ISCApad115

Number: 115	Date: 01/12/2008
Author: Chris Wellekens

Editor: Chris Wellekens

Dear Members,

Happy New Year to all members! Our Chinese colleagues will have the luck to celebrate New Year again in February!

Thanks to all of you who reported their interest in ISCApad by sending either congratulations or criticisms. They all helped us progress in our communication with the community.

The number of job offers is continually growing: it is certainly a proof of the confidence of employers in the efficiency of advertising in the ISCA community. I also hope that it is a proof of good health of speech science and technology.

The rocketing prices of oil will have a major impact on our transportation costs: let us wish that our annual conferences and workshops will attract as many people as in the previous years. We all know that personal contacts bring more to the cohesion of the community than any other type of communication. But there is still a bright future for a newsletter!

Professor em. Chris Wellekens

Institut Eurecom France

Dear Members,

Happy New Year to all members!

Thanks to all of you who reported their interest for ISCApad by sending either congratulations or criticisms. They all helped us progress in our communication with the community.

The number of job offers is continuously growing: it is certainly a proof of the confidence of employers in the efficiency of advertising in the ISCA community. I also hope that it is a proof of good health of speech science and technology.

The rocketting prices of oil will bear a lot on our transportation costs: let us wish that our annual conferences and workshops will attract as many people as in the previous years. We all know that personal contacts bring more to the cohesion of the community than any other communication mean. But there is still a bright future for a newsletter!

Professor em. Chris Wellekens

Institut Eurecom France

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ISCA News

GOOGLE SCHOLAR AND ISCA ARCHIVE

Google Scholar and the ISCA Archive

The indexing of the ISCA Archive (http://www.isca-speech.org/archive/) by the Google Scholar search engine (http://scholar.google.com/) is now thorough enough to be quite useful, so this seems like a good time to give an overview of the service. Google Scholar is a research literature search engine that provides full-text search for ISCA papers whose full text cannot be searched with other search engines. Google Scholar's citation tracking shows what papers have cited a particular paper, which can be very useful for finding follow-up work, related work and corrections. More details about these and other features are given below.

The titles, author lists, and abstracts of ISCA Archive papers are all on the public web, so they can be searched by a general-purpose search engine such as Google. However, the full texts of most ISCA papers are password protected and thus cannot be searched with a general-purpose search engine. Google Scholar, through an arrangement with ISCA, has access to the full text of ISCA papers. Google Scholar has similar arrangements with many other publishers. (On the other hand, general-purpose search engines index all sorts of web pages and other documents accessible through the public web, many of which will not be in the Google Scholar index. So it's often useful to perform the same search using both Google Scholar and a general-purpose search engine.)

Google Scholar automatically extracts citations from the full text of papers. It uses this information to provide a "Cited by" list for each paper in the Google Scholar index. This is a list of papers that have cited that paper. Google Scholar also provides an automatically generated "Related Articles" list for each paper. The "Cited by" and "Related Articles" lists are powerful tools for discovering relevant papers. Furthermore, the length of a paper's "Cited by" list can be used as a convenient (although imperfect) measure of the paper's impact. Discussions about the subtleties of using Google Scholar to measure impact can be found at http://www.harzing.com/resources.htm#/pop_gs.htm and http://blogs.nature.com/nautilus/2007/07/google_scholar_as_a_measure_of.html.

It's possible to restrict Google Scholar searches to papers published by ISCA by using Google Scholar's Advanced Search feature and entering "ISCA" in the "Return articles published in" field. If "ISCA" is entered in that field, and nothing is entered in the main search field, then the search results will show what ISCA papers are the most highly cited.

It should be noted that that there are many papers on ISCA-related topics which are not in the Google Scholar index. For example, it seems many ICPhS papers are missing. And old papers which have been scanned in from paper copies will either not have their full contents indexed, or will be indexed using imperfect OCR technology. Furthermore, as of November 2007 the indexing of the ISCA Archive by Google Scholar is still not 100% complete. There are a few different areas which are not perfectly indexed, but the biggest planned improvement is to start using OCR for the ISCA papers which have been scanned in from paper copies.

There may be a time lag between when a new event is added to the ISCA Archive in the future and when it appears in the Google Scholar index. This time lag may be longer than the usual lag of general-purpose search engines such as Google, because ISCA must create Google Scholar catalog data for every new event and because the Google Scholar index seems to update considerably more slowly than the Google index.

Acknowledgements: ISCA's arrangement with Google Scholar is a project of students Rahul Chitturi, Tiago Falk, David Gelbart, Agustin Gravano, and Francis Tyers, ISCA webmaster Matt Bridger, and ISCA Archive coordinator Wolfgang Hess. Our thanks to Google's Christian DiCarlo and Darcy Dapra, and the rest of the Google Scholar team.
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SIG's activities

A list of Speech Interest Groups can be found on our web.
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Courses, Internships

Motorola Labs - Center for Human Interaction Research (CHIR) l

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

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Books, Databases, Softwares

Books

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
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Spotlight on LDC Programmers and Software Tools

Spotlight on LDC Programmers and Software Tools -

- LDC Member Survey -

- Membership Fee Increases and Discounts -

LDC2007T36
- Chinese Treebank 6.0 (CTB 6.0) -

LDC2007S11
- 2004 Spring NIST Rich Transcription (RT-04S) Development Data -

- LDC Offices to Close for Winter Break -

Spotlight on LDC Programmers and Software Tools

As part of our 15th Anniversary celebration, we have highlighted one aspect of the LDC in our monthly newsletters. These features provided our members and data users with a glimpse of the broad range of the LDC's research activities. The last feature of the year will focus on the LDC's software programmers and the tools they create.

A large segment of LDC's programming group is led by Senior Research Programmer Kazuaki Maeda. Besides being a programmer, Maeda is a linguist specializing in phonetics, phonology and computational linguistics. The group currently has ten full-time staff, augmented as necessary by part-time programmers. LDC's programmers are adept in all major programming languages and can work across platforms; their work supports virtually every aspect of LDC's operation. More information about LDC's programmers can be found on our staff page .

One of the programming group's principal responsibilities is to develop workflow management software and annotation and transcription tools to support projects such as GALE and LCTL . Our goal is to make tools developed for general use broadly available. One such tool is XTrans, a next generation transcription tool that is designed to support transcription tasks in multiple languages on multiple platforms. Its versatile and powerful waveform display/playback component can load multiple audio files of different file formats and sampling rates at the same time. The virtual channel supported by XTrans provides the most natural method for transcribing overlapping speech. Virtual channel represents an audio source, not a physical channel, that is identified and transcribed in a given audio recording. A single-channel audio file can contain many audio sources. For instance, a round-table talk show with five speakers contains five audio sources in a single-channel audio recording. With XTrans, that file is modeled as a 5-virtual-channel audio file, and each virtual channel is transcribed independently. Additionally, if a recording consists of audio files with different sampling rates, XTrans will automatically resample them to the same rate. The LDC has used XTrans for many varied projects, and the tool has proven to be quick to learn and easy to master. We are currently working through licensing issues with organizations that provided libraries for XTrans. Once those issues are resolved, we will make XTrans generally available.

Two other general use tools developed by the LDC -- The Annotation Graph Toolkit (AGTK) and Champollion Tool Kit (CTK) -- are available on Sourceforge.net Like XTrans, these tools represent creative solutions to difficult problems:


- The Annotation Graph Toolkit (AGTK) is a primary resource for annotation tool development at LDC. AGTK is a suite of software components for building tools for annotating linguistic signals, time-series data which documents any kind of linguistic behavior (e.g. audio, video). Unlike the traditional approach of designing and implementing data structures and user interfaces for new tasks from scratch, AGTK allows developers to quickly prototype tools and define data formats. The flexible nature of the AG model means that data representations can be rapidly modified in response to evolving annotation task definitions. AGTK allows for rapid deployment of highly specialized, task-specific tools that maximize user interface ergonomics and improve the speed and accuracy of annotation.
- Champollion Tool Kit (CTK) was developed to address issues in aligning parallel text consisting of remote language pairs and a significant amount of noise. To achieve high precision and recall on manually-aligned text, CTK assumes a noisy input, that is, that a sizable percentage of alignments will not be one to one, and that the number of deletions and insertions will be significant. Furthermore, CTK differs from other lexicon-based approaches in assigning greater weight to less frequent translation pairs. CTK was first evaluated using Chinese-English parallel text but is designed to be used on as many language pairs as possible.
XTrans, AGTK and CTK are representative of the work by LDC's programmers, making it possible for us to support projects of increasing complexity and to distribute a growing variety of linguistic resources. The LDC Catalog contains several publications which were created using software tools developed by LDC's programming group. These include ACE data, Arabic Treebank publications, and NIST Rich Transcription corpora.

LDC Member Survey

In order to determine how the consortium as a whole views the LDC, we are conducting a survey of our active users. Each person and organization who licensed data and/or purchased an LDC membership in 2006 and 2007 will have received an email on December 17 that contained a link to the online survey. Those who complete the survey before January 14, 2008 will be eligible to win a $500 benefit good towards any corpus or membership purchase in 2008. There will be a blind drawing in January 2008 and one winner will be selected from the pool of respondents. Based on last year's response rates, each respondent will have an approximate 1 in 100 chance of winning!

Membership Fee Increases and Discounts

The LDC will raise membership fees effective January 1, 2008. Please click here for new pricing information and options for obtaining discounts on membership fees.



New Publications




(1) The Chinese Treebank project began at the University of Pennsylvania in 1998 and continues at Penn and the University of Colorado. Chinese Treebank 6.0 is the latest version produced from this effort, consisting of 780,000 words (over 1.28 million Chinese characters) that are segmented, part-of-speech tagged and fully bracketed. 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 broadcast news programs.

This release encompasses 2,036 text files, containing 28,295 sentences, 781,351 words and 1,285,149 hanzi (Chinese characters). The data is provided in two encodings: GBK and UTF-8, and the annotation has Penn Treebank-style labeled brackets. The data is provided in four different formats: raw text, word segmented, word segmented and POS-tagged, and syntactically bracketed. Chinese Treebank 6.0 (CTB 6.0) is distributed via web download.

2007 Subscription Members will automatically receive two copies of this corpus on disc. 2007 Standard Members may request a copy as part of their 16 free membership corpora. Nonmembers may license this data for US$700.



*

(2) The 2004 Spring NIST Rich Transcription (RT-04S) Development Data contains the test material (meeting speech and reference transcripts) used in the RT-04S evaluation administered by the NIST (National Institute of Standards and Technology) Speech Group. Rich Transcription (RT) is broadly defined as a fusion of speech-to-text technology and metadata extraction technologies designed to provide the basis for a generation of more usable transcriptions of human-human meeting speech.

The RT-04S development data consists of approximately 10 minutes of recordings of eight meetings held at ISCI, CMU, LDC and NIST. Although the development data is comprised of 10-minute excerpts from the same data collection sites which are represented in LDC2007S12 2004 Spring NIST Rich Transcription (RT-04S) Evaluation Data, it is not completely reflective of the evaluation test data since it contains lapel mics in lieu of head mics for the LDC and CMU data and some different distant mics for LDC data.

RT-04S included the following tasks in the meeting domain:

Speech-to-Text Transcription (STT) tasks

Microphone conditions:
�         Multiple distant microphones
�         Single distant microphone
�         Individual head microphone

Processing time conditions:
�         Unlimited time STT
�         Less than or equal to twenty times realtime
�         Less than or equal to ten times realtime
�         Less than or equal to one times realtime

Diarization (SPKR) task (who spoke when)

Microphone conditions:
�         Multiple distant microphones
�         Single distant microphone

Input conditions:
�         Speech input only
�         Speech plus reference transcript input

Processing time conditions:
�         Unlimited time
�         Less than or equal to twenty times realtime
�         Less than or equal to ten times realtime
�         Less than or equal to one time realtime

2004 Spring NIST Rich Transcription (RT-04S) Development Data is distributed on one DVD-ROM.
2007 Subscription Members will automatically receive two copies of this corpus. 2007 Standard Members may request a copy as part of their 16 free membership corpora. Nonmembers may license this data for US$2000.

LDC Offices to Close for Winter Break

The LDC would like to inform our customers that we will be closed from December 24, 2007 through January 1, 2008 in accordance with the University of Pennsylvania Winter Break Policy. Our offices will reopen on Wednesday, January 2, 2008. Requests received for membership renewals and corpora will be processed at that time.

Best wishes for a happy and safe holiday season!

Ilya Ahtaridis
Membership Coordinator --------------------------------------------------------------------
```
Linguistic Data Consortium                     Phone: (215) 573-1275 University of Pennsylvania                       Fax: (215) 573-2175 3600 Market St., Suite 810                         ldc@ldc.upenn.edu Philadelphia, PA 19104 USA                  http://www.ldc.upenn.edu/
```
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Journal on Multimodal User Interfaces

Journal on Multimodal User Interfaces

The development of Multimodal User Interfaces relies on systemic research involving signal processing, pattern analysis, machine intelligence and human computer interaction. This journal is a response to the need of common forums grouping these research communities. Topics of interest include, but are not restricted to:
- Fusion & Fission,
- Plasticity of Multimodal interfaces,
- Medical applications,
- Edutainment applications,
- New modalities and modalities conversion,
- Usability,
- Multimodality for biometry and security,
- Multimodal conversational systems.
The journal is open to three types of contributions:
- Articles: containing original contributions accessible to the whole research community of Multimodal Interfaces. Contributions containing verifiable results and/or open-source demonstrators are strongly encouraged.
- Tutorials: disseminating established results across disciplines related to multimodal user interfaces.
- Letters: presenting practical achievements / prototypes and new technology components.
JMUI is a Springer-Verlag publication from 2008.

The submission procedure and the publication schedule are described at:

www.jmui.org

The page of the journal at springer is:

http://www.springer.com/east/home?SGWID=5-102-70-173760003-0&changeHeader=true

More information:

Imre V�radi (varadi@tele.ucl.ac.be)
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Job openings

We invite all laboratories and industrial companies which have job offers to send them to the ISCApad editor: they will appear in the newsletter and on our website for free. (also have a look at http://www.isca-speech.org/jobs.html as well as http://www.elsnet.org/ Jobs)
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Speech Engineer/Senior Speech Engineer at Microsoft, Mountain View, CA,USA

Job Type: Full-Time
Send resume to Bruce Buntschuh
Responsibilities:
Tellme, now a subsidiary of Microsoft, is a company that is focused on delivering the highest quality voice recognition based applications while providing the highest possible automation to its clients. Central to this focus is the speech recognition accuracy and performance that is used by the applications. The candidate will be responsible for the development, performance analysis, and optimization of grammars, as well as overall speech recognition accuracy, in a wide variety of real world applications in all major market segments. This is a unique opportunity to apply and extend state of the art speech recognition technologies to emerging spaces such as information search on mobile devices.
Requirements:
� Strong background in engineering, linguistics, mathematics, machine learning, and or computer science.
� In depth knowledge and expertise in the field of speech recognition.
� Strong analytical skills with a determination to fully understand and solve complex problems.
� Excellent spoken and written communication skills.
� Fluency in English (Spanish a plus).
� Programming capability with scripting tools such as Perl.
Education:
MS, PhD, or equivalent technical experience in an area such as engineering, linguistics, mathematics, or computer science.
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Speech Technology and Software Development Engineer at Microsoft Redmond WA, USA



Speech Technology and Software Development Engineer

Speech Technologies and Modeling

Speech Component Group

Microsoft Corporation

Redmond WA, USA

Please contact: Yifan.Gong@microsoft.com

Microsoft's Speech Component Group has been working on automatic speech recognition (SR) in real environments. We develop SR products for multiple languages for mobile devices, desktop computers, and communication servers. The group now has an open position for speech scientists with a software development focus to work on our acoustic and language modeling technologies. The position offers great opportunities for innovation and technology and product development.

Responsibilities:

�     Design and implement speech/language modeling and recognition algorithms to improve recognition accuracy.
�     Create, optimize and deliver quality speech recognition models and other components tailored to our customers' needs.
�     Identify, investigate and solve challenging problems in the areas of recognition accuracy from speech recognition system deployments.
�     Improve speech recognition language expansion engineering process that ensures product quality and scalability.

