ISCA - International Speech
Communication Association


ISCApad Archive  »  2019  »  ISCApad #249  »  Resources

ISCApad #249

Monday, March 11, 2019 by Chris Wellekens

5 Resources
5-1 Books
5-1-1Pejman Mowlaee et al., 'Phase-Aware Signal Processing in Speech Communication: Theory and Practice', Wiley 2016

Phase-Aware Signal Processing in Speech Communication: Theory and Practice

Pejman Mowlaee, Johannes Stahl, Josef Kulmer, Florian Mayer

http://eu.wiley.com/WileyCDA/WileyTitle/productCd-1119238811.html

An overview on the challenging new topic of phase-aware signal processing

Speech communication technology is a key factor in human-machine interaction, digital hearing aids, mobile telephony, and automatic speech/speaker recognition. With the proliferation of these applications, there is a growing requirement for advanced methodologies that can push the limits of the conventional solutions relying on processing the signal magnitude spectrum.

Single-Channel Phase-Aware Signal Processing in Speech Communication provides a comprehensive guide to phase signal processing and reviews the history of phase importance in the literature, basic problems in phase processing, fundamentals of phase estimation together with several applications to demonstrate the usefulness of phase processing.

Key features:

  • Analysis of recent advances demonstrating the positive impact of phase-based processing in pushing the limits of conventional methods.
  • Offers unique coverage of the historical context, fundamentals of phase processing and provides several examples in speech communication.
  • Provides a detailed review of many references and discusses the existing signal processing techniques required to deal with phase information in different applications involved with speech.
  • The book supplies various examples and MATLAB® implementations delivered within the PhaseLab toolbox.

Single-Channel Phase-Aware Signal Processing in Speech Communication is a valuable single-source for students, non-expert DSP engineers, academics and graduate students.

ejman Mowlaee, Johannes Stahl, Josef Kulmer, Florian Mayer
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5-1-2Jean Caelen, Anne Xuereb, 'Dialogue : altérité, interaction, énaction'

 

Jean Caelen,Anne Xuereb

Dialogue : altérité, interaction, énaction

Editions universitaires européennes

 

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5-1-3Bäckström, Tom (with Guillaume Fuchs, Sascha Disch, Christian Uhle and Jeremie Lecomte), 'Speech Coding with Code-Excited Linear Prediction', Springer


 Speech Coding with Code-Excited Linear Prediction

Author: Bäckström, Tom

Invited chapters from: Guillaume Fuchs, Sascha Disch, Christian Uhle and Jeremie Lecomte

Publisher: Springer

http://www.springer.com/gp/book/9783319502021

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5-1-4Shinji Watanabe, Marc Delcroix, Florian Metze, John R. Hershey (Eds), 'New Era for Robust Seech Recognition', Springer.

Shinji Watanabe, Marc Delcroix, Florian Metze, John R. Hershey (Eds), 'New Era for Robust Seech Recognition', Springer.

https://link.springer.com/book/10.1007%2F978-3-319-64680-0

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5-1-5Fabrice Marsac, Rudolph Sock, CONSÉCUTIVITÉ ET SIMULTANÉITÉ en Linguistique, Langues et Parole, L'Harmattan,France

Nous avons le plaisir de vous annoncer la parution du volume thématique « CONSÉCUTIVITÉ ET SIMULTANÉITÉ en Linguistique, Langues et Parole » dans la Collection Dixit Grammatica (L’Harmattan, France) :
 
- CONSÉCUTIVITÉ ET SIMULTANÉITÉ en Linguistique, Langues et Parole – 1. Phonétique, Phonologie (Sous la direction de Camille Fauth, Jean-Paul Meyer, Fabrice Marsac & Rudolph Sock) • ISBN : 978-2-343-14277-7 • 5 mars 2018 • 172 pages http://www.editionsharmattan.fr/index.asp?navig=catalogue&obj=livre&no=59200&razSqlClone=1
 
