate Teaching Professor, Carnegie Mellon University

Abeer Alwan, Professor, University of California at Los Angeles

Helmer Strik, Assistant Professor, University of Nijmegen

 

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

 

- Use of speech technology for education

- Use of spoken language dialogue for education

- Applications using speech and natural language processing for education

- Intelligent tutoring systems using speech and natural language

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

- Assessment of tutoring software

- Assessment of student performance

 

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

Deadline for submissions: June 1, 2008.

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

Deadline for revisions of papers: November 31, 2008.

 

*Submission instructions:*

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

SIGDIAL 2008 9th SIGdial Workshop on Discourse and Dialogue
COLUMBUS, OHIO; June 19-20 2008 (with ACL/HLT 2008)
    http://www.sigdial.org/workshops/workshop9



Continuing with a series of successful workshops in Antwerp, Sydney,
Lisbon, Boston, Sapporo, Philadelphia, Aalborg, and Hong Kong, this
workshop spans the ACL and ISCA SIGdial interest area of discourse and
dialogue. This series provides a regular forum for the presentation of
research in this area to both the larger SIGdial community as well as
researchers outside this community. The workshop is organized by
SIGdial, which is sponsored jointly by ACL and ISCA. SIGdial 2008 will
be a workshop of ACL/HLT 2008.


TOPICS OF INTEREST

We welcome formal, corpus-based, implementation or analytical work on
discourse and dialogue including but not restricted to the following
three themes:

1. Discourse Processing and Dialogue Systems

Discourse semantic and pragmatic issues in NLP applications such as
text summarization, question answering, information retrieval
including topics like:

- Discourse structure, temporal structure, information structure
- Discourse markers, cues and particles and their use
- (Co-)Reference and anaphora resolution, metonymy and bridging
  resolution
- Subjectivity, opinions and semantic orientation

Spoken, multi-modal, and text/web based dialogue systems including
topics such as:

- Dialogue management models;
- Speech and gesture, text and graphics integration;
- Strategies for preventing, detecting or handling miscommunication
  (repair and correction types, clarification and under-specificity,
  grounding and feedback strategies);
- Utilizing prosodic information for understanding and for
  disambiguation;


2. Corpora, Tools and Methodology

Corpus-based work on discourse and spoken, text-based and multi-modal
dialogue including its support, in particular:

- Annotation tools and coding schemes;
- Data resources for discourse and dialogue studies;
- Corpus-based techniques and analysis (including machine learning);
- Evaluation of systems and components, including methodology, metrics
  and case studies;


3. Pragmatic and/or Semantic Modeling

The pragmatics and/or semantics of discourse and dialogue (i.e. beyond
a single sentence) including the following issues:

- The semantics/pragmatics of dialogue acts (including those which are
  less studied in the semantics/pragmatics framework);
- Models of discourse/dialogue structure and their relation to
  referential and relational structure;
- Prosody in discourse and dialogue;
- Models of presupposition and accommodation; operational models of
  conversational implicature.


SUBMISSIONS

The program committee welcomes the submission of long papers for full
plenary presentation as well as short papers and demonstrations. Short
papers and demo descriptions will be featured in short plenary
presentations, followed by posters and demonstrations.

- Long papers must be no longer than 8 pages, including title,
  examples, references, etc. In addition to this, two additional pages
  are allowed as an appendix which may include extended example
  discourses or dialogues, algorithms, graphical representations, etc.
- Short papers and demo descriptions should aim to be 4 pages or less
  (including title, examples, references, etc.).

Please use the official ACL style files:
    http://www.ling.ohio-state.edu/~djh/acl08/stylefiles.html

Submission/Reviewing will be managed by the EasyChair system. Link to
follow.

Papers that have been or will be submitted to other meetings or
publications must provide this information (see submission
format). SIGdial 2008 cannot accept for publication or presentation
work that will be (or has been) published elsewhere. Any questions
regarding submissions can be sent to the co-Chairs.

Authors are encouraged to make illustrative materials available, on
the web or otherwise. For example, excerpts of recorded conversations,
recordings of human-computer dialogues, interfaces to working systems,
etc.