Required competencies and skills:

�     Passion about speech technology and quality software, demonstrated ability relative to the design and implementation of speech recognition algorithms.
�     Strong desire for achieving excellent results, strong problem solving skills, ability to multi-task, handle ambiguities, and identify issues in complex SR systems.
�     Good software development skills, including strong aptitude for software design and coding. 3+ years of experience in C/C++ and programming with scripting languages are highly desirable.
�     MS or PhD degree in Computer Science, Electrical Engineering, Mathematics, or related disciplines, with strong background in speech recognition technology, statistical modeling, or signal processing.
�     Track record of developing SR algorithms, or experience in linguistic/phonetics, is a plus.

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PhD Research Studentship in Spoken Dialogue Systems- Cambridge UK

Applications are invited for an EPSRC sponsored studentship in Spoken Dialogue Systems leading to the PhD degree. The student will join a team lead by Professor Steve Young working on statistical approaches to building Spoken Dialogue Systems. The overall goal of the team is to develop complete working end-to-end systems which can be trained from real data and which can be continually adapted on-line. The PhD work will focus specifically on the use of Partially Observable Markov Decision Processes for dialogue modelling and techniques for learning and adaptation within that framework. The work will involve statistical modelling, algorithm design and user evaluation. The successful candidate will have a good first degree in a relevant area. Good programming skills in C/C++ are essential and familiarity with Matlab would be useful.
The studentship will be for 3 years starting in October 2007 or January 2008. The studentship covers University and College fees at the Home/EU rate and a maintenance allowance of 13000 pounds per annum. Potential applicants should email Steve Young with a brief CV and statement of interest in the proposed work area
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Elektrobit seeks SW-Engineers (m/f) for multimodal HMI Solutions (Speech Dialog)

Elektrobit Automotive Software is located in Erlangen, Germany and delivers ready-to-mass product implementations of a variety of software standards of the automotive industry and services to implement large software projects. The spectrum is enhanced with tools for HMI and control device development and in-house development, such as a navigation solution. We are developing solutions for multimodal HMIs in automotive infotainment/navigation systems. One focus are speech dialog systems. The challenge lies in realizing natural speech dialogue systems for different applications (navigation, mp3 player etc.) in an embedded environment. You will be designing and developing such speech solutions.

You have know how in one or more of the following areas:
Experience project co-ordination
Programming C/C++, perl, (Java) for windows and/or Linux
Speech recognition
Multimodal dialog systems
Speech synthesis /TTS
SW- Processes and SW-Tests
Experience in Object oriented Programming
Experience with Embedded Operating Systems

Your job description
Project coordination
Coordination of supplier and requirements from different applications
Development and specification of concepts for speech-related SW modules for different applications in embedded environments
Implementation of multimodal HMIs
Integration of speech modules in HMIs
Testing
We expect from you:
A degree in IT, electrical/electronic engineering, computational linguistics or similar
Good working knowledge of German and English
Innovative streak
Willingness to take responsibility in international Teams
We offer you:
A motivating working environment
Challenging work
Support in advancement

Please apply at www.elektrobit.com -> Automotive Software -> jobs
If you have any further questions Mr. Schr�r (Tel.-Nr. +49 (9131) 7701-516) or Mr. Huck (-217) will gladly answer them.
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Sound to Sense: 18 Fellowships in speech research

Sound to Sense (S2S) is a Marie Curie Research Training Network involving collaborative speech research amongst 13 universities in 10 countries. 18 Training Fellowships are available, of which 12 are predoctoral and 6 postdoctoral (or equivalent experience). Most but not all are planned to start in September or October 2007.
A research training network's primary aim is to support and train young researchers in professional and inter-disciplinary scientific skills that will equip them for careers in research. S2S's scientific focus is on cross-disciplinary methods for modelling speech recognition by humans and machines. Distinctive aspects of our approach include emphasis on richly-informed phonetic models that emphasize communicative function of utterances, multilingual databases, multiple time domain analyses, hybrid episodic-abstract computational models, and applications and testing in adverse listening conditions and foreign language learning.
Eleven projects are planned. Each can be flexibly tailored to match the Fellows' backgrounds, research interests, and professional development needs, and will fall into one of four broad themes.
1: Multilinguistic and comparative research on Fine Phonetic Detail (4 projects)
2: Imperfect knowledge/imperfect signal (2 projects)
3: Beyond short units of speech (2 projects)
4: Exemplars and abstraction (3 projects)
The institutions and senior scientists involved with S2S are as follows:
* University of Cambridge, UK (S. Hawkins (Coordinator), M. Ford, M. Miozzo, D. Norris. B. Post)
* Katholieke Universiteit, Leuven, Belgium (D. Van Compernolle, H. Van Hamme, K. Demuynck)
* Charles University, Prague, Czech Republic (Z. Palkov�, T. Dub?da, J. Vol�n)
* University of Provence, Aix-en-Provence, France (N. Nguyen, M. d'Imperio, C. Meunier)
* University Federico II, Naples, Italy (F. Cutugno, A. Corazza)
* Radboud University, Nijmegen, The Netherlands (L. ten Bosch, H. Baayen, M. Ernestus, C. Gussenhoven, H. Strik)
* Norwegian University of Science and Technology (NTNU), Trondheim, Norway (W. van Dommelen, M. Johnsen, J. Koreman, T. Svendsen)
* Technical University of Cluj-Napoca, Romania (M. Giurgiu)
* University of the Basque Country, Vitoria, Spain (M-L. Garcia Lecumberri, J. Cenoz)
* University of Geneva, Switzerland (U. Frauenfelder)
* University of Bristol, UK (S. Mattys, J. Bowers)
* University of Sheffield, UK (M. Cooke, J. Barker, G. Brown, S. Howard, R. Moore, B. Wells)
* University of York, UK. (R. Ogden, G. Gaskell, J. Local)
Successful applicants will normally have a degree in psychology, computer science, engineering, linguistics, phonetics, or related disciplines, and want to acquire expertise in one or more of the others.
Positions are open until filled, although applications before 1 May 2007 are recommended for starting in October 2007.
Further details are available from the web about:
+ the research network (92kB) and how to apply, + the research projects(328 kB).
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AT&T - Labs Research: Research Staff Positions - Florham Park, NJ

AT&T - Labs Research is seeking exceptional candidates for Research Staff positions. AT&T is the premiere broadband, IP, entertainment, and wireless communications company in the U.S. and one of the largest in the world. Our researchers are dedicated to solving real problems in speech and language processing, and are involved in inventing, creating and deploying innovative services. We also explore fundamental research problems in these areas. Outstanding Ph.D.-level candidates at all levels of experience are encouraged to apply. Candidates must demonstrate excellence in research, a collaborative spirit and strong communication and software skills. Areas of particular interest are
- Large-vocabulary automatic speech recognition
- Acoustic and language modeling
- Robust speech recognition
- Signal processing
- Speaker recognition
- Speech data mining
- Natural language understanding and dialog
- Text and web mining
- Voice and multimodal search
AT&T Companies are Equal Opportunity Employers. All qualified candidates will receive full and fair consideration for employment. More information and application instructions are available on our website at http://www.research.att.com/. Click on "Join us". For more information, contact Mazin Gilbert (mazin at research dot att dot com).
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Research Position in Speech Processing at UGent, Belgium

Background
Since March 2005, the universities of Leuven, Gent, Antwerp and Brussels have joined forces in a big research project, called SPACE (SPeech Algorithms for Clinical and Educational applications). The project aims at contributing to the broader application of speech technology in educational and therapeutic software tools. More specifically, it pursues the automatic detection and classification of reading errors in the context of an automatic reading tutor, and the objective assessment of disordered speech (e.g. speech of the deaf, dysarthric speech, ...) in the context of computer assisted speech therapy assessment. Specific for the target applications is that the speech is either grammatically and lexically incorrect or a-typically pronounced. Therefore, standard technology cannot be applied as such in these applications.

Job description

The person we are looking for will be in charge of the data-driven development of word mispronunciation models that can predict expected reading errors in the context of a reading tutor. These models must be integrated in the linguistic model of the prompted utterance, and achieve that the speech recognizer becomes more specific in its detection and classification of presumed errors than a recognizer which is using a more traditional linguistic model with context-independent garbage and deletion arcs. A challenge is also to make the mispronunciation model adaptive to the progress made by the user.

Profile

We are looking for a person from the EU with a creative mind, and with an interest in speech & language processing and machine learning. The work will require an ability to program algorithms in C and Python. Having experience with Python is not a prerequisite (someone with some software experience is expected to learn this in a short time span). Demonstrated experience with speech & language processing and/or machine learning techniques will give you an advantage over other candidates.

The job is open to a pre-doctoral as well as a post-doctoral researcher who can start in November or December. The job runs until February 28, 2009, but a pre-doctoral candidate aiming for a doctoral degree will get opportunities to do follow-up research in related projects.

Interested persons should send their CV to Jean-Pierre Martens (martens@elis.ugent.be). There is no real deadline, but as soon as a suitable person is found, he/she will get the job.
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Summer Inter positions at Motorola Schaumburg Illinois USA

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
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Post-doc at France Telcom R&D Lannion Brittany France

Post-Doc � France T�l�com R&d, Lannion : acquisition de contexte � partir de prise de son ambiante.

DeadLine: 31/12/2007

claude.marro@orange-ftgroup.com

Description du contexte

Des donn�es physiques de toute nature, provenant de l'environnement de l'utilisateur, peuvent �tre utilis�es dans la communication ambiante comme informations de contexte pour offrir des fonctionnalit�s de service ou d'interface nouvelles, en particulier au niveau de l'adaptation du service � la situation et l'activit� des utilisateurs. Ces donn�es sont acquises par divers capteurs r�partis dans l'environnement. Les donn�es de sc�ne audio issues de microphones sont parmi les plus riches que l'on puisse exploiter parmi toutes ces donn�es de capteurs, et elles pr�sentent surtout la particularit�, dans les applications de communication ambiante, d'�tre utilisables � la fois comme inputs fonctionnels (communication audio interpersonnelle) et comme inputs de contexte, pour lequel elles peuvent �tre combin�es avec des donn�es issues d'autres types de capteurs. L'objectif ici est de d�velopper des dispositifs permettant d'aboutir � cette double utilisation des donn�es audio. Le syst�me de prise de son doit dispara�tre de l'attention des utilisateurs et cette d�mat�rialisation en fait la principale difficult�. En effet, l'�loignement de la prise de son entra�ne une d�gradation de la parole utile et n�cessite une localisation de locuteur et une "focalisation" dans sa direction.

Acquisition et restitution audio fonctionnelle

Id�alement, l'objectif tr�s ambitieux pourrait �tre d'obtenir une acquisition et restitution de son qui puisse �tre efficace quel que soit l'endroit de la pi�ce o� se trouve une personne, en utilisant des microphones r�partis dans l'environnement. En raison des difficult�s �voqu�es ci-dessus, remporter ce challenge est hors de port�e de cette �tude, on proposera comme alternative un dispositif permettant une prise et restitution du son en un nombre limit� de points pr�cis�s � l'avance.

Deux approches multi-capteurs sont envisag�es : l'antenne acoustique � directivit� contr�l�e et le microphone ambisonique. L'int�r�t d'aborder ces deux techniques r�side dans leur compl�mentarit�. En particulier, la premi�re permet un design souple du diagramme de directivit� (en fonction de la g�om�trie, de la fr�quence, etc..) et est performante en moyenne et basse fr�quence (au d�triment de l'encombrement). Quant au microphone ambisonique, il a une taille r�duite (au d�triment des performances en moyenne et basse fr�quence) et permet de reproduire � l'identique un champ acoustique � distance. La premi�re phase de l'�tude permettra de d�terminer laquelle des approches est la plus adapt�e.

Acquisition de contexte sonore

La premi�re fonctionnalit� � �tudier est la localisation de sources sonores, fonction incontournable pour identifier la source vers lequel le syst�me doit pointer. La d�tection de pr�sence et la position du locuteur sont les informations contextuelles de base � extraire.

Si l'on consid�re que la localisation coupl�e � la prise de son multi-capteur constitue un outil d'analyse du champ sonore, il sera possible d'apporter d'autres informations de contexte. En effet, le d�veloppement de traitements sp�cifiques permettra par exemple de donner le nombre de locuteurs et leurs positions dans la pi�ce, leur pourcentage de locution, le niveau de bruit de la pi�ce, etc.

Une analyse plus fine du contexte sonore est � envisager comme une perspective de ce travail et ne sera abord� que si le temps le permet. Ceci concerne les informations � extraire qui n�cessiteraient l'usage de technologies telles que la reconnaissance vocale, la classification et l'indexation audio.

Notons que le syst�me de prise de son et les traitements d�velopp�s dans ce projet constitueront des pr�-requis n�cessaires pour la continuit� des travaux sur l'analyse fine du contexte sonore.

Profil

Aspects pratiques
- Aucune condition de nationalit� n'est requise
- Le chercheur b�n�ficiera d'un contrat � dur�e d�termin�e de France T�l�com, pour une dur�e de 12 � 18 mois, non renouvelable ni prolongeable.
- Le chercheur sera int�gr� � la division R&D de France T�l�com sur son site de Lannion, CRD Technologies, Laboratoire � Speech and Sound technologies and Processing�.
Comp�tences Techniques
- Traitement num�rique du signal (analyse spectrale, filtrage, etc.)
- Traitements multi-microphones
- Si possible bases en traitement de la parole et en acoustique
- Go�t pour les travaux de recherche applicative (analyse, mise au point et adaptation)
- Langages Matlab et C
Aptitudes

- Go�t pour le travail en �quipe.

- Bon niveau en anglais.

Niveau poste : ing�nieur en CDD de type post-doctorat - Dur�e : 12 � 18 mois.
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Ph.D. Program CMU-PORTUGAL

Ph.D. Program CMU-PORTUGAL in the area of Language and Information Technologies

The Language Technologies Institute (LTI) of the School of Computer Science at Carnegie Mellon University (CMU) offers a dual degree Ph.D.

Program in Language and Information Technologies in cooperation with Portuguese Universities. This Ph.D. program is part of the activities of the recently created Information and Communication Technologies Institute (ICTI), resulting from the Portugal-CMU Partnership.

The Language Technologies Institute, a world leader in the areas of speech processing, language processing, information retrieval, machine translation, machine learning, and bio-informatics, has been formed 20 years ago. The breadth of language technologies expertise at LTI enablesnew research in combinations of the core subjects, for example, inspeech-to-speech translation, spoken dialog systems, language-based tutoring systems, and question/answering systems.

The Portuguese consortium of Universities includes the Spoken LanguageSystems Lab (L2F) of INESC-ID Lisbon/IST, the Center of Linguistics of the University of Lisbon (CLUL/FLUL), the Centre for Human Language Technology and Bioinformatics at the University of Beira Interior(HULTIG/UBI) and the linguistics group at the University of Algarve (UALG). These four research centers (and the corresponding Universities), share expertise in the same language technologies as LTI, although with a strong focus on processing the Portuguese language.

Each Ph.D. student will receive a dual degree from LTI and the selected Portuguese University, being co-supervised by one advisor from each institute, and spending approximately half of the 5-year doctoral program at each institute. Most of the academic part will take place at LTI, during the first 2 years, where most of the required 8 courses will be taken, with a proper balance of focus areas (Linguistic, Computer Science, Statistical/Learning, Task Orientation). The remaining 3 years of the doctoral program will be dedicated to research, mostly spent at the Portuguese institute, with one or two visits to CMU per year.

The thesis topic will be in one of the research areas of the cooperation program, defined by the two advisors. Two multilingual topics have been identified as priority research areas: computer aided language learning (CALL) and speech-to-speech machine translation (S2SMT).

The doctoral students will be involved in one of these two projects aimed at building real HLT systems. These projects will involve at least two languages, one of them being Portuguese, the target language for the CALL system to be developed and either the source or target language (or both) for the S2SMT system. These two projects provide a focus for the proposed research; through them the collaboration will explore the maincore areas in language technology.