- CONSÉCUTIVITÉ ET SIMULTANÉITÉ en Linguistique, Langues et Parole – 2. Syntaxe, Sémantique (Sous la direction de Angelina Aleksandrova, Céline Benninger, Anne Theissen, Fabrice Marsac & Jean-Paul Meyer) • ISBN : 978-2-343-14278-4 • 5 mars 2018 • 300 pages http://www.editionsharmattan.fr/index.asp?navig=catalogue&obj=livre&no=59201&razSqlClone=1
 
- CONSÉCUTIVITÉ ET SIMULTANÉITÉ en Linguistique, Langues et Parole – 3. Didactique, Traductologie-Interprétation (Sous la direction de Jean-Paul Meyer, Mária Pal'ová & Fabrice Marsac) • ISBN : 978-2-343-14279-1 • 5 mars 2018 • 200 pages http://www.editionsharmattan.fr/index.asp?navig=catalogue&obj=livre&no=59202&razSqlClone=1
 
Cet ouvrage collectif, qui comprend trois tomes complémentaires, rassemble des études constituant les traces écrites de communications prononcées lors du colloque international éponyme s’étant tenu à l’Université de Strasbourg (France) en juillet 2015. Les tomes renferment des travaux originaux et novateurs traitant de la dynamicité complexe du couple consécutivité-simultanéité saisi dans le domaine des Sciences du Langage. Le contenu, délibérément interdisciplinaire, concerne non seulement l’ensemble des disciplines relatives aux Sciences du langage mais aussi d’autres disciplines scientifiques, connexes mais préoccupées par des problématiques résolument linguistiques. Les éditeurs de ce volume thématique espèrent que les divers points de vue linguistiques ainsi adoptés livreront aux lecteurs un état des connaissances actualisé relativement aux différentes problématiques traitées. Il va sans dire, par ailleurs, que les auteurs comme les éditeurs apprécieront tout retour constructif de la part des lecteurs.
 
 
Fabrice Marsac et Rudolph Sock Directeurs de Dixit Grammatica


 

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5-1-6Emmanuel Vincent (Editor), Tuomas Virtanen (Editor), Sharon Gannot (Editor), 'Audio Source Separation and Speech Enhancement', Wiley

 Emmanuel Vincent (Editor), Tuomas Virtanen (Editor), Sharon Gannot (Editor),

Audio Source Separation and Speech Enhancement:


https://www.wiley.com/en-us/Audio+Source+Separation+and+Speech+Enhancement-p-9781119279891

ISBN: 978-1-119-27989-1

October 2018

504 pages



This 500-page book provides a unifying view of source separation and enhancement,
including but not limited to array processing, matrix factorization, and deep learning
based methods, and speech and music applications, with consistent notation and
terminology across all chapters.

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5-1-7Jen-Tzung Chien, 'Source Separation and Machine Learning', Academic Press

Jen-Tzung Chien, 'Source Separation and Machine Learning', Academic Press

Source Separation and Machine Learning presents the fundamentals in adaptive learning
algorithms for Blind Source Separation (BSS) and emphasizes the importance of machine
learning perspectives. It illustrates how BSS problems are tackled through adaptive
learning algorithms and model-based approaches using the latest information on mixture
signals to build a BSS model that is seen as a statistical model for a whole system.
Looking at different models, including independent component analysis (ICA), nonnegative
matrix factorization (NMF), nonnegative tensor factorization (NTF), and deep neural
network (DNN), the book addresses how they have evolved to deal with multichannel and
singlechannel source separation.

Key features:
? Emphasizes the modern model-based Blind Source Separation (BSS) which closely connects
the latest research topics of BSS and Machine Learning
? Includes coverage of Bayesian learning, sparse learning, online learning,
discriminative learning and deep learning
? Presents a number of case studies of model-based BSS, using a variety of learning
algorithms that provide solutions for the construction of BSS systems

https://www.elsevier.com/books/source-separation-and-machine-learning/chien/978-0-12-804566-4

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5-1-8Ingo Feldhausen, « Methods in prosody: A Romance language perspective », Language Science Press (open access)

Nous sommes heureux de vous annoncer la parution d'un recueil validé par un comité de lecture et consacré aux méthodes de recherche en prosodie. Cet ouvrage est intitulé « Methods in prosody: A Romance language perspective ».