IMPORTANT DATES

Submission        Mar 14 2008
Notification      Apr 27 2008
Camera-Ready   May 16 2008
Workshop          June 19-20 2008


WEBSITES

Workshop website: http://www.sigdial.org/workshops/workshop9
Submission link: To be announced
SIGdial organization website: http://www.sigdial.org/
CO-LOCATION ACL/HLT 2008 website: http://www.acl2008.org/


CONTACT

For any questions, please contact the co-Chairs at:
Beth Ann Hockey  bahockey@ucsc.edu
David Schlangen  das@ling.uni-potsdam.de

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

                   Interspeech, 22-26 September 2008
                         Brisbane, Australia

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

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

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

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

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

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

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

Call for participation

            INFILE@CLEF2008

     Information, Filtering, Evaluation
    
          http://www.infile.org


INFILE welcomes participation of any institution to its first evaluation
campaign. This participation is free of charge and
participants can keep and use the development and evaluation data for
free after the evaluations for research and development purposes.

INFILE (INformation, Filtrage, Evaluation) is a cross-language adaptive
filtering evaluation campaign jointly organized by CEA,
Université de Lille 3 and ELDA. It is organized as a pilot track in CLEF
2008 and is supported by NIST TREC.

INFILE extends the last filtering track of TREC 2002 in the following ways:
-INFILE is crosslingual (English, French and Arabic); a corpus of 
100,000 comparable news-wire stories from Agence France Presse (AFP)
 for each language is used for the evaluations.

-Evaluation will be performed using an automatic interrogation of test 
systems with a simulated user feedback. Each system will be able to
 use the feedback at any time to increase performance.

The participant systems will have to provide a Boolean decision for each
document according to each filtering  profile. A curve of the evolution
of efficiency will be computed.
Although cross-lingual systems are encouraged, the campaign is also open
to monolingual systems.


Tasks and languages
-------------------
Two tasks and three languages are considered. The first task is
Information Filtering on general news and events. For this task,
participants will have to classify each transmitted news-wire into
zero, one or more different profiles. 30 general profiles will be made
available in 3 languages (Arabic, English and French).

The second task is Information Filtering on science and technology
domain. Participants will have to associate to each news-wire zero, one
or more science and technology profiles.
A total of 20 profiles will be available in 3 languages (Arabic, English
and French).

For each task, participants are free to register to monolingual
filtering (e.g. information filtering using profiles and news-wires in
the same language) or to crosslingual filtering (e.g. information
filtering according to profiles in one language and news-wires in
another language).

Corpus
------
The corpus consists of 300,000 news-wires in Arabic, English and French
from the news agency Agence France Presse covering the 2004-2006 period.
The news-wires are related to general news and events information and
are comparable between Arabic, English and French.


Protocol description
---------------------
General information about the domain of profiles is given to each
participant. 15 days afterwards, 50 profiles are given to participants
(30 general
profiles and 20 profiles related to science and technology). Profiles
are composed of a list of keywords (simple and complex noun phrases) and
up to 3
documents illustrating each profile.
 
Then, news-wires are transmitted by the organizer to an automated 
interface of each participating system. The interface returns a
 Boolean response for each profile. After reception of this response,
and if requested by the participant, the organizer sends a feedback
 consisting of expected profile assignments for each document 
submitted. Participants may adapt their system at any time using this
feedback.


Important Dates
---------------
    Registration Opens - Feb 11th, 2008
    Dry Run - June 2nd to June 14th, 2008
    Evaluation Run June 30th - July 19th
    Release of Human Assessments and Individual Results - August 4th, 2008
    Submission of Paper for Working Notes - 15 August 2008
    Workshop - 17-19 September 2008 CLEF Workshop

Contact
-------
    info@infile.org
    http://www.infile.org

CALL FOR PARTICIPATION

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

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

 ----%---

 I will attend the workshop.

 Name:

 Affiliation:

 Email:

 ---%---

SPEECH PRODUCTION WORKSHOP: INSTRUMENTATION-BASED APPROACH

Date: 5th JULY, 2008 (Saturday)

Location: ILPGA 19 rue des Bernardins 75005 Paris

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

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

[Voice Production]

J. Ohala      (to be announced)

  D. Demolin    Subglottal air pressure measurement

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

[Articulation 1]

  A. Marchal    Electropalatography, new development

 M-O. Berger   Multimodality acquisition of articulatory data

 P. Hoole       3D electromagnetic articulography, recent progress

[Articulation 2]

  R. Sock        Digital X-ray cinematography and database

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

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

[Demonstration]

  K. Honda      Noninvasive photoglottographic technique

  H. Takemoto   MRI-based vocal-tract solid models

Organizers

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

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

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

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

Titre et résumé à préciser

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

ACL-08: HLT Workshop on

Mobile Language Processing

http://www.mobileNLPworskshop.org

Columbus, Ohio, United States

June 19th or June 20th, 2008

** Paper submission deadline: March 7, 2007 **

************************************************************************

Mobile Language Processing

Mobile devices, such as ultra-mobile PCs, personal digital assistants, and smart phones have many unique characteristics that make them both highly desirable as well as difficult to use. On the positive side, they are small, convenient, personalizable, and provide an anytime-anywhere communication capability. Conversely, they have limited input and output capabilities, limited bandwidth, limited memory, and restricted processing power. 