The scholarship will be funded by the Foundation for Science and Technology (FCT), Portugal.

How to Apply

The application deadline for all Ph.D. programs in the scope of the CMU-Portugal partnership is December 15, 2007.

Students interested in the dual doctoral program must apply by filling the corresponding form at the CMU webpage http://www.lti.cs.cmu.edu/About/how-to-apply.html

The application form will be forwarded to the Portuguese University and to the Foundation for Science and Technology. Simultaneously, they should send an email to the coordinators of the Portuguese consortium and of the LTI admissions (Isabel Trancoso/Lori Levin):

Isabel.Trancoso@inesc-id.pt

lsl@cs.cmu.edu

All questions about the joint degree doctoral program should be directed to these two addresses.

The applications will be screened by a joint committee formed by representatives of LTI and representatives of the Portuguese Universities involved in the joint degree program. The candidates should indicate their scores in GRE and TOEFL tests.

Letters of recommendation are due by January 3rd.

Despite this particular focus on the Portuguese language, applications are not restricted to native or non-native speakers of Portuguese.
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Vitaver-Recruiting agency: Sr Manager Audio algorithm development

Position Code, Title and Location: 1515 - 0AK0 - Sr Manager Audio Algorithm Development -

Irvine, CA (Orange County)

Start Date: ASAP Remote or Onsite:  On location. No telecommuting or remote work.

Travel required: Occasionally. Additional Information: Below is all the information I have

from the Client.  Once I setup your interview, you will have the chance to ask them

directly anything I do not include here.

Description: Our Client is currently seeking a talented Engineering Manager to

manage a group of engineers developing and productizing audio algorithms

for Bluetooth audio devices (BT headsets, PNDs, car kits, etc)

- Work closely with marketing to define the product roadmaps, priorities, and schedules

- Work closely with tier 1 customers to describe and promote specific audio algorithms

- Work closely with application engineering and customers groups to ensure successful

adoption of designed algorithm

Requirements:  - Masters/BSc in engineering with a minimum of 10 years of industrial

experience in speech and audio processing

- Experience with implementation of speech enhancement devices (examples: VoIP,

speech compression, echo cancellation, noise suppression, beamforming, blind source

separation) on signal processing devices

- Experience working closely with marketing to define product directions and priorities

- Experience with the definition of requirements on telephony devices (for example,

BT headsets, cellular handsets, or cordless telephones)

- Some experience with optimization/evaluation of terminal (headset/ handset/ handsfree)

acoustics

 - Experience with programming on RISC CPUs and/or DSPs

- Experience with managing groups of engineers

- Strong analytical and problem solving skills

- Experience with working closely with customers in both a customer support and

 technical marketing role

 Please tell me if you feel confident with the requirements and comfortable delivering

on them.  As soon as you send me your reply to all questions below, and your resume

attached, we will talk on the phone. A technical interview with the Client will follow

as the last step.

1.	Are you a US citizen/Green Card/H visa holder (please, specify)?

2.	When will you become available?

3.	What is the yearly salary you expect? $ _______ per year on

W2 (Fulltime Employee)

4.	Very important: this is designed to save us both time and expedite a decision

by my Client.

Please complete the following Skill Matrix, answering each question with

A) number of years of experience with the skill,

B) your skill level on a scale from 1 to 5 (highest),

C) the last time you applied it. It has to be consistent with your resume.   I.e.:

Experience 4 years - Skill level 5 - Last Applied January 07

1)	Industrial experience in speech and audio processing (min 10 years): __ years;

Skill level: __; Last applied: __.

2)	Experience implementing speech enhancement devices on signal processing

devices: __ years; Skill level: __; Last applied: __.

3)	Experience with optimization/evaluation of terminal

(headset/ handset/ handsfree) acoustics: __ years; Skill level: __; Last applied: __.

4)	Programming on RISC CPUs and/or DSPs: __ years; Skill level: __;

Last applied: __.

5)	Do you have Bachelor's degree?

*   Please attach your resume as a Word document

 Alice Kondrat 
Senior Staff Recruiter Vitaver & Associates, Inc. 2385 Executive Center Drive 
#100 Boca Raton, FL 33431 
Alice@vitaver.com 
Direct Line: (561) 283-1136 Voice Line: (954) 840-3603 Fax: (866) 259 3777 
http://www.vitaver.com/

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Nuance: Software engineer speech dialog tools

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.

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Nuance: Speech scientist London UK

Nuance is the leading provider of speech and imaging solutions for businesses and consumers around the world. Every day, millions of users and thousands of businesses experience Nuance by calling directory assistance, requesting account information, dictating patient records, telling a navigation system their destination, 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
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Nuance: Research engineer speech engine

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

RESEARCH ENGINEER SPEECH ENGINE

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, improving and maintaining core ASR engine algorithms 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 developing, improving and maintaining core ASR engine algorithms for cutting edge speech and natural language understanding technologies for automotive and mobile devices.

- You will work on the design and development of more efficient, flexible ASR search algorithms with high focus on low memory and processor requirements.

QUALIFICATIONS:

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

- A background in (computational) linguistics, speech processing, ASR search, confidence values, grammars, statistics and machine learning, especially as related to natural language processing.

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

The following skills are a plus:

- You have experience with Python or other scripting languages.

- Broad knowledge about architectures of embedded platforms and processors.

- Understanding of databases

- 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.

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Nuance RESEARCH ENGINEER SPEECH DIALOG SYSTEMS:

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

    RESEARCH ENGINEER SPEECH DIALOGUE SYSTEMS

As part of our team, you will be creating speech technologies for embedded applications varying from simple command and control tasks up to natural language speech dialogues on mobile and automotive platforms.

OVERVIEW:

-You will work in Nuance's Embedded ASR research and production team, creating technology, tools and runtime software to enable our customers develop embedded speech applications. In our team of speech and language experts, you will work on natural language dialogue systems that define the state of the art.

- You will work at Nuance's International Headquarters in Merelbeke, a small town just 5km away from the heart of the picturesque city of Ghent, in the Flanders region of Belgium. Ghent has one of the most spectacular historic town centers of Europe and is known for its unique vibrant yet cozy charm, and is home to a large international university.

- 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 cutting edge natural language dialogue and speech recognition technologies for automotive embedded systems and mobile devices.

- You will design, implement, evaluate, optimize, and test new algorithms and tools for our speech recognition systems, both for research prototypes and deployed products, including all aspects of dialogue systems design, such as architecture, natural language understanding, dialogue modeling, statistical framework, and so forth.

- You will help the engine process multi-lingual natural and spontaneous speech in various noise conditions, given the challenging memory and processing power constraints of the embedded world.

QUALIFICATIONS:

- You have a university degree in computer science, (computational) linguistics, engineering, mathematics, physics, or a related field. A graduate degree is an asset.

-You have strong software and programming skills, especially in C/C++, ideally for embedded applications. Knowledge of Python or other scripting languages is a plus. [HQ1]

- You have experience in one or more of the following fields:

     dialogue systems

     applied (computational) linguistics

     natural language understanding

     language generation

     search engines

     speech recognition

     grammars and parsing techniques.

     statistics and machine learning techniques

     XML processing

-You are a team player, willing to take initiative and assume responsibility for your tasks, and are goal-oriented.

-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 strong asset.

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

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.

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CNRS tenure research position at LIMSI

CNRS tenure research position (CR2) at LIMSI

http://www.limsi.fr/tlp/postes07.html

[Application deadline 09-JAN-08 23:59 Paris time]

CR2 position 48/03 in section 48 (Communication sciences)

Information systems, multimedia and multilingual documents, knowledge

industry.

Job description

The candidate will conduct research in the field of information

processing with application to multimedia and multilingual

documents. The main research direction will be the development of

models and algorithms for automatic structuring, indexing, enrichment

and extraction of knowledge from speech data in a multilingual

context. It is envisioned that the models studied will be based on the

joint use of linguistic knowledge and statistical learning methods

applied to very large corpora. The candidate should have an expertise

in at least one of following areas: spoken language processing,

statistical based machine translation, and audio data mining. This

research will contribute to the development of emerging technologies

for knowledge engineering and information access. A variety of new

applications can be foreseen related to audiovisual archives, digital

libraries, the production of multimedia documents, and more generally

systems to organize and access multimedia and multilingual content,

especially on the Internet.

Descriptif du poste

Le titulaire du poste m�nera ses recherches dans le domaine du

traitement automatique de l'information avec pour champ applicatif les

documents multimedia et multilingues. La principale direction de

recherche envisag�e est le d�veloppement de mod�les et d'algorithmes

pour la structuration automatique, l'indexation, l'enrichissement et

l'extraction de connaissances � partir de la parole, dans un contexte

multilingue. Il est envisag� que les mod�les �tudi�s reposeront sur

l'utilisation conjointe de connaissances linguistiques et de m�thodes

d'apprentissage faisant usage de grandes masses de donn�es. Le

candidat aura une expertise au moins sur l'une des probl�matiques

suivantes: le traitement du langage parl�, la traduction par

apprentissage statistique � base de donn�es, et la fouille de donn�es

audio. Ces recherches contribueront au d�veloppement de syst�mes

d'information et de nouveaux usages. Ces usages concernent les

archives audiovisuelles, les bibliotheques num�riques, la production

de documents multimedia, et plus g�n�ralement les syst�mes de gestion

et d'acc�s aux contenus multim�dias et multilingues, en particulier

sur l'internet.
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Research Position in Speech Processing at Nagoya Institute of

Research Position in Speech Processing at Nagoya Institute of

Technology, Japan

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.
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PhD's fellowships at NTNU Trondheim-Norway

Two Ph.D. fellowships available at

Department of Electronics and Telecommunications

The Norwegian University of Science and Technology ( NTNU)

Trondheim, Norway

Closing date: January 27, 2008

The Ph.D. scholarships are part of the projects SIRKUS (Spoken Information Retrieval by Knowledge Utilization in Statistical Speech Processing) and SMUDI (Voice Control in Multi-Modal Dialogue).

PhD Fellowship in the SIRKUS project:
The vision of SIRKUS is to develop new architectures for speech recognition that can achieve human-like performance. There is evidence indicating that system performace based on the current paradigm is reaching an upper bound. We believe that in order to achieve our vision, knowledge about speech production and perception as well as a better understanding of speech per se needs to be incorporated into a statistical framework. This implies that new approaches to speech analysis need to be investigated and developed, and that a new statistical framework for modelling speech based on a set of analysis results needs to be defined.

The PhD project
An on-going PhD project will investigate and develop speech analysis methods that will supplement and augment the current methods that are based on spectral analysis. In particular, development of detectors of phonologically significant events, i.e. speech attributes, will be central. A set of speech analyses, including speech attribute detectors, will produce speech event sequences, which can constitute temporally asynchronous observation streams and may be correlated both in time and across observation streams.
This PhD project will in cooperation with the on-going project develop novel statistical modelling approaches suitable for modelling speech based on event sequences. Integration of the observation streams from the speech analyses in a statistical description of various linguistic units such as phonemes, syllables, words and sentences will be central.
The project will incorporate collaboration with several foreign partners. The scholarships will also include an international visiting research scholarship.

Qualifications:
We seek highly motivated individuals holding a masters degree in electronics engineering, signal processing, statistics, or other relevant disciplines. Experience in speech technology is desirable, but not an absolute requirement.

PhD Fellowship in the SMUDI project:
Large groups of disabled persons have great difficulties accessing information that is available on the internet. Many government and municipal agencies are in the process of changing their preferred interaction with the public, moving to internet based systems for submission of applications and information requests. In many cases it will be necessary for the user to provide information by filling in forms. Some examples are online shopping for goods and services, e.g. air travel and use of public services such as filling in tax return forms.

The PhD project
The goal of the PhD project is to develop a speech based system for filling in internet forms. The work will include methods for interpretation of the internet forms, speech recognition to transform the user speech to text, and integration of speech tehnology with other modalities for information presentation and user input.

In order to achieve a best possible performance for this task, the speech recognition ought to be at the user's computer. This will have the advantage that the speech recognizer can be adapted to the user's voice and pronunciation, and in addition provide a situation where the speech signal is not band limited or noise corrupted by the transmission channel. The challenge is that the recognizer cannot be tailored for filling in one particular form, and thus it will be necessary to develop a general large vocabulary speech recognizer (a "dictation engine") for Norwegian. The system should be able to define vocabulary and syntax dynamically by interpreting the content of the internet form.

Qualifications:
We seek highly motivated individuals holding a masters degree in electronics engineering, signal processing, statistics, natural language processing or other relevant disciplines. Experience in speech technology is desirable, but not an absolute requirement.

Information for both fellowships:
The PhD fellows will be associated with the Signal Processing Group at NTNU and will work in a strong and active scientific environment.

Award holders at NTNU are normally appointed for up to 4 years with 25% of the time spent on specified work. This work is primarily linked to teaching and is usually divided so that a relatively large part is done in the first half of the period of the appointment.
The appointments is at code 1017, salary level 43-58 in the national salary scheme, gross NOK. 325.600 - 423.800 per annum (1 NOK ~ 0.125 EUR), and are normally remunerated at wage level 43 of which 2% is deducted for the State Pension scheme. The salary might be adjusted after negotiation with the employer to reflect the applicant's experience.

Further information:
For more information on the SIRKUS position and the application requirements, please see http://www.iet.ntnu.no/projects/SIRKUS/Positions.html
For more information on the SMUDI position and the application requirements, please see http://www.iet.ntnu.no/projects/SMUDI/Positions.html

Interested candidates are encouraged to contact Professor Torbj�rn Svendsen (phone: +47-735-92674, email: torbjorn@iet.ntnu.no) for further information.
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Research position in Speech Recognition in the context of Spoken Document Retrieval at the University of Twente (NL)

Research position in Speech Recognition in the context of Spoken Document

Retrieval

The Human Media Interaction (HMI) group of the Computer Science department

at the University of Twente in Enschede, The Netherlands, has funding for a

(junior) research position in the area of speech recognition; the

application domain is multimedia information retrieval. We are looking for a

candidate interested in research and development work in speech recognition

for surprise data, data with unknown or highly fluctuating

characteristics, both at the acoustic level and word level, as for example

occurring in historical archives or commercials. Initially the position will be available for one year with an opportunity for continuation.

Requirements

- affinity with speech recognition technology demonstrated by an academic

degree in e.g., computational linguistics, computer science, phonetics

- provable knowledge/experience in at least two of the following areas:

speech recognition, NLP, DSP, machine-learning

- good scripting skills (Perl and alike); skills in C/C++ is regarded as a pro

- comfortable with Linux environment

- willing and comfortable to work in a team

Conditions of employment

Depending on demonstrable experience, gross salary starts with EUR 2.217,-

per month in the first year ((level 10, CAO Nederlandse Universiteiten)

Applications (in English or Dutch) should be sent to:

Prof. Dr. Franciska de Jong

HMI, Faculty of EEMCS

University of Twente,

P.O. Box 217

7500 AE Enschede

The Netherlands

fdejong@ewi.utwente.nl

Make sure to include the following:

- a letter of application with a motivation

- a curriculum vitae (including a list of publications and previous project=

s

worked on)

- the names and email addresses of two referents

Deadline: Februari 1, 2008.

For more information on the HMI research group, and its projects see

http://hmi.ewi.utwente.nl or contact

Prof. Dr. Franciska de Jong (fdejong@ewi.utwente.nl) or

Dr. Roeland Ordelman (ordelman@ewi.utwente.nl)
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C/C++ Programmer Munich, Germany

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 Stra�e 32

D-80337 M�nchen Germany

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Speech and Natural Language Processing Engineer at M*Modal, Pittsburgh.PA,USA

Speech and Natural Language Processing Engineer

M*Modal is a fast-moving speech technology company based in Pittsburgh, PA. Our portfolio of conversational speech recognition and natural language understanding technologies is widely recognized as the most advanced in the industry. We are a leading innovator in the field of conversational documentation services (CDS) - where speech recognition and natural language understanding are combined in a unique setup targeted to truly understand conversational speech and turn it directly into actionable and meaningful data. Our proprietary speech understanding technology - operating on M*Modal's computing grid hosted in our national data center - is already redefining the way clinical information is captured in healthcare.