Il est publié par Language Science Press, une maison d’édition open access. Le livre peut-être téléchargé gratuitement en cliquant sur le lien suivant :

http://langsci-press.org/catalog/book/183

La table des matières est la suivante :

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

Introduction
Ingo Feldhausen, Jan Fliessbach & Maria del Mar Vanrell                                                                   iii

Foreword
Pilar Prieto                                                                                                                                              vii

I Large corpora and spontaneous speech

1) Using large corpora and computational tools to describe prosody: An
exciting challenge for the future with some (important) pending problems to solve

Juan María Garrido Almiñana                                                                                                                  3

2) Intonation of pronominal subjects in Porteño Spanish: Analysis of 
spontaneous speech

Andrea Pešková                                                                                                                                     45

II Approaches to prosodic analysis

3) Multimodal analyses of audio-visual information: Some methods and
issues in prosody research

Barbara Gili Fivela                                                                                                                                 83

4) The realizational coefficient: Devising a method for empirically
determining prominent positions in Conchucos Quechua

Timo Buchholz & Uli Reich                                                                                                                 123

5) On the role of prosody in disambiguating wh-exclamatives and
wh-interrogatives in Cosenza Italian

Olga Kellert, Daniele Panizza & Caterina Petrone                                                                               165

III Elicitation methods

6) The Discourse Completion Task in Romance prosody research: Status
quo and outlook

Maria del Mar Vanrell, Ingo Feldhausen & Lluïsa Astruc                                                                    191

7) Describing the intonation of speech acts in Brazilian Portuguese:
Methodological aspects

João Antônio de Moraes & Albert Rilliard                                                                                           229

Indexes                                                                                                                                                  263

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

N'hésitez pas à diffuser la parution de cet ouvrage auprès de vos collègues qui pourraient s'y intéresser.

Bien cordialement,

Ingo Feldhausen
(Co-coordinateur d'ouvrage)

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5-2 Database
5-2-1Linguistic Data Consortium (LDC) update (February 2019)

 February 2019 Newsletter

 In this newsletter:

 

Only two weeks left to enjoy 2019 membership discounts

 

Spring 2019 LDC Data Scholarship recipients

 

LDC’s new language game

 

New publications:

DEFT Chinese Committed Belief Annotation

IARPA Babel Lithuanian Language Pack IARPA-babel304b-v1.0b

Multi-Language Conversational Telephone Speech 2011 -- Arabic Group

Multilingual ATIS

_____________________________________________________________________________

 

Only two weeks left to enjoy 2019 membership discounts

There is still time to save on 2019 membership fees. Through March 1, all organizations receive a discount on the 2019 membership fee (up to 10%) when they choose to join or renew. For more information on membership benefits, visit Join LDC.

 

Spring 2019 LDC Data Scholarship recipients

Congratulations to the recipients of LDC's Spring 2019 Data Scholarships:

Colin Annand: University of Cincinnati (USA); PhD. Psychology. Colin is awarded a copy of Switchboard-1 Release 2 for his research involving the relationship between speech patterns and conversation content.

Si Chen: Huazhong University of Science and Technology (China); B.S. Communication Engineering. Si is awarded a copy of ACE 2005 Multilingual Training Corpus for his work on event extraction.

Noor-e-Hira: Fatima Jinnah Women University (Pakistan); MSc. Computer Sciences. Noor is awarded a copy of NIST 2008 Open Machine Translation (OpenMT) Evaluation for her research in machine translation.

Matthew Roddy: Trinity College Dublin (Ireland); Ph.D. Electrical Engineering. Matthew is awarded copies of 2000 HUB5 English Evaluation Speech and Transcripts for his work in spoken dialogue systems.

Ammara Zafar: Fatima Jinnah Women University (Pakistan); MSc Computer Sciences. Ammara awarded a copy of NIST 2009 Open Machine Translation (OpenMT) Evaluation for her research in machine translation.

For information about the program, visit the Data Scholarship page.