The purpose of this workshop is to provide a forum for discussing the challenges in natural and spoken language processing and systems research that are unique to this domain. We argue that mobile devices not only provide an ideal opportunity for language processing applications but also offer new challenges for NLP and spoken language understanding research.

For instance, mobile devices are beginning to integrate sensors (most commonly for location detection through GPS, Global Positioning Systems) that can be exploited by context/location aware NLP systems; another interesting research direction is the use of information from multiple devices for "distributed" language modeling and inference. To give some concrete examples, knowing the type of web queries made from nearby devices or from a specific location or a specific ‘context' can be combined for various applications and could potentially improve information retrieval results. Learned language models could be transferred from device to device, propagating and updating the language models continuously and in a decentralized manner.

Processing and memory limitations incurred by executing NLP and speech recognition on small devices need to be addressed. Some applications and practical considerations may require a client/server or distributed architecture: what are the implications for language processing systems in using such architectures?

The limitation of the input and output channels necessitates typing on increasingly smaller keyboards which is quite difficult, and similarly reading on small displays is challenging. Speech interfaces for dictation or for understanding navigation commands and/or language models for typing suggestions would enhance the input channel, while NLP systems for text classification, summarization and information extraction would be helpful for the output channel.  Speech interfaces, language generation and dialog systems would provide a natural way to interact with mobile devices.

Furthermore, the growing market of cell phones in developing regions can be used for delivering applications in the areas of health, education and economic growth to rural communities.  Some of the challenges in this area are the limited literacy, the many languages and dialects spoken and the networking infrastructure.

We solicit papers on topics including, but not limited to the following:

·       Special challenges of NLP for mobile devices

·       Applications of NLP for mobile devicesNLP enhanced by sensor data

·       Distributed NLP

·       Speech and multimodal interfaces

·       Machine translation

·       Language model sharing

·       Applications for the developing regions

The goal of this one day workshop is to provide a forum to allow both industrial and academic researchers to share their experiences and visions, to present results, compare systems, exchange ideas and formulate common goals.

"Keynote Speaker: Dr. Lisa Stifelman, Principal User Experience Manager at Tellme/Microsoft. The title of her talk is soon to be announced."

 - Paper submission deadline: March 7, 2008

- Notification of acceptance: April 8, 2008
- Camera-ready Copy: April 18, 2008

Organizing committee:

Rosario, Barbara        Intel Research    

Paek, Tim               Microsoft Research    

Contact

For questions about the workshop, please contact Barbara Rosario (barbara.rosario@intel.com).

 We accept position papers (2 pages), short research or demo papers (4 pages), and regular papers (8 content pages with 1 extra page for references). Papers must be submitted through the submission system at
https://www.softconf.com/acl08/ACL08-WS07/submit.html

Please use the LaTeX or Microsoft Word style files available at http://ling.osu.edu/acl08/stylefiles.html.

Important dates:

Following previous successful workshops between 1999 and 2006, the

4TH TUTORIAL AND RESEARCH WORKSHOP
PERCEPTION AND INTERACTIVE TECHNOLOGIES FOR SPEECH-BASED SYSTEMS
(PIT08)

will be held at the Kloster Irsee in southern Germany from June 16 to
June 18, 2008.

Please follow this link to visit our workshop website
http://it.e-technik.uni-ulm.de/World/Research.DS/irsee-workshops/pit08/introduction.html

Submissions will be short/demo or full papers of 4-10 pages.

Important dates:
**February 22, 2008: Deadline for Long, Short and Demo Papers**
March 15, 2008: Author notification
April 1, 2008: Deadline for final submission of accepted paper
April 18, 2008: Deadline for advance registration
June 7, 2008: Final programme available on the web

The workshop will be technically co-sponsored by the IEEE Signal
Processing Society. It is envisioned to publish the proceedings in the
LNCS/LNAI Series by Springer.