We are seeking an experienced and dedicated speech and natural language processing engineer who wants to push the frontiers of conversational speech understanding. Join our renowned research and development team, and add to our unique blend of scientific and engineering excellence.

Responsibilities:
- You will be working with other members of the R&D team to continuously improve our speech and natural language understanding technologies.
- You will participate in designing and implementing algorithms, tools and methodologies in the area of automatic speech recognition and natural language processing/understanding.
- You will collaborate with other members of the R&D team to identify, analyze and resolve technical issues.
Requirements:
- Solid background in speech recognition, natural language processing, machine learning and information extraction.
- 2+ years of experience participating in software development projects
- Proficient with Java, C++ and scripting (e.g. Python, Perl, ...)
- Excellent analytical and problem-solving skills
- Integrate and communicate well in small R&D teams
- Masters degree in CS or related engineering fields
- Experience in a healthcare-related field a plus
In June 2007 M*Modal moved to a great new office space in the Squirrel Hill area of Pittsburgh. We are excited to be growing and are looking for individuals who have a passion for the work they do and are interested in becoming a member of a dynamic work group of smart passionate drivers who also know how to have fun.

M*Modal offers a top-notch benefits package that includes medical, dental and vision coverage, short-term disability, matching 401K savings plan, holidays, paid-time-off and tuition refund. If you would like to be considered for this opportunity, please send your resume and cover letter to Mary Ann Gamble at maryann.gamble@mmodal.com.

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Senior Research Scientist -- Speech and Natural Lgage Processing at M*Modal, Pittsburgh, PA,USA

Senior Research Scientist -- Speech and Natural Language Processing

M*Modal is a fast-moving speech technology company based in Pittsburgh, PA. Our portfolio of conversational speech recognition and natural language understanding technologies is widely recognized as the most advanced in the industry. We are a leading innovator in the field of conversational documentation services (CDS) - where speech recognition and natural language understanding are combined in a unique setup targeted to truly understand conversational speech and turn it directly into actionable and meaningful data. Our proprietary speech understanding technology - operating on M*Modal's computing grid hosted in our national data center - is already redefining the way clinical information is captured in healthcare.

We are seeking an experienced and dedicated senior research scientist who wants to push the frontiers of conversational speech understanding. Join our renowned research and development team, and add to our unique blend of scientific and engineering excellence.

Responsibilities:
- Plan and perform research and development tasks to continuously improve a state-of-the-art speech understanding system
- Take a leading role in identifying solutions to challenging technical problems
- Contribute original ideas and turn them into product-grade software implementations
- Collaborate with other members of the R&D team to identify, analyze and resolve technical issues
Requirements:
- Solid research & development background with 3+ years of experience in speech recognition research, covering at least two of the following topics: speech processing, acoustic modeling, language modeling, decoding, LVCSR, natural language processing/understanding, speaker verification/identification, audio mining
- Working knowledge of Machine Learning, Information Extraction and Natural Language Processing algorithms
- 3+ years of experience participating in large-scale software development projects using C++ and Java.
- Excellent analytical, problem-solving and communication skills
- PhD with focus on speech recognition or Masters degree with 3+ years industry experience working on automatic speech recognition
- Experience and/or education in medical informatics a plus
- Working experience in a healthcare related field a plus
In June 2007 M*Modal moved to a great new office space in the Squirrel Hill area of Pittsburgh. We are excited to be growing and are looking for individuals who have a passion for the work they do and are interested in becoming a member of a dynamic work group of smart passionate drivers who also know how to have fun.

M*Modal offers a top-notch benefits package that includes medical, dental and vision coverage, short-term disability, matching 401K savings plan, holidays, paid-time-off and tuition refund. If you would like to be considered for this opportunity, please send your resume and cover letter to Mary Ann Gamble at maryann.gamble@mmodal.com.

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Journals

Papers accepted for FUTURE PUBLICATION in Speech

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.

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Special Issue on Non-Linear and Non-Conventional Speech Processing-Speech Communication

Speech Communication

Call for Papers: Special Issue on Non-Linear and Non-Conventional Speech Processing

Editors: Mohamed CHETOUANI, UPMC

Marcos FAUNDEZ-ZANUY, EUPMt (UPC)

Bruno GAS, UPMC

Jean Luc ZARADER, UPMC

Amir HUSSAIN, Stirling

Kuldip PALIWAL, Griffith University

The field of speech processing has shown a very fast development in the past twenty years, thanks to both technological progress and to the convergence of research into a few mainstream approaches. However, some specificities of the speech signal are still not well addressed by the current models. New models and processing techniques need to be investigated in order to foster and/or accompany future progress, even if they do not match immediately the level of performance and understanding of the current state-of-the-art approaches.

An ISCA-ITRW Workshop on "Non-Linear Speech Processing" will be held in May 2007, the purpose of which will be to present and discuss novel ideas, works and results related to alternative techniques for speech processing departing from the mainstream approaches: http://www.congres.upmc.fr/nolisp2007

We are now soliciting journal papers not only from workshop participants but also from other researchers for a special issue of Speech Communication on "Non-Linear and Non-Conventional Speech Processing"

Submissions are invited on the following broad topic areas:

I. Non-Linear Approximation and Estimation

II. Non-Linear Oscillators and Predictors

III. Higher-Order Statistics

IV. Independent Component Analysis

V. Nearest Neighbours

VI. Neural Networks

VII. Decision Trees

VIII. Non-Parametric Models

IX. Dynamics of Non-Linear Systems

X. Fractal Methods

XI. Chaos Modelling

XII. Non-Linear Differential Equations

All fields of speech processing are targeted by the special issue, namely :

1. Speech Production

2. Speech Analysis and Modelling

3. Speech Coding

4. Speech Synthesis

5. Speech Recognition

6. Speaker Identification / Verification

7. Speech Enhancement / Separation

8. Speech Perception
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IEEE Transactions on Audio, Speech and Language Processing

Special Issue on Multimodal Processing in Speech -based Interactions in the IEEE Transactions on Audio, Speech and Language Processing has been extended to January 15, 2008. A URL to point to is: http://www.ewh.ieee.org/soc/sps/tap/sp_issue/special-issue-multimodal.pdf
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Future Conferences

Publication policy: Hereunder, you will find very short announcements of future events. The full call for participation can be accessed on the conference websites
See also our Web pages (http://www.isca-speech.org/) on conferences and workshops.
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Future Interspeech conferences

INTERSPEECH 2008-ICSLP

September 22-26, 2008, Brisbane, Queensland, Australia
Conference Website
Chairman: Denis Burnham, MARCS, University of West Sydney.

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INTERSPEECH 2009-EUROSPEECH

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

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INTERSPEECH 2010-ICSLP

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."

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Future ISCA Technical and Research Workshops

ITRW Odyssey 2008

The Speaker and Language Recognition Workshop
21-25 January 2008, Stellenbosch, South Africa
Topics
* Speaker recognition(identification, verification, segmentation, clustering)
* Text dependent and independent speaker recognition
* Multispeaker training and detection
* Speaker characterization and adaptation
* Features for speaker recognition
* Robustness in channels
* Robust classification and fusion
* Speaker recognition corporaand evaluation
* Use of extended training data
* Speaker recognition with speaker recognition
* Forensics, multimodality and multimedia speaker recogntion
* Speaker and language confidence estimation
* Language, dialect and accent recognition
* Speaker synthesis and transformation
* Biometrics
* Human recognition
* Commercial applications
Paper submission
Proaspective authors are invited to submit papers written in English via the Odyssey website. The style guide, templates,and submission form can be downloaded from the Odyssey website. Two members of the scientific committee will review each paper. Each accepted paper must have at least one registered author. The Proceedings will be published on CD
Schedule
Draft paper due July 15, 2007
Notification of acceptance September 15,2007
Final paper due October 30, 2007
Preliminary program November 30, 2007
Workshop January 21-25, 2008
Futher informations: venue, registation...
On the workshop website
Chairs
Niko Brummer, Spescom Data Voice, South Africa
Johan du Preez.Stellenbosch University,South Africa
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ISCA ITRW speech analysis and processing for knowledge discovery

June 4 - 6, 2008
Aalborg, Denmark
Workshop website
Humans are very efficient at capturing information and messages in speech, and they often perform this task effortlessly even when the signal is degraded by noise, reverberation and channel effects. In contrast, when a speech signal is processed by conventional spectral analysis methods, significant cues and useful information in speech are usually not taken proper advantage of, resulting in sub-optimal performance in many speech systems. There exists, however, a vast literature on speech production and perception mechanisms and their impacts on acoustic phonetics that could be more effectively utilized in modern speech systems. A re-examination of these knowledge sources is needed. On the other hand, recent advances in speech modelling and processing and the availability of a huge collection of multilingual speech data have provided an unprecedented opportunity for acoustic phoneticians to revise and strengthen their knowledge and develop new theories. Such a collaborative effort between science and technology is beneficial to the speech community and it is likely to lead to a paradigm shift for designing next-generation speech algorithms and systems. This, however, calls for a focussed attention to be devoted to analysis and processing techniques aiming at a more effective extraction of information and knowledge in speech.
Objectives:
The objective of this workshop is to discuss innovative approaches to the analysis of speech signals, so that it can bring out the subtle and unique characteristics of speech and speaker. This will also help in discovering speech cues useful for improving the performance of speech systems significantly. Several attempts have been made in the past to explore speech analysis methods that can bridge the gap between human and machine processing of speech. In particular, the time varying aspects of interactions between excitation and vocal tract systems during production seem to elude exploitation. Some of the explored methods include all-pole and polezero modelling methods based on temporal weighting of the prediction errors, interpreting the zeros of speech spectra, analysis of phase in the time and transform domains, nonlinear (neural network) models for information extraction and integration, etc. Such studies may also bring out some finer details of speech signals, which may have implications in determining the acoustic-phonetic cues needed for developing robust speech systems.
The Workshop:
G will present a full-morning common tutorial to give an overview of the present stage of research linked to the subject of the workshop
G will be organised as a single series of oral and poster presentations
G each oral presentation is given 30 minutes to allow for ample time for discussion
G is an ideal forum for speech scientists to discuss the perspectives that will further future research collaborations.
Potential Topic areas:
G Parametric and nonparametric models
G New all-pole and pole-zero spectral modelling
G Temporal modelling
G Non-spectral processing (group delay etc)
G Integration of spectral and temporal processing
G Biologically-inspired speech analysis and processing
G Interactions between excitation and vocal tract systems
G Characterization and representation of acoustic phonetic attributes
G Attributed-based speaker and spoken language characterization
G Analysis and processing for detecting acoustic phonetic attributes
G Language independent aspects of acoustic phonetic attributes detection
G Detection of language-specific acoustic phonetic attributes
G Acoustic to linguistic and acoustic phonetic mapping
G Mapping from acoustic signal to articulator configurations
G Merging of synchronous and asynchronous information
G Other related topics
Call for papers. Notification of review:
The submission deadline is January 31, 2008.
Registration
Fees for early and late registration for ISCA and non-ISCA members will be made available on the website during September 2007.
Venue:
The workshop will take place at Aalborg University, Department of Electronic Systems, Denmark. See the workshop website for further and latest information.
Accommodation:
There are a large number of hotels in Aalborg most of them close to the city centre. The list of hotels, their web sites and telephone numbers are given on the workshop website. Here you will also find information about transportation between the city centre and the university campus.
How to reach Aalborg:
Aalborg Airport is half an hour away from the international Copenhagen Airport. There are many daily flight connections between Copenhagen and Aalborg. Flying with Scandinavian Airlines System (SAS) or one of the Star Alliance companies to Copenhagen enables you to include Copenhagen-Aalborg into the entire ticket, and this way reducing the full transportation cost. There is also an hourly train connection between the two cities; the train ride lasts approx. five hours
Organising Committee:
Paul Dalsgaard, B. Yegnanarayana, Chin-Hui Lee, Paavo Alku, Rolf Carlson, Torbj�rn Svendsen,
Important dates
Submission of full and final: January 31, 2008 on the Website
http://www.es.aau.dk/ITRW/
Notification of review results: No later than March 30., 2008.
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ITRW on Evidence-based Voice and Speech Rehabilitation in Head





ISCA Workshop

Evidence-based Voice and Speech Rehabilitation in Head & Neck Oncology

Amsterdam, May 15-16, 2008

Evidence-based Voice and Speech Rehabilitation is of increasing relevance in Head & Neck Oncology. The number of patients requiring treatment for cancer in the upper respiratory and vocal tract keeps rising. Moreover, treatment - whether it concerns an "organ preservation protocol" or traditional surgery and radiotherapy - negatively impacts the function of organs vital for communication. A "function preservation treatment" does, unfortunately, not yet exist. This workshop seeks to assemble the latest and most relevant knowledge on evidence-based voice and speech rehabilitation. Aside from the main topic (voice and speech rehabilitation after total laryngectomy), other areas, such as vocal issues in early-stage larynx carcinoma, and various stages of oral / oropharyngeal carcinoma will be addressed.

The workshop comprises four topical sessions (see below). Each session includes two keynote lectures plus a round-table discussion and (maximally 10) poster presentations pertinent to the session's topic. A work document, based on the keynote lectures, will form the basis for each round-table discussion. This work document will contain all presently available research evidence, discuss its (clinical) relevance and will formulate directions and areas of interest for future research. The keynote lectures, work documents and poster papers are to be compiled into Workshop Proceedings, and will be published under ISCA flag (website: http://www.isca-speech.org/). It is our aim to make these Proceedings available at the workshop. This will result in a useful and traceable �State of the Art' handbook/CD/web publication.