LDC’s new language game

LDC’s new language game, NameThatLanguage, tests your skill at recognizing the language spoken in short audio clips. The game includes thousands of clips to prevent memorization and offers a real challenge that increases as you progress. In addition to being fun, the game provides useful data on language confusability and linguistic diversity. Game results will be shared freely for research. New clips and more languages continue to be added providing ongoing challenges and new research data. Help support language research by playing! https://namethatlanguage.org

_____________________________________________________________________________

 

New publications:

 

(1) DEFT Chinese Committed Belief Annotation was developed by LDC and consists of approximately 83,000 tokens of Chinese discussion forum text annotated for 'committed belief,' which marks the level of commitment displayed by the author to the truth of the propositions expressed in the text.

 

DARPA's Deep Exploration and Filtering of Text (DEFT) program aimed to address remaining capability gaps in state-of-the-art natural language processing technologies related to inference, causal relationships, and anomaly detection. LDC supported the DEFT program by collecting, creating, and annotating a variety of data sources.

 

DEFT Chinese Committed Belief Annotation is distributed via web download.

 

2019 Subscription Members will automatically receive copies of this corpus. 2019 Standard Members may request a copy as part of their 16 free membership corpora. Non-members may license this data for $1000. 

 

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(2) IARPA Babel Lithuanian Language Pack IARPA-babel304b-v1.0b was developed by Appen for the IARPA (Intelligence Advanced Research Projects Activity) Babel program. It contains approximately 210 hours of Lithuanian conversational and scripted telephone speech collected in 2013 and 2014 along with corresponding transcripts.

 

The Lithuanian speech in this release represents that spoken in the Auk?taitian and Samogitian dialect regions of Lithuania. The gender distribution among speakers is approximately equal; speakers' ages range from 16 years to 71 years. Calls were made using different telephones (e.g., mobile, landline) from a variety of environments including the street, a home or office, a public place, and inside a vehicle.

 

IARPA Babel Lithuanian Language Pack IARPA-babel304b-v1.0b is distributed via web download.

 

2019 Subscription Members will receive copies of this corpus provided they have submitted a completed copy of the special license agreement. 2019 Standard Members may request a copy as part of their 16 free membership corpora. Non-members may license this data for $25.

 

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(3) Multi-Language Conversational Telephone Speech 2011 -- Arabic Group was developed by LDC and is comprised of approximately 117 hours of telephone speech in distinct dialects of colloquial Arabic: Iraqi, Levantine and Maghrebi.

 

The data were collected primarily to support research and technology evaluation in automatic language identification, and portions of these telephone calls were used in the NIST 2011 Language Recognition Evaluation (LRE). LRE 2011 focused on language pair discrimination for 24 languages/dialects, some of which could be considered mutually intelligible or closely related.

 

Multi-Language Conversational Telephone Speech 2011 -- Arabic Group is distributed via web download.

 

2019 Subscription Members will automatically receive copies of this corpus. 2019 Standard Members may request a copy as part of their 16 free membership corpora. Non-members may license this data for $2500.

 

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(4) Multilingual ATIS was developed by Google Inc. and consists of 5,871 utterances from ATIS2 (LDC93S5), ATIS3 Training Data (LDC94S19), and ATIS3 Test Data (LDC95S26) annotated and translated into Hindi and Turkish.

 

The ATIS (Air Travel Information Services) collection was developed to support the research and development of speech understanding systems. Participants were presented with various hypothetical travel planning scenarios and asked to solve them by interacting with partially or completely automated ATIS systems. The resulting utterances were recorded and transcribed. Data was collected in the early 1990s at five US sites: Raytheon BBN, Carnegie Mellon University, MIT Laboratory for Computer Science, National Institute for Standards and Technology, and SRI International.

 

The original English utterances were manually translated into Hindi and Turkish. This release also includes the original English utterance and the machine translation back into English of the manual target language utterance translation. Each utterance is annotated with named entities via table lookup; markers include city, airline, airport names, and dates.

 

Multilingual ATIS is distributed via web download.

 

2019 Subscription Members will automatically receive copies of this corpus. 2019 Standard Members may request a copy as part of their 16 free membership corpora. Non-members may license this data at no cost. 

 

 

Membership Office

Linguistic Data Consortium

University of Pennsylvania

T: +1-215-573-1275

E: ldc@ldc.upenn.edu

M: 3600 Market St. Suite 810

      Philadelphia, PA 19104

 

 

 

 



 

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5-2-2ELRA - Language Resources Catalogue - Update (October 2018)
ELRA - Language Resources Catalogue - Update
-------------------------------------------------------
We are happy to announce that 2 new Written Corpora and 4 new Speech resources are now available in our catalogue.