We welcome you to the workshop.

Elisabeth André, Laila Dybkjaer, Wolfgang Minker, Heiko Neumann,
Michael Weber, Roberto Pieraccini

PIT'08 Organising Committee

- 
Wolfgang Minker
University of Ulm
Department of Information Technology
Albert-Einstein-Allee 43
D-89081 Ulm
Phone: +49 731 502 6254/-6251
Fax:   +49 691 330 3925516
http://it.e-technik.uni-ulm.de/World/Research.DS/

Young Researchers' Roundtable on Spoken Dialog Systems
June 21st, 2008, Columbus, Ohio,

Second Call for Participation

The Young Researchers' Roundtable on Spoken Dialog Systems is an annual
workshop designed for students, post docs, and junior researchers
working in research related to spoken dialogue systems in both academia
and industry. The roundtable provides an open forum where participants
can discuss their research interests, current work and future plans. The
workshop is meant to provide an interdisciplinary forum for creative
thinking about current issues in spoken dialogue systems research, and
help create a stronger international network of young researchers
working in the field. The workshop is co-located with ACL 2008  and
will occur the day after the 9th SIGdial workshop.

Workshop Format

Workshop events will include small informal discussion groups, a larger
Question & Answers style discussion with senior researchers from
academia and industry, and an opt-in demo presentation session. There
will also be time for participants to have informal discussions over
coffee with senior researchers on potential career opportunities. The
small discussion groups are intended to allow participants to exchange
ideas on key research topics, and identify issues that are likely to be
important in the coming years. The results of each discussion group will
then presented and discussed in plenary sessions. The topics for
discussion are still open and will be determined by participant
submissions and finalized online before the workshop. Potential
participants should submit a short paper, as described below in the
submission process to get accepted to the workshop.

In addition to the traditional one day event, a half day extension on
the topic of  "Frameworks and Grand Challenges for Dialog System
Evaluation" is under consideration for the morning of June 22nd, 2008.
The aim of this extra extension is to provide an opportunity for dialog
systems researchers to discuss issues of evaluation, and hopefully
determine an agenda for a future evaluation event or framework.
Organization of this extended event will depend on interest; we
therefore, as described below, invite potential participants to indicate
their interest with their YRR08 submission.

Submission Process

We invite participation from students, post docs, and junior researchers
in academia or industry who are currently working in spoken dialog
systems research. We also invite participation from those who are
working in related fields such as linguistics, psychology, or speech
processing, as applied to spoken dialogue systems.  Please note that by
'young researchers' the workshop's organizers mean to target students
and researchers in the field who are at a relatively early stage of
their careers, and in no way mean to imply that participants must meet
certain age restrictions.

Potential participants should submit a 2-page position paper and suggest
topics for discussion and whether they would be interested in attending
the extended session on Sunday morning. A template and specific
submission instructions will be available on http://www.yrrsds.org/ on
March 1.  Submissions will be accepted on a rolling basis from that day
until the maximum number of participants for the workshop (50) is
reached, or until the submission deadline (May 10th, 2008) is reached.
Proceedings from previous years' workshops are also available on our web
site. Specific questions can be directed to the organizing committee at
yrr08-organizers__AT_googlegroups_DOT_com

Important Dates

Submissions:            March 1st, 2008
Submissions deadline:   May 10th, 2008
Notification:           May 20th, 2008
Registration begins:    to be announced
Workshop:               June 21st, 2008

Organizing Committee

Hua Ai, Intelligent Systems Program, University of Pittsburgh, USA
Carlos Gómez Gallo, Department of Computer Science, University of
Rochester, USA
Robert J. Ross, Department of Computer Science, University of Bremen,
Germany
Sabrina Wilske, Department of Computational Linguistics, Saarland
University, Germany
Andi Winterboer, Institute for Communicating and Collaborative Systems,
University of Edinburgh, UK
Craig Wootton, University of Ulster, Belfast, Northern Ireland

Local Organization

Tim Weale, Department of Computer Science and Engineering, The Ohio
State University, USA

Second announce of the AERFAI (Spanish
IAPR) Summer School "New Trends in Pattern Recognition for Language
Technologies" to be held in Bilbao in June 23th-28th.