Prof. Dr. Frans JM Hilgers

Prof. Dr. Louis CW Pols

Dr. Maya van Rossum

Venue:

Tinbergen lecture hall, Royal Netherlands Academy of Arts and Sciences. Kloveniersburgwal 29, Amsterdam

More information can be obtained from the website www.fon.hum.uva.nl/webhnr/
or by sending a
Organization:

Prof. Dr. Frans JM Hilgers

Prof. Dr. Louis CW Pols

Dr. Maya van Rossum

Institute of Phonetic Sciences - Amsterdam Center for Language and Communication, University of Amsterdam

Department of Head and Neck Oncology and Surgery

The Netherlands Cancer Institute - Antoni van Leeuwenhoek Hospital

Department of Otolaryngology, Academic Medical Center, University of Amsterdam

International Faculty

Prof. Philip C Doyle, PhD                University of Western Ontario, London, Canada

Prof. Tanya L Eadie, PhD                University of Washington, Seattle, USA

Prof. Dr. Dr. Ulrich Eysholdt             University of Erlangen-Nuremberg, Germany

Prof. Britta Hammarberg, PhD        Karolinska University, Stockholm, Sweden

Prof. Jeffrey P Searle, PhD             University of Kansas, Kansas City, USA

Local Faculty

Dr. Annemieke H Ackerstaff ²

Dr. Corina J van As-Brooks ²

Dr. Michiel WM van den Brekel ^2,3

Prof. Dr. Frans Hilgers ^{1,2, 3}

Petra Jongmans, MA ^{1, 2}

Lisette van der Molen, MA ²

Prof. Dr. Louis CW Pols ¹

Dr. Maya van Rossum ^{2, 4}

Dr. Irma M Verdonck-de Leeuw ⁵

¹ Institute of Phonetic Sciences/Amsterdam Center of Language and Communication, University of Amsterdam

² The Netherlands Cancer Institute, Amsterdam

³ Academic Medical Center, University of Amsterdam

⁴ University Medical Center Leiden

⁵ Free University Medical Center, Amsterdam

Course secretariat: Mrs. Marion van Zuilen

The Netherlands Cancer Institute

Plesmanlaan 121 1066CX Amsterdam, The Netherlands
Teleph
Organization:

Prof. Dr. Frans JM Hilgers

Prof. Dr. Louis CW Pols

Dr. Maya van Rossum

Institute of Phonetic Sciences - Amsterdam Center for Language and Communication, University of Amsterdam

Department of Head and Neck Oncology and Surgery

The Netherlands Cancer Institute - Antoni van Leeuwenhoek Hospital

Department of Otolaryngology, Academic Medical Center, University of Amsterdam

International Faculty

Prof. Philip C Doyle, PhD                University of Western Ontario, London, Canada

Prof. Tanya L Eadie, PhD                University of Washington, Seattle, USA

Prof. Dr. Dr. Ulrich Eysholdt             University of Erlangen-Nuremberg, Germany

Prof. Britta Hammarberg, PhD        Karolinska University, Stockholm, Sweden

Prof. Jeffrey P Searle, PhD             University of Kansas, Kansas City, USA

Local Faculty

Dr. Annemieke H Ackerstaff ²

Dr. Corina J van As-Brooks ²

Dr. Michiel WM van den Brekel ^2,3

Prof. Dr. Frans Hilgers ^{1,2, 3}

Petra Jongmans, MA ^{1, 2}

Lisette van der Molen, MA ²

Prof. Dr. Louis CW Pols ¹

Dr. Maya van Rossum ^{2, 4}

Dr. Irma M Verdonck-de Leeuw ⁵

¹ Institute of Phonetic Sciences/Amsterdam Center of Language and Communication, University of Amsterdam

² The Netherlands Cancer Institute, Amsterdam

³ Academic Medical Center, University of Amsterdam

⁴ University Medical Center Leiden

⁵ Free University Medical Center, Amsterdam

Course secretariat: Mrs. Marion van Zuilen

The Netherlands Cancer Institute

Plesmanlaan 121 1066CX Amsterdam, The Netherlands

Telephone +3120-512-2550; Fax +3120-512-2554one +3120-512-2550; Fax +3120-512-2554n e-mail to f.hilgers@nki.nl or kno@nki.nl

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ISCA TR Workshop on Experimental Linguistics

August 2008, Athens, Greece
Website
Prof. Antonis Botinis
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Audio Visual Speech Processing Workshop (AVSP )

Tentative location:Queensland coast near Brisbane (most likely South Stradbroke Island)
Tentative date: 27-29 September 2008 (immediately after Interspeech 2008)
Following in the footsteps of previous AVSP workshops / conferences, AVSP workshop (ISCA Research and Tutorial Workshop) will be hold concomitantly to Interspeech2008, Brisbane, Australia, 22-26 September 2008. The aim of AVSP2008 is to bring together researchers and practitioners in areas related to auditory-visual speech processing. These include human and machine AVSP, linguistics, psychology, and computer science. One of the aims of the AVSP workshops is to foster collaborations across disciplines, as AVSP research is inherently multi-disciplinary. The workshop will include a number of tutorials / keynote addresses by internationally renowned researchers in the area of AVSP.
Organizers
Roland Goecke, Simon Lucey, Patrick Lucey
Australian National University,RSISE, Bldg. 115, Australian National University, Canberra, ACT 0200, Australia
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Robust ASR Workshop

Santiago, Chile
October-November 2008
Dr. Nestor Yoma

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Forthcoming events supported (but not organized) by ISCA

3rd International Conference on Large-scale Knowledge Resources (LKR 2008)

3-5 March, 2008, Tokyo Institute of Technology, Tokyo Japan
Website
Sponsored by: 21st Century Center of Excellence (COE) Program "Framework for Systematization and Application of Large-scale",Tokyo Institute of Technology
In the 21st century, we are now on the way to the knowledge-intensive society in which knowledge plays ever more important roles. Research interest should inevitably shift from information to knowledge, namely how to build, organize, maintain and utilize knowledge are the central issues in a wide variety of fields. The 21st Century COE program, "Framework for Systematization and Application of Large-scale Knowledge Resources (COE-LKR)" conducted by Tokyo Institute of Technology is one of the attempts to challenge these important issues. Inspired by this project, LKR2008 aims at bringing together diverse contribution in cognitive science, computer science, education and linguistics to explore design, construction, extension, maintenance, validation, and application of knowledge.
Topics of Interest to the conference includes:
Infrastructure for Large-scale Knowledge
Grid computing
Network computing
Software tools and development environments
Database and archiving systems
Mobile and ubiquitous computing
Systematization for Large-scale Knowledge
Language resources
Multi-modal resources
Classification, Clustering
Formal systems
Knowledge representation and ontology
Semantic Web
Cognitive systems
Collaborative knowledge
Applications and Evaluation of Large-scale Knowledge
Archives for science and art
Educational media
Information access
Document analysis
Multi-modal human interface
Web applications
Organizing committee General conference chair: Furui, Sadaoki (Tokyo Institute of Technology)
Program co-chairs: Ortega, Antonio (University of Southern California)
Tokunaga, Takenobu (Tokyo Institute of Technology)
Publication chair: Yonezaki, Naoki (Tokyo Institute of Technology)
Publicity chair: Yokota, Haruo (Tokyo Institute of Technology)
Local organizing chair: Shinoda, Koichi (Tokyo Institute of Technology)
Submission
Since we are aiming at an interdisciplinary conference covering wide range of topics concerning large-scale knowledge resources, authors are requested to add general introductory description in the beginning of the paper so that readers of other research area can understand the importance of the work. Note that one of the reviewers of each paper is assigned from other topic area to see if this requirement is fulfilled.
There are two categories of paper presentation: oral and poster. The category of the paper should be stated at submission. Authors are invited to submit original unpublished research papers, in English, up to 12 pages for oral presentation and 4 pages for poster presentation, strictly following the LNCS/LNAI format guidelines available at the Springer LNCS Web page. . Details of the submission procedure will be announced later on.
Reviewing
The reviewing of the papers will be blind and managed by an international Conference Program Committee consisting of Area Chairs and associated Program Committee Members. Final decisions on the technical program will be made by a meeting of the Program Co-Chairs and Area Chairs. Each submission will be reviewed by at least three program committee members, and one of the reviewers is assigned from a different topic area.
Publication
The conference proceedings will be published by Springer-Verlag in their Lecture Notes in Artificial Intelligence (LNAI), which will be available at the conference.
Important dates
Paper submission deadline: 30 August, 2007
Notification of acceptance: 10 October, 2007
Camera ready papers due: 10 November, 2007
e-mail correspondence
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Call for Papers (Preliminary version) Speech Prosody 2008

Campinas, Brazil, May 6-9, 2008
Speech Prosody 2008 will be the fourth conference of a series of international events of the Special Interest Groups on Speech Prosody (ISCA), starting by the one held in Aix-en Provence, France, in 2002. The conferences in Nara, Japan (2004), and in Dresden, Germany (2006) followed the proposal of biennial meetings, and now is the time of changing place and hemisphere by trying the challenge of offering a non-stereotypical view of Brazil. It is a great pleasure for our labs to host the fourth International Conference on Speech Prosody in Campinas, Brazil, the second major city of the State of S�o Paulo. It is worth highlighting that prosody covers a multidisciplinary area of research involving scientists from very different backgrounds and traditions, including linguistics and phonetics, conversation analysis, semantics and pragmatics, sociolinguistics, acoustics, speech synthesis and recognition, cognitive psychology, neuroscience, speech therapy, language teaching, and related fields. Information: sp2008_info@iel.unicamp.br. Web site: http://sp2008.org. We invite all participants to contribute with papers presenting original research from all areas of speech prosody, especially, but nor limited to the following.
Scientific Topics
Prosody and the Brain
Long-Term Voice Quality
Intonation and Rhythm Analysis and Modelling
Syntax, Semantics, Pragmatics and Prosody
Cross-linguistic Studies of Prosody
Prosodic variability
Prosody in Discourse
Dialogues and Spontaneous Speech
Prosody of Expressive Speech
Perception of Prosody
Prosody in Speech Synthesis
Prosody in Speech Recognition and Understanding
Prosody in Language Learning and Acquisition
Pathology of Prosody and Aids for the Impaired
Prosody Annotation in Speech Corpora
Others (please, specify)
Organising institutions
Speech Prosody Studies Group, IEL/Unicamp | Lab. de Fon�tica, FALE/UFMG | LIACC, LAEL, PUC-SP
Important Dates
Call for Papers: May 15, 2007
Full Paper Submission: Nov. 2nd, 2007
Notif. of Acceptance: Dec. 14th, 2007
Early Registration: Jan. 14th, 2008
Conference: May 6-9, 2008

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CFP:The International Workshop on Spoken Languages Technologies for Under- The International Workshop on Spoken Languages Technologies for Under-resourced languages (SLTU)

The International Workshop on Spoken Languages Technologies for Under-resourced languages (SLTU)

languages (SLTU) Hanoi University of Technology, Hanoi, Vietnam,

May 5 - May 7, 2008.

EXTENDED DEADLINE  30 January 2008

 Workshop Web Site :        http://www.mica.edu.vn/sltu

The STLU meeting is a technical conference focused on spoken language processing for

under-resourced languages. This first workshop will focus on Asian languages, and

the idea is to mainly (but not exclusively) target languages of the area (Vietnamese,

Khmer, Lao, Chinese dialects, Thai, etc.). However, all contributions on other

under-resourced languages of the world are warmly welcomed.  The workshop aims

at gathering researchers working on:

* ASR, synthesis and speech translation for under-resourced languages

* portability issues  * fast resources acquisition (speech, text, lexicons, parallel corpora)

* spoken language processing for languages with rich morphology

* spoken language processing for languages without separators

* spoken language processing for languages without writing system

Important dates

* Paper submission dqte: EXTENDED to January 30, 2008

* Notification of Paper Acceptance: February 20, 2008

* Author Registration Deadline: March 1, 2008   Scientific Committee

* Pr Tanja Schultz, CMU, USA

* Dr Yuqing Gao, IBM, USA

* Dr Lori Lamel, LIMSI, France

* Dr Laurent Besacier, LIG, France

* Dr Pascal Nocera, LIA, France

* Pr Jean-Paul Haton, LORIA, France

* Pr Luong Chi Mai, IOIT, Vietnam

* Pr Dang Van Chuyet, HUT, Vietnam

* Pr Pham Thi Ngoc Yen, MICA, Vietnam

* Dr Eric Castelli, MICA, Vietnam

* Dr Vincent Berment, LIG Laboratory, France

* Dr Briony Williams, University of Wales, UK

Local Organizing Committee

* Pr Nguyen Trong Giang, HUT/MICA

* Pr Ha Duyen Tu, HUT

* Pr Pham Thi Ngoc Yen, HUT/MICA

* Pr Genevi�ve Caelen-Haumont, MICA

* Dr Trinh Van Loan, HUT

* Dr Mathias Rossignol, MICA

* M. Hoang Xuan Lan, HUT

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SIGDIAL 2008 9th SIGdial Workshop on Discourse and Dialogue

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

** Submission Deadline: Feb 15 2008 **

1st CALL FOR PAPERS

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 Feb 15 2008

Notification Mar 31 2008

Final submissions Apr 14 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
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LIPS 2008 Visual Speech Synthesis Challenge

LIPS 2008: Visual Speech Synthesis Challenge

LIPS 2008 is the first visual speech synthesis challenge. It will be

held as a special session at INTERSPEECH 2008 in Brisbane, Australia

(http://www.interspeech2008.org). The aim of this challenge is to

stimulate discussion about subjective quality assessment of synthesised

visual speech with a view to developing standardised evaluation procedures.

In association with this challenge a training corpus of audiovisual

speech and accompanying phoneme labels and timings will be provided to

all entrants, who should then train their systems using this data. (As

this is the first year the challenge will run and to promote wider

participation, proposed entrants are free to use a pre-trained model).

Prior to the session a set of test sentences (provided as audio, video

and phonetic labels) must be synthesised on-site in a supervised room. A

series of double-blind subjective tests will then be conducted to

compare each competing system against all others. The overall winner

will be announced and presented with their prize at the closing ceremony

of the conference.

All entrants will submit a 4/6 (TBC) page paper describing their system

to INTERSPEECH indicating that the paper is addressed to the LIPS special

session. A special edition of the Eurasip Journal on Speech, Audio and Music

Processing in conjunction with the challenge is also scheduled.

To receive updated information as it becomes available, you can join the

mailing list by visiting

https://mail.icp.inpg.fr/mailman/listinfo/lips_challenge. Further

details will be mailed to you in due course.

Please invite colleagues to join and dispatch this email largely to your

academic and industrial partners. Besides a large participation of

research groups in audiovisual speech synthesis and talking faces we

particularly welcome participation of the computer game industry.

Please confirm your willingness to participate in the challenge, submit

a paper describing your work and join us in Brisbane by sending an email

to sascha.fagel@tu-berlin.de, b.theobald@uea.ac.uk,

gerard.bailly@gipsa-lab.inpg.fr

Organising Committee

Sascha Fagel, University of Technology, Berlin - Germany

Barry-John Theobald, University of East Anglia, Norwich - UK

Gerard Bailly, GIPSA-Lab, Dpt. Speech & Cognition, Grenoble - France
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Future Speech Science and Technology Events

e-Forensics 2008: The 1st International Conference on Forensic Applications and Techniques in Telecommunications, Information and Multimedia

with associated Workshops:
WKDD: First International Workshop on Knowledge Discovery and Data Mining
and WFST: First International Workshop on Forensics Sensor Technology
Adelaide, Australia, January 21-23, 2008
Website
The birth of the Internet as a commercial entity and the development of increasingly sophisticated digital consumer technology has led to new opportunities for criminals and new challenges for law enforcement. Those same technologies also offer new tools for scientific investigation of evidence.
In telecommunications, Voice over IP raises significant challenges for call intercept and route tracing. Information systems are becoming overwhelmingly large and the challenges of associating relevant information from one source with another require new sophisticated tools. And consumer multimedia devices, especially video and still cameras, are increasingly becoming the tools of choice to create potentially illegal content. At the same time, the scientific gathering and investigation of evidence at a crime scene is being pushed towards digital techniques, raising questions of the veracity and completeness of evidence.
PAPERS: The aim of this conference is to bring together state of the art research contributions in the development of tools, protocols and techniques which assist in the investigation of potentially illegal activity associated with electronic communication and electronic devices. Investigative practice and requirements for presentation of evidence in court are to be considered key underlying themes. This might include discovery, analysis, handling and storage of digital evidence; meeting the legal burden of proof; and the establishment of the forensic chain of evidence.
Technical papers describing original, previously unpublished research, not currently under review by another conference or journal, are solicited. Technical papers which clearly identify how the specific contributions fit to an overall working solution are particularly of interest.
Topics include, but are not limited to, the following:
* Voice over IP call tracing and intercept
* Records tracing and data mining
* Fraud management in commercial transactions
* Techniques for addressing identity theft
* Geo-location techniques for cellular, ad-hoc, wireless and IP network communications
* Distributed data association across massive, disparate database systems
* Data carving
* Multimedia source identification
* Image tamper identification
* Image association and recognition
* Motion analysis
* Voice analysis
* Watermarking and applications
* Transaction tracking
* Digital evidence storage and handling protocols
General papers on electronic security will be considered where there is a clear application to the underlying topic of forensic investigation.
Important Dates:
Paper Registration Deadline: September 28, 2007
Notification of Acceptance: November 16, 2007
Final paper submission and author's registration: December 3, 2007
Conference Dates: January 21-23, 2008
Committee
General Chair - Matthew Sorell, University of Adelaide, Australia
Technical Program Committee Chair - Chang-Tsun Li, University of Warwick, UK
Publicity Chair - Gale Spring, RMIT University, Australia
For details of the Workshops, and further information regarding paper submission and the conference, please refer to the conference website
Please direct all enquiries by e-mail
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LangTech2008

The language and speech technology conference.
Rome, 28-29 February 2008
San Michele a Ripa conference centre.
Website
We are most delighted to welcome you to join us at the LangTech2008 conference which will be held at the San Michele a Ripa convention center in Rome, February 28-29, 2008. After two successful national conferences on speech and language technology (2002, 2006), the ForumTal decided to promote an international event in the field. A follow up of the previous LangTech conferences (Berlin, Paris), LangTech2008 aims at giving a chance to the industrial and research communities, and public administration, to share and discuss language and speech technologies. The conference will feature world-class speakers, exhibits, lecture and poster sessions.