ELRA-W0126 Training and test data for Arabizi detection and transliteration
ISLRN: 986-364-744-303-9
The dataset is composed of : a collection of mixed English and Arabizi text intended to train and test a system for the automatic detection of code-switching in mixed English and Arabizi texts ; and a set of 3,452 Arabizi tokens manually transliterated into Arabic, intended to train and test a system that performs Arabizi to Arabic transliteration.
For more information, see: http://catalog.elra.info/en-us/repository/browse/ELRA-W0126/

ELRA-W0127 Normalized Arabic Fragments for Inestimable Stemming (NAFIS)
ISLRN: 305-450-745-774-1
This is an Arabic stemming gold standard corpus composed by a collection of 37 sentences, selected to be representative of Arabic stemming tasks and manually annotated. Compiled sentences belong to various sources (poems, holy Quran, books, and periodics) of diversified kinds (proverb and dictum, article commentary, religious text, literature, historical fiction). NAFIS is represented according to the TEI standard.   
For more information, see: http://catalog.elra.info/en-us/repository/browse/ELRA-W0127/

ELRA-S0396 Mbochi speech corpus

ISLRN: 747-055-093-447-8
This corpus consists of 5131 sentences recorded in Mbochi, together with their transcription and French translation, as well as the results from the work made during  JSALT workshop: alignments at the phonetic level and various results of unsupervised word segmentation from audio. The audio corpus is made up of 4,5 hours, downsampled at 16kHz, 16bits, with Linear PCM encoding. Data is distributed into 2 parts, one for training consisting of 4617 sentences, and one for development consisting of 514 sentences.
For more information, see: http://catalog.elra.info/en-us/repository/browse/ELRA-S0396/

ELRA-S0397 Chinese Mandarin (South) database

ISLRN: 503-886-852-083-2
This database contains the recordings of 1000 Chinese Mandarin speakers from Southern China (500 males and 500 females), from 18 to 60 years? old, recorded in quiet studios. Recordings were made through microphone headsets and consist of 341 hours of audio data (about 30 minutes per speaker), stored in .WAV files as sequences of 48 KHz Mono, 16 bits, Linear PCM.
For more information, see: http://catalog.elra.info/en-us/repository/browse/ELRA-S0397/

ELRA-S0398 Chinese Mandarin (North) database
ISLRN: 353-548-770-894-7
This database contains the recordings of 500 Chinese Mandarin speakers from Northern China (250 males and 250 females), from 18 to 60 years? old, recorded in quiet studios. Recordings were made through microphone headsets and consist of 172 hours of audio data (about 30 minutes per speaker), stored in .WAV files as sequences of 48 KHz Mono, 16 bits, Linear PCM.
For more information, see: http://catalog.elra.info/en-us/repository/browse/ELRA-S0398/

ELRA-S0401 Persian Audio Dictionary
ISLRN: 133-181-128-420-9
This dictionary consists of more than 50,000 entries (along with almost all wordforms and proper names) with corresponding audio files in MP3 and English transliterations. The words have been recorded with standard Persian (Farsi) pronunciation (all by a single speaker). This dictionary is provided with its software.
For more information, see: http://catalog.elra.info/en-us/repository/browse/ELRA-S0401/

For more information on the catalogue, please contact Valérie Mapelli mailto:mapelli@elda.org

If you would like to enquire about having your resources distributed by ELRA, please do not hesitate to contact us.

Visit our On-line Catalogue: http://catalog.elra.info
Visit the Universal Catalogue: http://universal.elra.info
Archives of ELRA Language Resources Catalogue Updates: http://www.elra.info/en/catalogues/language-resources-announcements/












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5-2-3Speechocean – update (March 2019)

 

 

 

 

Accented English Speech Recognition Corpus --- Speechocean

 

 

Speechocean: The World’s Leading A.I. Data Resource & Service Supplier

 

At present, we are capable to provide data services with 110+ languages and dialects across the world. For more detailed information, please visit our website:

http://kingline.speechocean.com

 

Language

Speakers

Total Hours

Chinese English

3478

1513

Hong Kong English

200

378

Japanese English

1005

902

Korean English

116

207

Singaporean English

203

366

Indonesian English

804

378

Indian English

2394

3540

French English

130

191

German English

200

314

Italian English

120

189

Portuguese English

100

154

Spanish English

200

326

 

 

More Information

 

  • Information of Speaker: Selected non-native English speakers. Balanced covering ages, gender and regional accents.