url: http://www.ehu.es/aerfaiss08

SIGIR-08 Workshop

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

http://and2008workshop.googlepages.com/

Call for Papers


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

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

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

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

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

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

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

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


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

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

eNTERFACE'08 the next international summer workshop on multimodal

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

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

http://entreface08.limsi.fr

Please consider proposing projects and participate to the workshop (see

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

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

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

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

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

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

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

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

Foundation.

eNTERFACE'08 Important Dates

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

the format provided by the Author's kit

. January 10rd, 2008: Notification of project acceptance

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

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

Christophe d'Alessandro

CNRS-LIMSI, BP 133 - F91403 Orsay France

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

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

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

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

ANALYSIS AND UNDERSTANDING OF CONTENT

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

INTEGRATION OF MULTIPLE SEMANTIC REPRESENTATIONS

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

SEMANTIC INTERFACES

• Natural-Language Interface
• Multimodal Interfaces

APPLICATIONS

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

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

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

In addition to Technical Papers, the conference will feature

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

SUBMISSIONS

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

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

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

- http://www.eusipco2008.org/

DEADLINE FOR SUBMISSION: February 8, 2008

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

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

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

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

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

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

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

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

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

IMPORTANT DATES:

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

Submission deadline of proposals for tutorials: February 8, 2008

Notification of Acceptance: April 30, 2008

Conference: August 25-29, 2008

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

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

Brno, Czech Republic, 8-12 September 2008

http://www.tsdconference.org/

The conference is organized by the Faculty of Informatics, Masaryk

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

West Bohemia, Pilsen. The conference is supported by International

Speech Communication Association.

Venue: Brno, Czech Republic

TSD SERIES

TSD series evolved as a prime forum for interaction between

researchers in both spoken and written language processing from the

former East Block countries and their Western colleagues. Proceedings

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

in Artificial Intelligence (LNAI) series.

TOPICS

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

text corpora and tagging

transcription problems in spoken corpora

sense disambiguation

links between text and speech oriented systems

parsing issues

parsing problems in spoken texts

multi-lingual issues

multi-lingual dialogue systems

information retrieval and information extraction

text/topic summarization

machine translation

semantic networks and ontologies

semantic web

speech modeling

speech segmentation

speech recognition

search in speech for IR and IE

text-to-speech synthesis

dialogue systems

development of dialogue strategies

prosody in dialogues

emotions and personality modeling

user modeling

knowledge representation in relation to dialogue systems

assistive technologies based on speech and dialogue

applied systems and software

facial animation

visual speech synthesis

Papers on processing of languages other than English are strongly

encouraged.

PROGRAM COMMITTEE

Frederick Jelinek, USA (general chair)

Hynek Hermansky, Switzerland (executive chair)

FORMAT OF THE CONFERENCE

The conference program will include presentation of invited papers,

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

be presented in plenary or topic oriented sessions.

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

for additional informal interactions.

CONFERENCE PROGRAM

The conference program will include oral presentations and

poster/demonstration sessions with sufficient time for discussions of

the issues raised.

IMPORTANT DATES

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

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

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

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

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

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

demonstrations sent to the authors

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

The contributions to the conference will be published in proceedings

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

conference.

OFFICIAL LANGUAGE

of the conference will be English.

ADDRESS

All correspondence regarding the conference should be

addressed to

Dana Hlavackova, TSD 2008

Faculty of Informatics, Masaryk University

Botanicka 68a, 602 00 Brno, Czech Republic

phone: +420-5-49 49 33 29

fax: +420-5-49 49 18 20

email: tsd2008@tsdconference.org

LOCATION

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

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

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

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

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

region.

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

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

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



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



Topics of Interest:

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



Submissions:

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

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



Key Dates:

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



Organising Committee:

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



Programme Committee:

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



Websites:

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

10-13 September 2008, Zadar, Croatia

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

TECHNICAL CO-SPONSORS

IEEE Region 8

EURASIP - European Assoc. Signal, Speech and Image Processing

IEEE Croatia Section

IEEE Croatia Section Chapter of the Signal Processing Society

IEEE Croatia Section Joint Chapter of the AP/MTT Societies

TOPICS

--> Image and Video Processing

--> Multimedia Communications

--> Speech and Audio Processing

--> Wireless Commununications

--> Telecommunications

--> Antennas and Propagation

--> e-Learning and m-Learning

--> Navigation Systems

--> Ship Electronic Systems

--> Power Electronics and Automation

--> Naval Architecture

--> Sea Ecology

--> Special Session Proposals - A special session consist

of 5-6 papers which should present a unifying theme

from a diversity of viewpoints; deadline for proposals

is February 04, 2008.