PAPERS SUBMISSION DEADLINE: 30th November 2007

EXHIBITION BOOTHS RESERVATION: Reduced Fares until 15th November 2007

REGISTRATION: Reduced Fees until 31st December 2007

A golden promotional opportunity for all language technology SMEs!

LangTech 2008, http://www.langtech.it/en/, the language technology business conference, is featuring a special elevator session for small and medium sized enterprises, SMEs.

An elevator session is a session with very short presentations.

If you seek business partners, you are invited to participate in LangTech 2008 in Rome, February 28-29, and make yourself known to the audience.

A committee of European experts shall choose a total of 10 SMEs from anywhere in Europe and beyond to give a 5 min self-promotional presentation in English before a floor of venture capitalists, business peers, large technology corporations and other interested parties.

A jury will select three of the presenting companies, and award the first, second and third LangTech Prize.

Submissions must be received by 30 December 2007.

The lucky candidates will be informed by 15 January 2008.

We will offer a reduced fee to LangTech 2008 to all SMEs selected to present at the elevator session.

If you wish to submit a request to present your SME for this unique opportunity, please contact sme@langtech.it immediately, and visit the web site dedicated to LangTech 2008, http://www.langtech.it/en/, where you can download a short slide set with guidelines for preparing your candidature.

Dr Calzolari would be pleased if you could spread the Conference Announcement and the Call for SMEs

Presentations to anyone you consider potentially interested in the event.

Dr PAOLA BARONI

Researcher

Consiglio Nazionale delle Ricerche

Istituto di Linguistica Computazionale

Area della Ricerca di Pisa

Via Giuseppe Moruzzi

56124 Pisa

ITALY

Phone: [+39] 050 315 2873

Fax: [+39] 050 315 2834

e-Mail: paola.baroni@ilc.cnr.it

URL: http://www.ilc.cnr.it

Skype: paola.baroni
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AVIOS

San Diego, March 10 - 12, 2008
The defining conference on Voice Search
From the Applied Voice Input Output Society and Bill Meisel's TMA Associates
Voice Search 2008 will be held at the San Diego Marriott Hotel and Marina, San Diego, California, March 10 - 12, 2008. Voice Search is a rapidly evolving technology and market. AVIOS (the Applied Voice Input Output Society) and Bill Meisel (president of TMA Associates and Editor of Speech Strategy News) are joining together to launch this new conference as a definitive resource for companies that will be impacted by this important trend.
"Voice Search" suggests an analogy to "Web Search," which has been a runaway success for both users and providers. The maturing of speech recognition and text-to-speech synthesis--and the recent involvement of large companies--has validated the availability of the core functionality necessary to support this analogy. The conference explores the possibilities, limitations, and differences of Voice Search and Web search.
Web search made the Web an effective and required marketing tool. Will Voice Search do the same for the telephone channel? The potential impact on call centers is another key issue covered by the conference.
The agenda covers:
� What Voice Search is and will become
� Applications of Voice Search
� The appropriate use of speech technology to support voice search
� Insight for service providers, enterprises, Web services, and call centers that want to take advantage of this new resource
� Marketing channels and business models in Voice Search
� Emerging supporting technology, development tools, and delivery platforms supporting Voice Search
� Dealing with the surge of calls created by Voice Search.
Specific topics that will be covered at Voice Search 2008 include:
Applications
- Automated directory assistance and local search
- Voice information searches by telephone
- Ad-supported information access by phone
- Audio/Video searches on the Web and enterprises
- Speech analytics-extracting business intelligence from audio files
- Converting voicemail to searchable text
- Other new applications and services
- Application examples and demonstrations
Markets
- How the voice search market is developing
- The changing role of the telephone in marketing
- Business models
- The right way to deliver audio ads
- Justifying subscriber fees
Delivery
- Platforms, tools, and services for effectively delivering these applications
- Implementation examples and demonstrations
- Hosted versus customer-premises solutions
- Supporting multiple modes of interaction
- Key sources of technology and service
Contact centers
- The impact of Voice Search on contact centers
- Speech automation to handle the increased call flow
- Moving from handling problems to building customer relationships
Technology
- Speech recognition methods supporting voice search
- Text-to-speech quality and alternatives
- Supporting multimodal solutions
- Supporting standards
- Delivering responsive applications
- Voice User Interface issues and solutions in voice search
Sponsorships are available:
http://www.voicesearchconference.com/sponsor.htm
We're interested in proposals for speaking (available slots are limited):
http://www.voicesearchconference.com/talk.htm
Registration is open with an early-registration discount:
http://www.voicesearchconference.com/registration.htm
Other information:
What is Voice Search?
Voice Search News
About AVIOS
About Bill Meisel and TMA Associates
Or contact cONTACT.
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CfP-2nd INTERNATIONAL CONFERENCE ON LANGUAGE AND AUTOMATA THEORY AND APPLICATIONS (LATA 2008)

Tarragona, Spain, March 13-19, 2008
Website http://www.grlmc.com
AIMS:
LATA is a yearly conference in theoretical computer science and its applications. As linked to the International PhD School in Formal Languages and Applications that is being developed at the host institute since 2001, LATA 2008 will reserve significant room for young computer scientists at the beginning of their career. It will aim at attracting scholars from both classical theory fields and application areas (bioinformatics, systems biology, language technology, artificial intelligence, etc.). SCOPE:
Topics of either theoretical or applied interest include, but are not limited to:
- words, languages and automata
- grammars (Chomsky hierarchy, contextual, multidimensional, unification, categorial, etc.)
- grammars and automata architectures
- extended automata
- combinatorics on words
- language varieties and semigroups
- algebraic language theory
- computability
- computational and structural complexity
- decidability questions on words and languages
- patterns and codes
- symbolic dynamics
- regulated rewriting
- trees, tree languages and tree machines
- term rewriting
- graphs and graph transformation
- power series
- fuzzy and rough languages
- cellular automata
- DNA and other models of bio-inspired computing
- symbolic neural networks
- quantum, chemical and optical computing
- biomolecular nanotechnology
- automata and logic
- algorithms on automata and words
- automata for system analysis and programme verification
- automata, concurrency and Petri nets
- parsing
- weighted machines
- transducers
- foundations of finite state technology
- grammatical inference and algorithmic learning
- text retrieval, pattern matching and pattern recognition
- text algorithms
- string and combinatorial issues in computational biology and bioinformatics
- mathematical evolutionary genomics
- language-based cryptography
- data and image compression
- circuits and networks
- language-theoretic foundations of artificial intelligence and artificial life
- digital libraries
- document engineering
STRUCTURE:
LATA 2008 will consist of:
- 3 invited talks (to be announced in the second call for papers)
- 2 tutorials (to be announced in the second call for papers)
- refereed contributions
- open sessions for discussion in specific subfields or on professional issues
SUBMISSIONS:
Authors are invited to submit papers presenting original and unpublished research. Papers should not exceed 12 pages and should be formatted according to the usual LNCS article style. Submissions have to be sent through the web page.
PUBLICATION:
A volume of proceedings (expectedly LNCS) will be available by the time of the conference. A refereed volume of selected proceedings containing extended papers will be published soon after it as a special issue of a major journal.
REGISTRATION:
The period for registration will be open since January 7 to March 13, 2008. Details about how to register will be provided through the website of the conference.
Early registration fees: 250 euros
Early registration fees (PhD students): 100 euros
Registration fees: 350 euros
Registration fees (PhD students): 150 euros
FUNDING:
25 grants covering partial-board accommodation will be available for nonlocal PhD students. To apply, the candidate must e-mail her/his CV together with a copy of the document proving her/his status as a PhD student.
IMPORTANT DATES:
Paper submission: November 16, 2007
Application for funding (PhD students): December 7, 2007
Notification of funding acceptance or rejection: December 21, 2007
Notification of paper acceptance or rejection: January 18, 2008
Early registration: February 1, 2008
Final version of the paper for the proceedings: February 15, 2008
Starting of the conference: March 13, 2008
Submission to the journal issue: May 23, 2008
FURTHER INFORMATION:
E-mail
Website http://www.grlmc.com
ADDRESS:
LATA 2008
Research Group on Mathematical Linguistics
Rovira i Virgili University
Plaza Imperial Tarraco, 1
43005 Tarragona, Spain
Phone: +34-977-559543
Fax: +34-977-559597
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CfP Workshop on Empirical Approaches to Speech Rhythm

CALL FOR PAPERS

*** Workshop on Empirical Approaches to Speech Rhythm ***

Centre for Human Communication

UCL

Abstracts due: 31st January 2008

Workshop date: 28th March 2008

Empirical studies of speech rhythm are becoming increasingly popular.

Metrics for the quantification of rhythm have been applied to

typological, developmental, pathological and perceptual questions.

The prevalence of rhythm metrics based on durational characteristics

of consonantal and vocalic intervals (e.g. deltaV, deltaC, %V, nPVI-

V, rPVI-C, VarcoV and VarcoC) indicate the need for agreement about

their relative efficacy and reliability. More fundamentally, it

remains to be demonstrated whether such metrics really quantify

speech rhythm, a controversial and elusive concept.

Confirmed speakers:

Francis Nolan (Cambridge) - keynote speaker

Fred Cummins (UCD)

Volker Dellwo (UCL)

Klaus Kohler (Kiel)

Elinor Payne (Oxford)

Petra Wagner (Bonn)

Laurence White (Bristol)

Abstracts:

We invite abstract submissions for a limited number of additional

oral presentations, and for poster presentations. We welcome

abstracts that address any or all of the following questions:

- What is speech rhythm?

- How should we measure speech rhythm?

- Which rhythm metrics are most effective and reliable?

- What can rhythm metrics tell us?

- What are the limitations of rhythm metrics?

Publication:

It is intended that a limited number of contributions to the workshop

may be published in a special issue of Phonetica. Initial selection

of papers will be made after the workshop with a view to compiling a

thematically coherent publication. Selected papers will subsequently

be reviewed.

Important dates:

Abstracts must be received by: 31st January 2008

Notification of acceptance: 15th February 2008

Date of Workshop: 28th March 2008

Abstract submission:

Abstracts should be sent to: rhythm2008@phon.ucl.ac.uk. Abstracts

should be in Word or rtf format, 12pt Times New Roman, 1.5 line

spacing, and no longer than one page of A4. The file should be

entitled RhythmWorkshop-[name].doc, where [name] is the last name of

the first author. The abstract should start with:

- the title of the abstract in bold and centered;

- the name(s) and department(s) of the author(s) in italics and

centered;

- the email address(es) of the author(s), centred.

The body of the abstract should be justified left and right.

Further information:

For more information and updates please check www.phon.ucl.ac.uk/

rhythm2008. Email enquiries should be directed to

rhythm2008@phon.ucl.ac.uk.

On behalf of the scientific organizing committee:

Volker Dellwo, Elinor Payne, Petra Wagner and Laurence White
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IEEE International Conference on Acoustics, Speech and Signal Processng ICASSP

Las Vegas USA

30 March 4 April 2008

The world's largest and most comprehensive technical conference focused on signal processing.

Website: http://www.icassp2008.org/
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CfAbstracts Prosody and expressivity in speech and music

CALL FOR ABSTRACTS :

Prosody and expressivity in speech and music

Satellite Event around Speech Prosody 2008 / First EMUS Conference - Expressivity in MUsic and Speech
http://www.sp2008.org/events.php / http://recherche.ircam.fr/equipes/analyse-synthese/EMUS

Campinas, Brazil, May 5th, 2008
[Abstract submission deadline: January 30th, 2008]
Keywords: emotion, expressivity, prosody, music, acquisition, perception, production,
interpretation, cognitive sciences, neurosciences, acoustic analysis.

DESCRIPTION:
Speech and music conceal a treasure of "expressive potential" for they can activate sequences of varied
emotional experiences in the listener. Beyond their semiotic differences, speech and music share acoustic
features such as duration, intensity, and pitch, and have their own internal organization, with their own
rhythms, colors, timbres and tones.
The aim of this workshop is to question the connections between various forms of expressivity, and the
prosodic and gestural dimensions in the spheres of music and speech. We will first tackle the links
between speech and music through enaction and embodied cognition. We will then work on computer
modelling for speech and music synthesis. The third part will focus on musicological and aesthetic
perspectives. We will end the workshop with a round table in order to create a dialogue between the
various angles used to apprehend prosody and expressivity in both speech and music.

FRAMEWORK:
This workshop will be the starting point of a string of events on the relations between language and music:
May 16th: Prosody, Babbling and Music (Ecole Normale Sup�rieure Lettres et Sciences Humaines, Lyon)
June 17-18th: Prosody of Expressivity in Music and Speech (IRCAM, Paris)
September 25th and 26th: Semiotics and microgenesis of verbal and musical forms (RISC, Paris).
Our aim is to make links between several fields of research and create a community interested in the
relations between music and language. The project will be materialized in a final publication of the
keynote papers of those four events.

SUBMISSION PROCEDURE:
The workshop will host about ten posters.
Authors should submit an extended abstract to: beller@ircam.fr in pdf format by January 30, 2008.
We will send an email confirming the reception of the submission. The suggested abstract length is
maximum 1 page, formatted in standard style.
The authors of the accepted abstracts will be allocated as poster highlights. Time will be allocated
in the programme for poster presentations and discussions.
Before the workshop, the extended abstracts (maximum 4 pages) will be made available to a broader
audience on the workshop web site. We also plan to maintain the web page after the workshop and
encourage the authors to submit slides and posters with relevant links to their personal web pages.

KEY DATES:
Dec 10: Workshop announcement and Call for Abstracts
Jan 30: Abstract submission deadline
Mar 28: Notification of acceptance
Apr 25: Final extended abstracts due
May 5: Workshop

SCIENTIFIC COMMITTEE:
Christophe d'Alessandro (LIMSI, Orsay);
Antoine Auchlin (University of Geneva, Linguistics Department);
Gr�gory Beller (IRCAM);
Nick Campbell (ATR, Nara);
Anne Lacheret (MODYCO,
Nanterre University) ;
Sandra Madureira (PUC-SP);
Aliyah Morgenstern (ICAR, Ecole Normale Sup�rieure Lettres et Sciences Humaines) ;
Nicolas Obin (IRCAM)

ORGANISERS:
- University of Geneva, Linguistics Department (Antoine Auchlin)
- IRCAM (Gr�gory Beller and Nicolas Obin)
- MODYCO, Nanterre University (Anne Lacheret)
- ICAR, Ecole Normale Sup�rieure Lettres et Sciences Humaines (Aliyah Morgenstern)

CONTACT :
For questions/ suggestions about the workshop, please contact beller@ircam.fr
Please refer to http://recherche.ircam.fr/equipes/analyse-synthese/EMUS for
up-to-date information about the workshop.