  • Recording Environment: Quiet or noisy environment.

  • Recording Platform:Desktop or mobile

  • Recording Content: Sentences, conversation, words…

  • Post Processing: Proofreading, transcription, annotation and quality control.

  • Lexicon: Included

 

Contact Us:

Email: contact@speechocean.com

 

 

 

 


 


 

 

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5-2-4Google 's Language Model benchmark
 Here is a brief description of the project.

'The purpose of the project is to make available a standard training and test setup for language modeling experiments.

The training/held-out data was produced from a download at statmt.org using a combination of Bash shell and Perl scripts distributed here.

This also means that your results on this data set are reproducible by the research community at large.

Besides the scripts needed to rebuild the training/held-out data, it also makes available log-probability values for each word in each of ten held-out data sets, for each of the following baseline models:

  • unpruned Katz (1.1B n-grams),
  • pruned Katz (~15M n-grams),
  • unpruned Interpolated Kneser-Ney (1.1B n-grams),
  • pruned Interpolated Kneser-Ney (~15M n-grams)

 

Happy benchmarking!'

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5-2-5Forensic database of voice recordings of 500+ Australian English speakers

Forensic database of voice recordings of 500+ Australian English speakers

We are pleased to announce that the forensic database of voice recordings of 500+ Australian English speakers is now published.

The database was collected by the Forensic Voice Comparison Laboratory, School of Electrical Engineering & Telecommunications, University of New South Wales as part of the Australian Research Council funded Linkage Project on making demonstrably valid and reliable forensic voice comparison a practical everyday reality in Australia. The project was conducted in partnership with: Australian Federal Police,  New South Wales Police,  Queensland Police, National Institute of Forensic Sciences, Australasian Speech Sciences and Technology Association, Guardia Civil, Universidad Autónoma de Madrid.

The database includes multiple non-contemporaneous recordings of most speakers. Each speaker is recorded in three different speaking styles representative of some common styles found in forensic casework. Recordings are recorded under high-quality conditions and extraneous noises and crosstalk have been manually removed. The high-quality audio can be processed to reflect recording conditions found in forensic casework.

The database can be accessed at: http://databases.forensic-voice-comparison.net/

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5-2-6Audio and Electroglottographic speech recordings

 

Audio and Electroglottographic speech recordings from several languages

We are happy to announce the public availability of speech recordings made as part of the UCLA project 'Production and Perception of Linguistic Voice Quality'.

http://www.phonetics.ucla.edu/voiceproject/voice.html

Audio and EGG recordings are available for Bo, Gujarati, Hmong, Mandarin, Black Miao, Southern Yi, Santiago Matatlan/ San Juan Guelavia Zapotec; audio recordings (no EGG) are available for English and Mandarin. Recordings of Jalapa Mazatec extracted from the UCLA Phonetic Archive are also posted. All recordings are accompanied by explanatory notes and wordlists, and most are accompanied by Praat textgrids that locate target segments of interest to our project.

Analysis software developed as part of the project – VoiceSauce for audio analysis and EggWorks for EGG analysis – and all project publications are also available from this site. All preliminary analyses of the recordings using these tools (i.e. acoustic and EGG parameter values extracted from the recordings) are posted on the site in large data spreadsheets.

All of these materials are made freely available under a Creative Commons Attribution-NonCommercial-ShareAlike-3.0 Unported License.

This project was funded by NSF grant BCS-0720304 to Pat Keating, Abeer Alwan and Jody Kreiman of UCLA, and Christina Esposito of Macalester College.

