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

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

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

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

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

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

Eva Hajicova - Membership Services 

 Emmanuelle Foxonet - ISCA Secretariat  for the ISCA board 

INTERSPEECH 2008 incorporating SST 08

September 22-26, 2008

Brisbane Convention & Exhibition Centre

Brisbane, Australia

http://www.interspeech2008.org/

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

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

including (but not limited to):

             Human Speech Production, Perception and Communication; 

             Speech and Language Technology; 

             Spoken Language Systems; and 

            Applications, Resources, Standardisation and Evaluation

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

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

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

   format via the conference website by 7 April 2008. 

   

  Important Dates 

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

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

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

  Tutorial Day: Monday, 22 September 2008 

  Main conference: 23-26 September 2008 

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

   

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

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

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

NEW Deadline for sending one page abstract = JUNE 9TH


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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

New Publications

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

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

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

*

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

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

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

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

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

Text definition: explains the concept denoted by the synset. 

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

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

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

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

*

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

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

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

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

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

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

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

Important dates:

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

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

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

Music and speech share numerous aspects (language, structural, acoustics, cognitive), as long in their production, that in their representation and their perception. This list has for object to warn its users, various events dealing with the study of the links between music and speech. It thus intends to connect several communities, their allowing each to take advantage of a stimulating interaction.

As a member of the speech or music community, you are invited to
subscribe to musicspeech group. The group will be moderated and
maintained by IRCAM.

Group details:
* Name: musicspeech
* Home page: http://listes.ircam.fr/wws/info/musicspeech
* Email address: musicspeech@ircam.fr

Greg Beller, IRCAM,
moderator, musicspeech list

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

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

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

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

CHIR's mission:

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

Intern profiles:

1) Acoustic environment/event detection and classification.

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

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

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

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

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

4) Integrated Voice Search Technology For Mobile Devices.

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

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

If you are interested, please send your resume to:

Dusan Macho, CHIR-Motorola Labs

Email: dusan.macho@motorola.com

Tel: +1-847-576-6762

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

SOFTWARE ENGINEER SPEECH DIALOGUE TOOLS

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

OVERVIEW:

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

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

- You will work in an international company and cooperate with people on various locations including in Europe, America and Asia. You may occasionally be asked to travel.

RESPONSIBILITIES:

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

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

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

QUALIFICATIONS:

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

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

- You have experience with Python or other scripting languages.

- GUI programming experience is a strong asset.

The following skills are a plus:

- Understanding of communication protocols

- Understanding of databases

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

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

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

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

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

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

- Knowledge of other languages is a plus.

CONTACT:

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

Deanna Roe                 Deanna.roe@nuance.com

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

ABOUT US:

Nuance is the leading provider of speech and imaging solutions for businesses and consumers around the world.  Every day, millions of users and thousands of businesses experience Nuance by calling directory assistance, requesting account information, dictating patient records, telling a navigation system their destination, or digitally reproducing documents that can be shared and searched.  With more than 3000 employees worldwide, we are committed to make the user experience more enjoyable by transforming the way people interact with information and how they create, share and use documents. Making each of those experiences productive and compelling is what Nuance is about.

 

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

To strengthen our International Professional Services team, based in London, we are currently looking for a

                            Speech Scientist, London, UK

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

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

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

Job Overview

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

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

     Required Skills

• MS or PhD in Computer Science, Engineering, Computational Linguistics, Physics, Mathematics, or related field (or equivalent)
• Strong analytical and problem solving skills and ability to troubleshoot issues
• Good judgment and quick-thinking
• Strong programming skills, preferably Perl or Python
• Excellent written and verbal communications skills
• Ability to scope work taking technical, business and time-frame constraints into consideration
• Works well in a team and in a fast-paced environment

Beneficial Skills

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

Nagoya Institute of Technology is seeking a researcher for a

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

EMIME ("Efficient multilingual interaction in mobile environment")

involving Nagoya Institute of Technology and other five European

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

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

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

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

position provides opportunities to collaborate with other researchers

in a variety of national and international projects. The competitive

salary is calculated according to qualifications based on NIT scales.

The candidate should have a strong background in speech signal

processing and some experience with speech synthesis and recognition.

Desired skills include familiarity with latest spectrum of technology

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

For more information, please contact Keiichi Tokuda

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

About us

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

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

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

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

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

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

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

outstanding, continuous contribution of developing high-performance,

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

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

continuous speech recognition engine "Julius"

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

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

involved in numerous national and international collaborative

projects. SPL also has close partnerships with many industrial

companies, in order to transfer its research into commercial

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

Funai, Asahi-Kasei, ATR.

Project summary of EMIME

The EMIME project will help to overcome the language barrier by

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

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

to produce spoken output in another language, while continuing to

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

cross-lingual spoken communication is an important, but little

explored, topic. It is essential for providing more natural

interaction and making the computing device a less obtrusive element

when assisting human-human interactions.

We will build on recent developments in speech synthesis using hidden

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

recognition. Using a common statistical modeling framework for

automatic speech recognition and speech synthesis will enable the use

of common techniques for adaptation and multilinguality.

Significant progress will be made towards a unified approach for

speech recognition and speech synthesis: this is a very powerful

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

project, we will explore the use of speaker adaptation across

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

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

characteristics when producing output speech in another language.

Our objectives are to:

1. Personalize speech processing systems by learning individual

characteristics of a user's speech and reproducing them in

synthesized speech.

2. Introduce a cross-lingual capability such that personal

characteristics can be reproduced in a second language not spoken

by the user.

3. Develop and better understand the mathematical and theoretical

relationship between speech recognition and synthesis.

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

cross-lingual personalization.

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

practical mobile application.

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

We offer   

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

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

- the possibility to participate in international scientific    conferences.   

- a permanent job in the center of Munich.  

- excellent possibilities for development within our fast    growing company.    

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

We expect  

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

-experience with parallel algorithms and thread    programming.  

-experience with object-oriented design of software    systems.  

-good knowledge of English or German.   

Desirable is  

-experience with optimization of algorithms.  

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

-experience with Delphi or Turbo Pascal.   

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

digital publishing AG  

Freddy Ertl  f.ertl@digitalpublishing.de  

Tumblinger Straße 32  

D-80337 München Germany

Motorola Labs - Center for Human Interaction Research (CHIR) 

located in Schaumburg Illinois, USA, 

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

CHIR's mission

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

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

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

capabilities and to continually evolving knowledge about the user.

Intern profiles

1) Acoustic environment/event detection and classification. 

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

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

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

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

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

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

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

adaptation is highly preferred.

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

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

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

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

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

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

Development experience on Windows CE/Mobile desired.

4) Integrated Voice Search Technology For Mobile Devices

Candidate should be proficient in information retrie