PROGRAM
http://www.sp2008.org/events/EMUS-conferences.pdf
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CfP- LREC 2008 - 6th Language Resources and Evaluation Conference

Palais des Congr�s Mansour Eddahbi, MARRAKECH - MOROCCO
MAIN CONFERENCE: 28-29-30 MAY 2008
WORKSHOPS and TUTORIALS: 26-27 MAY and 31 MAY- 1 JUNE 2008
Conference web site
The sixth international conference on Language Resources and Evaluation (LREC) will be organised in 2008 by ELRA in cooperation with a wide range of international associations and organisations.
CONFERENCE TOPICS
Issues in the design, construction and use of Language Resources (LRs): text, speech, multimodality
- Guidelines, standards, specifications, models and best practices for LRs
- Methodologies and tools for LRs construction and annotation
- Methodologies and tools for the extraction and acquisition of knowledge
- Ontologies and knowledge representation
- Terminology
- Integration between (multilingual) LRs, ontologies and Semantic Web technologies
- Metadata descriptions of LRs and metadata for semantic/content markup
Exploitation of LRs in different types of systems and applications
- For: information extraction, information retrieval, speech dictation, mobile communication, machine translation, summarisation, web services, semantic search, text mining, inferencing, reasoning, etc.
- In different types of interfaces: (speech-based) dialogue systems, natural language and multimodal/multisensorial interactions, voice activated services, etc.
- Communication with neighbouring fields of applications, e.g. e-government, e-culture, e-health, e-participation, mobile applications, etc.
- Industrial LRs requirements, user needs
Issues in Human Language Technologies evaluation
- HLT Evaluation methodologies, protocols and measures
- Validation, quality assurance, evaluation of LRs
- Benchmarking of systems and products Usability evaluation of HLT-based user interfaces, interactions and dialog systems
- Usability and user satisfaction evaluation
General issues regarding LRs & Evaluation
- National and international activities and projects
- Priorities, perspectives, strategies in national and international policies for LRs
- Open architectures
- Organisational, economical and legal issues
Special Highlights
LREC targets the integration of different types of LRs - spoken, written, and other modalities - and of the respective communities. To this end, LREC encourages submissions covering issues which are common to different types of LRs and language technologies.
LRs are currently developed and deployed in a much wider range of applications and domains. LREC 2008 recognises the need to encompass all those data that interact with language resources in an attempt to model more complex human processes and develop more complex systems, and encourages submissions on topics such as:
- Multimodal and multimedia systems, for Human-Machine interfaces, Human-Human interactions, and content processing
- Resources for modelling language-related cognitive processes, including emotions
- Interaction/Association of language and perception data, also for robotic systems
PROGRAMME
The Scientific Programme will include invited talks, oral presentations, poster and demo presentations, and panels. There is no difference in quality between oral and poster presentations. Only the appropriateness of the type of communication (more or less interactive) to the content of the paper will be considered.
SUBMISSIONS AND DATES
Submitted abstracts of papers for oral and poster or demo presentations should consist of about 1500-2000 words.
- Submission of proposals for oral and poster/demo papers: 31 October 2007
Proposals for panels, workshops and tutorials will be reviewed by the Programme Committee.
- Submission of proposals for panels, workshops and tutorials: 31 October 2007
PROCEEDINGS
The Proceedings on CD will include both oral and poster papers, in the same format. In addition a Book of Abstracts will be printed.
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Call for Papers: HLT & NLP within the Arabic world Workshop at LREC 2008

*
HLT & NLP within the Arabic world: Arabic Language and **local ** languages processing: Status Updates and Prospects *

Please refer to http://www.lrec-conf.org/lrec2008/Workshops.html for details.

* Motivation and Aims*

This Workshop intends to add value to the issues addressed during the main conference (Human Language Technologies (HLT) & Natural Language Processing (NLP)) and enhance the work carried out at different places to process Arabic language(s) and more generally Semitic languages and other local and foreign languages spoken in the region.

It should bring together people who are actively involved in Arabic Written and Spoken language processing in a mono- or cross/multilingual context, and give them an opportunity to update the community through reports on completed and ongoing work as well as on the availability of LRs, evaluation protocols and campaigns, products and core technologies (in particular open source ones). This should enable the participants to develop a common view on where we stand with respect to these particular set of languages and to foster the discussion of the future of this research area. Particular attention will be paid to activities involving technologies such as Machine Translation, Cross-Lingual Information Retrieval/extraction, Summarization, Speech to text transcriptions, etc., and languages such as Arabic varieties, Amazigh, Amharic, Hebrew, Maltese, and other local languages. Evaluation methodologies and resources for evaluation of HLT are also a main focus.

* Topics of Interest *

The submissions should address some of the following issues:

� Issues in the design, the acquisition, creation, management, access, distribution, use of Language Resources (Standard Arabic, Colloquial Arabic, other Semitic languages, Amazigh, Coptic, Maltese, English/French spoken locally, etc.)

� Impact on LR collections/processing and NLP of the crucial issues related to "code switching" between different dialects and languages

� Specific issues related to the above-mentioned languages such as role of morphology, named entities, corpus alignment, etc.)

� Multilinguality issues including relationship between Colloquial and Standard Arabic

� Exploitation of LR in different types of applications

� Industrial LR requirements and community's response;

� Benchmarking of systems and products; resources for benchmarking and evaluation for written and spoken language processing;

� Focus on some key technologies such as MT (all approaches e.g. Statistical, Example-Based, etc.), Information Retrieval, Speech Recognition, Spoken Documents Retrieval, CLIR, Question-Answering, Summarization,

� Local, regional, and international activities and projects;

� Needs, possibilities, forms, initiatives of/for regional and international cooperation.

* Submission Details (more on http://www.lrec-conf.org/lrec2008/Workshops.html) *

Submissions must be in English. Abstracts for workshop contributions should not exceed Four A4 pages (excluding references). An additional title page should state: the title; author(s); affiliation(s); and contact author's e-mail address, as well as postal address, telephone and fax numbers.

Submission is to be sent by email, preferably in Postscript or PDF format, to: arabic@elda.org mailto:choukri@elda.org to arrive before * 15 February 2008 * .

Registration to LREC'08 will be required for participation, so potential participants are invited to refer to the main conference website for all details not covered in the present call ( http://www.lrec-conf.org/lrec2008/)

* Important Dates *

Call for papers: 3 January 2008

Deadline for abstract submissions: 15 February 2008

Notification of acceptance: 14 March 2008

Final version of accepted paper: 11 April 2008

Workshop full-day: Saturday 31^st May 2008

* Workshop chair *

Khalid Choukri (ELRA/ELDA, France )

* Workshop Co-chairs *

Mona Diab, Columbia University , USA

Bente Maegaard (CST, University of Copenhagen , Denmark )

Paolo Rosso, Universidad Polit�cnica Valencia , Spain

Abdelhadi Soudi ENIM ( Morocco )

Ali Farghaly, Oracle USA and Monterey Institute of International Studies,
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CfP ELRA Workshop on Evaluation

CALL FOR PAPERS

ELRA Workshop on Evaluation

Looking into the Future of Evaluation: when automatic metrics meet

task-based and performance-based approaches

To be held in conjunction with the 6th International Language Resources

and Evaluation Conference (LREC 2008)

27 May 2008, Palais des Congr�s Mansour Eddahbi, Marrakech

Background

Automatic methods to evaluate system performance play an important role

in the development of a language technology system. They speed up

research and development by allowing fast feedback, and the idea is also

to make results comparable while aiming to match human evaluation in

terms of output evaluation. However, after several years of study and

exploitation of such metrics we still face problems like the following ones:

* they only evaluate part of what should be evaluated

* they produce measurements that are hard to understand/explain, and/or

hard to relate to the concept of quality

* they fail to match human evaluation

* they require resources that are expensive to create

etc. Therefore, an effort to integrate knowledge from a multitude of

evaluation activities and methodologies should help us solve some of

these immediate problems and avoid creating new metrics that reproduce

such problems.

Looking at MT as a sample case, problems to be immediately pointed out

are twofold: reference translations and distance measurement. The former

are difficult and expensive to produce, they do not cover the usually

wide spectrum of translation possibilities and what is even more

discouraging, worse results are obtained when reference translations are

of higher quality (more spontaneous and natural, and thus, sometimes

more lexically and syntactically distant from the source text).

Regarding the latter, the measurement of the distance between the source

text and the output text is carried out by means of automatic metrics

that do not match human intuition as well as claimed. Furthermore,

different metrics perform differently, which has already led researchers

to study metric/approach combinations which integrate automatic methods

into a deeper linguistically oriented evaluation. Hopefully, this should

help soften the unfair treatment received by some rule-based systems,

clearly punished by certain system-approach sensitive metrics.

On the other hand, there is the key issue of � what needs to be measured

�, so as to draw the conclusion that � something is of good quality �,

or probably rather � something is useful for a particular purpose �. In

this regard, works like those done within the FEMTI framework have shown

that aspects such as usability, reliability, efficiency, portability,

etc. should also be considered. However, the measuring of such quality

characteristics cannot always be automated, and there may be many other

aspects that could be usefully measured.

This workshop follows the evolution of a series of workshops where

methodological problems, not only for MT but for evaluation in general,

have been approached. Along the lines of these discussions and aiming to

go one step further, the current workshop, while taking into account the

advantages of automatic methods and the shortcomings of current methods,

should focus on task-based and performance-based approaches for

evaluation of natural language applications, with key questions such as:

- How can it be determined how useful a given system is for a given task?

- How can focusing on such issues and combining these approaches with

our already acquired experience on automatic evaluation help us develop

new metrics and methodologies which do not feature the shortcomings of

current automatic metrics?

- Should we work on hybrid methodologies of automatic and human

evaluation for certain technologies and not for others?

- Can we already envisage the integration of these approaches?

- Can we already plan for some immediate collaborations/experiments?

- What would it mean for the FEMTI framework to be extended to other HLT

applications, such as summarization, IE, or QA? Which new aspects would

it need to cover?

We solicit papers that address these questions and other related issues

relevant to the workshop.

Workshop Programme and Audience Addressed

This full-day workshop is intended for researchers and developers on

different evaluation technologies, with experience on the various issues

concerned in the call, and interested in defining a methodology to move

forward.

The workshop feature invited talks, submitted papers, and will conclude

with a discussion on future developments and collaboration.

Workshop Chairing Team

Gregor Thurmair (Linguatec Sprachtechnologien GmbH, Germany) - chair

Khalid Choukri (ELDA - Evaluations and Language resources Distribution

Agency, France) - co-chair

Bente Maegaard (CST, University of Copenhagen, Denmark) - co-chair

Organising Committee

Victoria Arranz (ELDA - Evaluations and Language resources Distribution

Agency, France)

Khalid Choukri (ELDA - Evaluations and Language resources Distribution

Agency, France)

Christopher Cieri (LDC - Linguistic Data Consortium, USA)

Eduard Hovy (Information Sciences Institute of the University of

Southern California, USA)

Bente Maegaard (CST, University of Copenhagen, Denmark)

Keith J. Miller (The MITRE Corporation, USA)

Satoshi Nakamura (National Institute of Information and Communications

Technology, Japan)

Andrei Popescu-Belis (IDIAP Research Institute, Switzerland)

Gregor Thurmair (Linguatec Sprachtechnologien GmbH, Germany)

Important dates

Deadline for abstracts: Monday 28 January 2008

Notification to Authors: Monday 3 March 2008

Submission of Final Version: Tuesday 25 March 2008

Workshop: Tuesday 27 May 2008

Submission Format

Abstracts should be no longer than 1500 words and should be submitted in

PDF format to Gregor Thurmair at g.thurmair@linguatec.de.
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CALL for JEP/TALN/RECITAL 2008 - Avignon France

JEP-TALN-RECITAL'08
- XXVIIemes Journees d'Etude sur la Parole (JEP'08)
- 15eme conference sur le Traitement Automatique des Langues Naturelles (TALN'08)
- 10eme Rencontre des Etudiants Chercheurs en Informatique pour le Traitement Automatique des Langues (RECITAL'08)

Universite d'Avignon et des Pays de Vaucluse
Avignon du 9 au 13 Juin 2008

Pour la troisieme fois, apres Nancy en 2002 et Fes en 2004, l'AFCP(Association Francophone pour la Communication Parlee) et l'ATALA (Association pour le Traitement Automatique des Langues) organisent conjointement leur principale conference afin de reunir en un seul lieu les deux communautes du traitement de la langue orale et ecrite.

Le colloque jeunes chercheurs RECITAL'08 est egalement associe a cet evenement.

Les conferences invitees ainsi qu'une session orale thematique seront organisees sous forme de sessions pleinieres communes aux trois conferences. L'inscription est commune aux trois evenements et les participants recevront l'ensemble des actes sur CDROM. La langue officielle est le francais.

Organise par l e LIA (Laboratoire Informatique d'Avignon), cet evenement se tiendra, du 9 au 13 juin 2008, a l'Universite d'Avignon et des Pays du Vaucluse (site centre ville - sainte Marthe).

Les JEP 2008 sont organisees sous l'egide de l'AFCP (Association Francophone pour la Communication Parlee), avec le soutien de l'ISCA (International Speech Communication Association).

CALENDRIER

Date limite de soumission: 11 fevrier 2008

Notification aux auteurs: 28 mars 2008

Conference: 9-13 juin 2008

THEMES

Les communications porteront sur la communication parlee et le traitement de la parole dans leurs differents aspects. Les themes de la conference incluent, de facon non limitative:

Production de parole

Acoustique de la parole

Perception de parole

Phonetique et phonologie

Prosodie

Reconnaissance et comprehension de la parole

Reconnaissance de la langue et du locuteur

Modeles de langage

Synthese de la parole

Analyse, codage et compression de la parole

Applications a composantes orales (dialogue, indexation...)

Evaluation, corpus et ressources

Psycholinguistique

Acquisition de la parole et du langage

Apprentissage d'une langue seconde

Pathologies de la parole

CRITERES DE SELECTION

Les auteurs sont invites a soumettre des travaux de recherche

originaux, n'ayant pas fait l'objet de publications anterieures. Les

contributions proposees seront examinees par au moins deux

specialistes du domaine. Seront considerees en particulier:

- l'importance et l'originalite de la contribution,

- la correction du contenu scientifique et technique,

- la discussion critique des resultats, en particulier par rapport aux autres travaux du domaine

- la situation des travaux dans le contexte de la recherche internationale,

- l'organisation et la clarte de la presentation

- l'adequation aux themes de la conference.

Les articles selectionnes seront publies dans les actes de la conference.

MODALITES DE SOUMISSION

Les articles soumis ne devront pas depasser 4 pages en Times 10, sur deux colonnes, format A4. Une feuille de style LaTeX et un modee Word sont disponibles sur le site de la conference www.lia.univ-avignon.fr/jep-taln08/.

Les articles devront etre soumis - sous forme electronique - sur le site web de la conference avant le 11 fevrier 2008. Les documents devront etre envoyes exclusivement au format PDF.

BOURSES

L'AFCP offre un certain nombre de bourses pour les doctorants et jeunes chercheurs desireux de prendre part a la conference, voir : www.afcp-parole.org/doc/bourses.htm

L'ISCA apporte egalement un soutien financier aux jeunes chercheurs participant a des manifestations scientifiques sur la parole et le langage, voir : www.isca-speech.org/grants.html

Site web

CALL FOR WORKSHOPS AND TUTORIALS

For the third time, after Nancy in 2002 and Fes in 2004, the French speech association AFCP and the French NLP association ATALA are jointly organising their main conference in order to group together the two research community working in the fields of Speech and Natural Language Processing.

The conference will include oral and poster communications, invited conferences, workshops and tutorials. Workshop and tutorials will be held on June 13, 2008.

The official languages are French and English.

IMPORTANT DATES

Deadline for proposals: November 22nd 2007

Approval by the TALN committee: November 30th 2007

Final version for inclusion in the proceedings: April 4th 2008

Workshop and tutorials: June 13th 2008

OBJECTIVES

Workshops can be organized on any specific aspect of NLP. The aim of these sessions is to facilitate an in-depth discussion of this theme.

A workshop has its own president and its own program committee. The president is responsible for organizing a call for paper/participation and for the coordination of his program committee. The organizers ofthe main TALN conference will only take in charge the organization of the usual practical details (rooms, coffee breaks, proceedings).

Workshops will be organized in parallel sessions on the last day of the conference (2 to 4 sessions of 1:30).

Tutorials will be held on the same day.

HOW TO SUBMIT

Workshop and Tutorial proposals will be sent by email to taln08@atala.org before November 22nd, 2007.

** Workshop proposals will contain an abstract presenting the proposed theme, the program committee list and the expected length of the session.

** Tutorial proposals will contain an abstract presenting the proposed theme, a list of all the speakers and the expected length of the session (1 or 2 sessions of 1:30).

The TALN program committee will make a selection of the proposals and announce it on November 30th, 2008.

FORMAT

Conferences will be given in French or English (for non French native speakers). Papers to be published in the proceedings will conform to the TALN style sheet which is available on the conference web site. Worshop papers should not be longer than 10 pages in Times 12 (references included).