Pat Keating (UCLA)

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5-2-7EEG-face tracking- audio 24 GB data set Kara One, Toronto, Canada

We are making 24 GB of a new dataset, called Kara One, freely available. This database combines 3 modalities (EEG, face tracking, and audio) during imagined and articulated speech using phonologically-relevant phonemic and single-word prompts. It is the result of a collaboration between the Toronto Rehabilitation Institute (in the University Health Network) and the Department of Computer Science at the University of Toronto.

 

In the associated paper (abstract below), we show how to accurately classify imagined phonological categories solely from EEG data. Specifically, we obtain up to 90% accuracy in classifying imagined consonants from imagined vowels and up to 95% accuracy in classifying stimulus from active imagination states using advanced deep-belief networks.

 

Data from 14 participants are available here: http://www.cs.toronto.edu/~complingweb/data/karaOne/karaOne.html.

 

If you have any questions, please contact Frank Rudzicz at frank@cs.toronto.edu.

 

Best regards,

Frank

 

 

PAPER Shunan Zhao and Frank Rudzicz (2015) Classifying phonological categories in imagined and articulated speech. In Proceedings of ICASSP 2015, Brisbane Australia

ABSTRACT This paper presents a new dataset combining 3 modalities (EEG, facial, and audio) during imagined and vocalized phonemic and single-word prompts. We pre-process the EEG data, compute features for all 3 modalities, and perform binary classi?cation of phonological categories using a combination of these modalities. For example, a deep-belief network obtains accuracies over 90% on identifying consonants, which is signi?cantly more accurate than two baseline supportvectormachines. Wealsoclassifybetweenthedifferent states (resting, stimuli, active thinking) of the recording, achievingaccuraciesof95%. Thesedatamaybeusedtolearn multimodal relationships, and to develop silent-speech and brain-computer interfaces.

 

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5-2-8TORGO data base free for academic use.

In the spirit of the season, I would like to announce the immediate availability of the TORGO database free, in perpetuity for academic use. This database combines acoustics and electromagnetic articulography from 8 individuals with speech disorders and 7 without, and totals over 18 GB. These data can be used for multimodal models (e.g., for acoustic-articulatory inversion), models of pathology, and augmented speech recognition, for example. More information (and the database itself) can be found here: http://www.cs.toronto.edu/~complingweb/data/TORGO/torgo.html.

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5-2-9Datatang

Datatang is a global leading data provider that specialized in data customized solution, focusing in variety speech, image, and text data collection, annotation, crowdsourcing services.

 

Summary of the new datasets (2018) and a brief plan for 2019.

 

 

 

? Speech data (with annotation) that we finished in 2018 

 

Language
Datasets Length
  ( Hours )
French
794
British English
800
Spanish
435
Italian
1,440
German
1,800
Spanish (Mexico/Colombia)
700
Brazilian Portuguese
1,000
European Portuguese
1,000
Russian
1,000

 

?2019 ongoing  speech project 

 

Type

Project Name

Europeans speak English

1000 Hours-Spanish Speak English

1000 Hours-French Speak English

1000 Hours-German Speak English

Call Center Speech

1000 Hours-Call Center Speech

off-the-shelf data expansion

1000 Hours-Chinese Speak English

1500 Hours-Mixed Chinese and English Speech Data

 

 

 

On top of the above,  there are more planed speech data collections, such as Japanese speech data, children`s speech data, dialect speech data and so on.  

 

What is more, we will continually provide those data at a competitive price with a maintained high accuracy rate.

 

 

 

If you have any questions or need more details, do not hesitate to contact us jessy@datatang.com 

 

It would be possible to send you with a sample or specification of the data.

 

 

 


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5-2-10Fearless Steps Corpus (University of Texas, Dallas)

Fearless Steps Corpus

John H.L. Hansen, Abhijeet Sangwan, Lakshmish Kaushik, Chengzhu Yu Center for Robust Speech Systems (CRSS), Eric Jonsson School of Engineering, The University of Texas at Dallas (UTD), Richardson, Texas, U.S.A.