KEYNOTE TALKS

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

Image Processing using Quadratic Volterra Filters

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

of Technology, AUSTRIA:

Testbeds and Rapid Prototyping in Wireless Systems

* Professor Paul Cross, University College London, UK:

GNSS Data Modeling: The Key to Increasing Safety and

Legally Critical Applications of GNSS

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

The Role of Resonators in the Generation of Voice

Signals

SPECIAL SESSIONS

SS1: "VISNET II - Networked Audiovisual Systems"

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

Systems Research, University of Surrey, UNITED KINGDOM

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

SS2: "Computer Vision in Art"

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

University of Ljubljana, Faculty of Computer and Information

Science, Computer Vision Laboratory, SLOVENIA

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

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

SUBMISSION

Papers accepted by two reviewers will be published in

symposium proceedings available at the symposium and

abstracted/indexed in the INSPEC and IEEExplore database.

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

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

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

postal submissions will be accepted. Authors should prepare

their papers according to ELMAR-2008 paper sample, convert

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

web-based submission system by March 03, 2008.

SCHEDULE OF IMPORTANT DATES

Deadline for submission of full papers: March 03, 2008

Notification of acceptance mailed out by: April 21, 2008

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

Preliminary program available online by: May 12, 2008

Registration forms and payment deadline: May 19, 2008

Accommodation deadline: June 02, 2008

GENERAL CO-CHAIRS

Ive Mustac, Tankerska plovidba, Zadar, Croatia

Branka Zovko-Cihlar, University of Zagreb, Croatia

PROGRAM CHAIR

Mislav Grgic, University of Zagreb, Croatia

CONTACT INFORMATION

Assoc.Prof. Mislav Grgic, Ph.D.

FER, Unska 3/XII

HR-10000 Zagreb

CROATIA

Telephone: + 385 1 6129 851=20

Fax: + 385 1 6129 568=20

E-mail: elmar2008 (_) fer.hr

For further information please visit:

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

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

"Dynamique de la nasalité"

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

La nasalité et les processus de nasalisation dans les langues sont des phénomènes étudiés dans différents champs disciplinaires proches ou éloignés, et qui ne se rencontrent pas forcément (dialectologie, typologie linguistique, phonétique acoustique et articulatoire, phonologie, morphophonologie, linguistique historique, traitement du signal…). Les disciplines impliquées dans l’étude de ces processus sont des domaines de recherche qui ont chacun leur réseau de diffusion scientifique, avec des publications spécifiques, des colloques et autres manifestations scientifiques propres à chaque discipline, très souvent même des structures de laboratoires disjointes. Les espaces de production et de diffusion du savoir distincts rendent difficiles les échanges entre chercheurs de ces communautés. Pourtant, dans ces disciplines, les spécialistes s’accordent à reconnaître que les processus de nasalisation sont des phénomènes dynamiques complexes dont les mécanismes linguistiques, physiques et physiologiques ne sont pas encore clairement élucidés. Les objectifs de cette école sont d’accompagner et de favoriser le rapprochement de ces disciplines connexes, de faire se rencontrer des communautés scientifiques qui évoluent chacune dans des réseaux spécifiques, de connaître les terminologies, les formalismes, d’échanger les outils et les données, de stimuler les interactions et favoriser la transdisciplinarité des projets de recherche. Cette école sera aussi le moyen de susciter la création de réseaux d’échanges et de collaboration, comme de projets de recherche transdisciplinaires.

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

Possibilité d’accueil : 50 participants

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

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

Invited Speakers:

·         Nestor Becerra (Universidad de Santiago, Chile)

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

·         Giussepe Ricardi (University of Trento, Italy)

Next Generation Spoken Language Interfaces

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

Embodied conversational agents  in verbal and non-verbal communication

·         Yannis Stilianou (University of Crete, Grecia)

Voice Conversion: State of the art and Perspectives

Important dates:

·         Full paper submission: July 20, 2008

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

Contact information:

VJTH’2008

Dept. Electronics and Telecommunications

Faculty of Engineering

Alda. Urkijo s/n

48013 Bilbao

Tel.: +34 946 013 969

Fax.: +34 946 014 259

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

EVALUATION CAMPAIGNS                ALBAYZIN-08 System Evaluation Proposal

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

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

• Language recognition

Registration Form

• Speech synthesis

Registration Form

• Automatic text translation

Registration Form

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

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