Contact: taln08@atala.org

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4th IEEE Tutorial and Research Workshop on PERCEPTION AND INTERACTIVE TECHNOLOGIES FOR SPEECH-BASED SYSTEMS

June 16 - 18, 2008
Kloster Irsee, Germany
The 4th IEEE Tutorial and Research Workshop on 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.
The workshop focuses on advanced speech-based human-computer interaction where various contextual factors are modelled and taken into account when users communicate with computers. This includes mechanisms, architectures, design issues, applications, evaluation and tools. Prototype and product demonstrations will be very welcome.
The workshop will bring together researchers from various disciplines such as, for example, computer science and engineering sciences, medical, psychological and neurosciences, as well as mathematics. It will provide a forum for the presentation of research and applications and for lively discussions among researchers as well as industrialists in different fields.
WORKSHOP THEMES
Papers may discuss theories, applications, evaluation, limitations, general tools and techniques. Discussion papers that critically evaluate approaches or processing strategies and prototype demonstrations are especially welcome.
- Speech recognition and semantic analysis
- Dialogue management models
- Adaptive dialogue modelling
- Recognition of emotions from speech, gestures, facial expressions and physiological data
- User modelling
- Planning and reasoning capabilities for coordination and conflict description
- Conflict resolution in complex multi-level decisions
- Multi-modality such as graphics, gesture and speech for input and output
- Fusion and information management
- Computer-supported collaborative work
- Attention selection and guidance
- Learning and adaptability
- Visual processing and recognition for advanced human-computer interaction
- Databases and corpora
- Psychophysiological evaluation and usability analysis
- Evaluation strategies and paradigms
- Prototypes and products
WORKSHOP PROGRAMME
The format of the workshop will be a non-overlapping mixture of oral and poster sessions. A number of tutorial lectures will be given by internationally recognised experts from the area of Perception and Interactive Technologies for Speech-Based Systems.
All poster sessions will be opened by an oral summary by the session chair. A number of poster sessions will be succeeded by a discussion session focussing on the subject of the session. It is our belief that this general format will ensure a lively and valuable workshop.
The organisers would like to encourage researchers and industrialists to take the opportunity to bring their applications as well as their demonstrator prototypes and design tools for demonstration to the workshop. If sufficient interest is shown, a special demonstrator/poster session will be organised and followed by a discussion session.
The official language of the workshop is English. At the opening of the workshop hardcopies of the proceedings, published in the LNCS/LNAI/LNBI Series by Springer, will be available.
TIMING AND DATES
February 10, 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 early bird registration
June 7, 2008: Final programme available on web
June 16 - 18, 2008: Workshop
Further information may be found on our workshop website.
CONTACT
Wolfgang Minker,/a>
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
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CfP IIS2008 Workshop on Spoken Language and Understanding and Dialog Systems

2nd CALL FOR PAPERS

IIS 2008 Workshop on Spoken Language Understanding  and Dialogue Systems

Zakopane, Poland    18 June 2008

 http://nlp.ipipan.waw.pl/IIS2008/luna.html

Submission deadline: 31 January 2008

The workshop is organized by the IST LUNA (http://www.ist-luna.eu/)  projects members and it is aimed to give an opportunity to share ideas on problems related to communication with computer systems in natural language and  dialogue systems.

SCOPE

The main area of interest of the workshop is human-computer interaction  in natural language and include among others:

 - spontaneous speech recognition,

- preparation of speech corpora,

- transcription problems in spoken corpora

- parsing problems in spoken texts

- semantic interpretation of text,

- knowledge representation in relation to dialogue systems,

- dialogue models,

- spoken language understanding.

SUBMISSIONS

The organizers invite long (10 pages) and short (5 pages) papers. The   papers will be refereed on the  basis of long abstracts (4 pages) by an  international committee. The final papers are to be prepared using LaTeX.   The conference proceedings in paper and electronic form will be  distributed at the conference. They will be available on-line after the conference.      I

IMPORTANT DATES

Submission deadline (abstracts)         31 January 2008

Notification of acceptance:    		  29 February 2008

Full papers, camera-ready version due:  31 March  2008

Workshop:                               18 June 2008

ORGANISERS

Malgorzata Marciniak mm@ipipan.waw.pl

Agnieszka Mykowiecka  agn@ipian.waw.pl

Krzysztof Marasek kmarasek@pjwstk.edu.pl

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JHU Summer Workshop on Language Engineering

JHU Summer Workshops
CALL FOR TEAM RESEARCH PROPOSALS
Deadline: Wednesday, October 17, 2007
The Center for Language and Speech Processing at Johns Hopkins University invites one-page research proposals for a Summer Workshop on Language Engineering, to be held in Baltimore, MD, USA, July 7 to August 14, 2008.
Workshop proposals should be suitable for a six-week team exploration, and should aim to advance the state of the art in any of the various fields of Language Engineering including speech recognition, machine translation, information retrieval, text summarization and question answering. Research topics selected for investigation by teams in previous workshops may serve as good examples for your proposal. (See http://www.clsp.jhu.edu/workshops.)
This year's workshop will be sponsored by NSF and supported in part by the newly established Human Language Technology Center of Excellence (CoE). All relevant topics of scientific interest are welcomed. Proposals can receive special priority if they contribute to one of the following long-term challenges:
* AUTOMATIC POPULATION OF A KNOWLEDGE BASE FROM TEXT: Devise and develop technology to automatically populate a large knowledge base (KB) by accumulating entities, events, and relations from vast quantities of text from various formal and informal genres in multiple languages. Devise methods to do this effectively for resource rich and/or resource poor languages. The aim is to disambiguate and normalize entities, events, and relations in such a way that the KB could represent changes over time thus reflecting text sources.
* ROBUST TECHNOLOGY FOR SPEECH: Technologies like speech-to-text, speaker identification, and language identification share a common weakness: accuracy degrades disproportionately with changes in input (microphone, genre, speaker, etc.). Seemingly small amounts of noise or diverse data sources cause machines to break where humans would quickly and effectively adapt. The aim is to develop technology whose performance would be minimally degraded by input signal variations.
* PARALLEL PROCESSING FOR SPEECH AND LANGUAGE: A broad variety of pattern recognition problems in speech and language require a large amount of computation and must be run on a large amount of data. There is a need to optimize these algorithms to increase throughput and improve cost effectiveness. Proposals are invited both for novel parallelizable algorithms and for hardware configurations that achieve higher throughput or lower speed-power product than can be achieved by optimizing either alone.
An independent panel of experts will screen all received proposals for suitability. Results of this screening will be communicated no later than October 19, 2007. Authors passing this initial screening will be invited to Baltimore to present their ideas to a peer-review panel on November 2-4, 2007. It is expected that the proposals will be revised at this meeting to address any outstanding concerns or new ideas. Two or three research topics and the teams to tackle them will be selected for the 2008 workshop.
We attempt to bring the best researchers to the workshop to collaboratively pursue the selected topics for six weeks. Authors of successful proposals typically become the team leaders. Each topic brings together a diverse team of researchers and students. The senior participants come from academia, industry and government. Graduate student participants familiar with the field are selected in accordance with their demonstrated performance, usually by the senior researchers. Undergraduate participants, selected through a national search, will be rising seniors who are new to the field and have shown outstanding academic promise.
If you are interested to participate in the 2008 Summer Workshop we ask that you submit a one-page research proposal for consideration, detailing the problem to be addressed. If your proposal passes the initial screening, we will invite you to join us for the organizational meeting in Baltimore (as our guest) for further discussions aimed at consensus. If a topic in your area of interest is chosen as one of the two or three to be pursued next summer, we expect you to be available for participation in the six-week workshop. We are not asking for an ironclad commitment at this juncture, just a good faith understanding that if a project in your area of interest is chosen, you will actively pursue it.
Proposals should be submitted via e-mail to clsp@jhu.edu by 5PM ET on Wed, October 17, 2007.

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eNTERFACE 2008 Orsay Paris

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

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Calls for EUSIPCO 2008-Lausanne Switzerland

CALL FOR PAPERS

CALL FOR SPECIAL SESSIONS AND CALL FOR TUTORIALS

EUSIPCO-2008 - 16th European Signal Processing Conference

August 25-29, 2008, Lausanne, Switzerland - http://www.eusipco2008.org

The 2008 European Signal Processing Conference (EUSIPCO-2008) is the sixteenth in a series of conferences promoted by EURASIP, the European Association for Signal, Speech, and Image Processing (www.eurasip.org ). Formerly biannual, this conference is now ayearly event. 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 special sessions and tutorials are also invited.

For the first time, access to the tutorials will be free to all registered participants!

IMPORTANT DATES:

Proposals for Special Sessions: December 7, 2007

Proposals for Tutorials: February 8, 2008

Electronic submission of Full papers (5 pages A4): February 8, 2008

Notification of Acceptance: April 30, 2008

Conference: August 25-29, 2008

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

Prof. Jean-Philippe Thiran

EPFL - Signal Processing Institute

EUSIPCO-2008 General Chair
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TDS 2008 11th Int.Conf. on Text, Speech and Dialogue

TSD 2008 - PRELIMINARY ANNOUNCEMENT

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)

Eneko Agirre, Spain

Genevieve Baudoin, France

Jan Cernocky, Czech Republic

Attila Ferencz, Romania

Alexander Gelbukh, Mexico

Louise Guthrie, GB

Jan Hajic, Czech Republic

Eva Hajicova, Czech Republic

Patrick Hanks, Czech Republic

Ludwig Hitzenberger, Germany

Jaroslava Hlavacova, Czech Republic

Ales Horak, Czech Republic

Eduard Hovy, USA

Ivan Kopecek, Czech Republic

Steven Krauwer, The Netherlands

Siegfried Kunzmann, Germany

Natalija Loukachevitch, Russia

Vaclav Matousek, Czech Republic

Hermann Ney, Germany

Elmar Noeth, Germany

Karel Oliva, Czech Republic

Karel Pala, Czech Republic

Nikola Pavesic, Slovenia

Vladimir Petkevic, Czech Republic

Fabio Pianesi, Italy

Josef Psutka, Czech Republic

James Pustejovsky, USA

Leon Rothkrantz, The Netherlands

Ernst G. Schukat-Talamazzini, Germany

Pavel Skrelin, Russia

Pavel Smrz, Czech Republic

Marko Tadic, Croatia

Tamas Varadi, Hungary

Zygmunt Vetulani, Poland

Taras Vintsiuk, Ukraine

Yorick Wilks, GB

Victor Zakharov, Russia

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).
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CfP 50th International Symposium ELMAR-2008

50th International Symposium ELMAR-2008

10-13 September 2008, Zadar, Croatia

Submission deadline: March 03, 2008

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

CALL FOR PAPERS AND SPECIAL SESSIONS

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/
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International Seminar on Speech Production ISSP 2008

The International Seminar on Speech Production (ISSP-2008) will be held in Strasbourg (Haguenau, 25 km north of Strasbourg-France) in December 2008, from Monday the 8th to Friday the 12th.

Please take note of the following important dates:

(1) Submission of a 2 page abstract (Times 12): March 28th, 2008

(2) Notification of acceptance: April 21st, 2008

(3) Early registration: 5th May, 2008

(4) Full paper submission: September 19th, 2008

(5) Late registration: September 22nd, 2008

(6) Conference dates: Monday, December 8 to Friday, December 12.

The conference website is under construction, and will de operational

before the end of this year.

We are looking forward to seeing you in Alsace.

Kind regards,

The Organizers

Rudolph Sock (IPS), Susanne Fuchs (ZAS Phonetik, Berlin) & Yves Laprie

(INRIA- LORIA, Nancy)
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ISCApad

Editor: Chris Wellekens

ISCA News

GOOGLE SCHOLAR AND ISCA ARCHIVE

SIG's activities

Courses, Internships

Motorola Labs - Center for Human Interaction Research (CHIR) l

Books, Databases, Softwares

Books

Spotlight on LDC Programmers and Software Tools

Spotlight on LDC Programmers and Software Tools -

Spotlight on LDC Programmers and Software Tools

Membership Fee Increases and Discounts

Journal on Multimodal User Interfaces

Journal on Multimodal User Interfaces

Job openings

Speech Engineer/Senior Speech Engineer at Microsoft, Mountain View, CA,USA

Speech Technology and Software Development Engineer at Microsoft Redmond WA, USA

PhD Research Studentship in Spoken Dialogue Systems- Cambridge UK

Elektrobit seeks SW-Engineers (m/f) for multimodal HMI Solutions (Speech Dialog)

Sound to Sense: 18 Fellowships in speech research

AT&T - Labs Research: Research Staff Positions - Florham Park, NJ

Research Position in Speech Processing at UGent, Belgium

Summer Inter positions at Motorola Schaumburg Illinois USA

Post-doc at France Telcom R&D Lannion Brittany France

Ph.D. Program CMU-PORTUGAL

Vitaver-Recruiting agency: Sr Manager Audio algorithm development

Nuance: Software engineer speech dialog tools

Nuance: Speech scientist London UK

Nuance: Research engineer speech engine

Nuance RESEARCH ENGINEER SPEECH DIALOG SYSTEMS:

CNRS tenure research position at LIMSI

Research Position in Speech Processing at Nagoya Institute of

PhD's fellowships at NTNU Trondheim-Norway

Two Ph.D. fellowships available at

Department of Electronics and Telecommunications

The Norwegian University of Science and Technology ( NTNU)

Trondheim, Norway

Research position in Speech Recognition in the context of Spoken Document Retrieval at the University of Twente (NL)

C/C++ Programmer Munich, Germany

Speech and Natural Language Processing Engineer at M*Modal, Pittsburgh.PA,USA

Senior Research Scientist -- Speech and Natural Lgage Processing at M*Modal, Pittsburgh, PA,USA

Journals

Papers accepted for FUTURE PUBLICATION in Speech

Special Issue on Non-Linear and Non-Conventional Speech Processing-Speech Communication

IEEE Transactions on Audio, Speech and Language Processing

Future Conferences

Future Interspeech conferences

INTERSPEECH 2008-ICSLP

INTERSPEECH 2009-EUROSPEECH

INTERSPEECH 2010-ICSLP

Future ISCA Technical and Research Workshops

ITRW Odyssey 2008

ISCA ITRW speech analysis and processing for knowledge discovery

ITRW on Evidence-based Voice and Speech Rehabilitation in Head

ISCA TR Workshop on Experimental Linguistics

Audio Visual Speech Processing Workshop (AVSP )

Robust ASR Workshop

Forthcoming events supported (but not organized) by ISCA

3rd International Conference on Large-scale Knowledge Resources (LKR 2008)

Call for Papers (Preliminary version) Speech Prosody 2008

CFP:The International Workshop on Spoken Languages Technologies for Under- The International Workshop on Spoken Languages Technologies for Under-resourced languages (SLTU)

SIGDIAL 2008 9th SIGdial Workshop on Discourse and Dialogue

LIPS 2008 Visual Speech Synthesis Challenge

Future Speech Science and Technology Events

e-Forensics 2008: The 1st International Conference on Forensic Applications and Techniques in Telecommunications, Information and Multimedia

LangTech2008

AVIOS

CfP-2nd INTERNATIONAL CONFERENCE ON LANGUAGE AND AUTOMATA THEORY AND APPLICATIONS (LATA 2008)

CfP Workshop on Empirical Approaches to Speech Rhythm

IEEE International Conference on Acoustics, Speech and Signal Processng ICASSP

CfAbstracts Prosody and expressivity in speech and music

CfP- LREC 2008 - 6th Language Resources and Evaluation Conference

Call for Papers: HLT & NLP within the Arabic world Workshop at LREC 2008

CfP ELRA Workshop on Evaluation

CALL for JEP/TALN/RECITAL 2008 - Avignon France

4th IEEE Tutorial and Research Workshop on PERCEPTION AND INTERACTIVE TECHNOLOGIES FOR SPEECH-BASED SYSTEMS

CfP IIS2008 Workshop on Spoken Language and Understanding and Dialog Systems

JHU Summer Workshop on Language Engineering

eNTERFACE 2008 Orsay Paris