NASA’s Apollo program is a great achievement of mankind in the 20th century. CRSS, UT-Dallas has undertaken an enormous Apollo data digitization initiative where we proposed to digitize Apollo mission speech data (~100,000 hours) and develop Spoken Language Technology based algorithms to analyze and understand various aspects of conversational speech. Towards achieving this goal, a new 30 track analog audio decoder is designed to decode 30 track Apollo analog tapes and is mounted on to the NASA Soundscriber analog audio decoder (in place of single channel decoder). Using the new decoder all 30 channels of data can be decoded simultaneously thereby reducing the digitization time significantly. 
We have digitized 19,000 hours of data from Apollo missions (including entire Apollo-11, most of Apollo-13, Apollo-1, and Gemini-8 missions). This audio archive is named as “Fearless Steps Corpus”. This is one of the most unique and singularly large naturalistic audio corpus of such magnitude. Automated transcripts are generated by building Apollo mission specific custom Deep Neural Networks (DNN) based Automatic Speech Recognition (ASR) system along with Apollo mission specific language models. Speaker Identification System (SID) to identify the speakers are designed. A complete diarization pipeline is established to study and develop various SLT tasks. 
We will release this corpus for public usage as a part of public outreach and promote SLT community to utilize this opportunity to build naturalistic spoken language technology systems. The data provides ample opportunity setup challenging tasks in various SLT areas. As a part of this outreach we will be setting “Fearless Challenge” in the upcoming INTERSPEECH 2018. We will define and propose 5 tasks as a part of this challenge. The guidelines and challenge data will be released in the Spring 2018 and will be available for download for free. The five challenges are, (1) Automatic Speech Recognition (2) Speaker Identification (3) Speech Activity Detection (4) Speaker Diarization (5) Keyword spotting and Joint Topic/Sentiment detection.
Looking forward for your participation (John.Hansen@utdallas.edu) 

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5-2-11SIWIS French Speech Synthesis Database
The SIWIS French Speech Synthesis Database includes high quality French speech recordings and associated text files, aimed at building TTS systems, investigate multiple styles, and emphasis. A total of 9750 utterances from various sources such as parliament debates and novels were uttered by a professional French voice talent. A subset of the database contains emphasised words in many different contexts. The database includes more than ten hours of speech data and is freely available.
 
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5-2-12JLCorpus - Emotional Speech corpus with primary and secondary emotions
JLCorpus - Emotional Speech corpus with primary and secondary emotions:
 

For further understanding the wide array of emotions embedded in human speech, we are introducing an emotional speech corpus. In contrast to the existing speech corpora, this corpus was constructed by maintaining an equal distribution of 4 long vowels in New Zealand English. This balance is to facilitate emotion related formant and glottal source feature comparison studies. Also, the corpus has 5 secondary emotions along with 5 primary emotions. Secondary emotions are important in Human-Robot Interaction (HRI), where the aim is to model natural conversations among humans and robots. But there are very few existing speech resources to study these emotions,and this work adds a speech corpus containing some secondary emotions.

Please use the corpus for emotional speech related studies. When you use it please include the citation as:

Jesin James, Li Tian, Catherine Watson, 'An Open Source Emotional Speech Corpus for Human Robot Interaction Applications', in Proc. Interspeech, 2018.

To access the whole corpus including the recording supporting files, click the following link: https://www.kaggle.com/tli725/jl-corpus, (if you have already installed the Kaggle API, you can type the following command to download: kaggle datasets download -d tli725/jl-corpus)

Or if you simply want the raw audio+txt files, click the following link: https://www.kaggle.com/tli725/jl-corpus/downloads/Raw%20JL%20corpus%20(unchecked%20and%20unannotated).rar/4

The corpus was evaluated by a large scale human perception test with 120 participants. The link to the survey are here- For Primary emorion corpus: https://auckland.au1.qualtrics.com/jfe/form/SV_8ewmOCgOFCHpAj3

For Secondary emotion corpus: https://auckland.au1.qualtrics.com/jfe/form/SV_eVDINp8WkKpsPsh

These surveys will give an overall idea about the type of recordings in the corpus.

The perceptually verified and annotated JL corpus will be given public access soon.

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5-3 Software
5-3-1Release of the version 2 of FASST (Flexible Audio Source Separation Toolbox).
Release of the version 2 of FASST (Flexible Audio Source Separation Toolbox). http://bass-db.gforge.inria.fr/fasst/ This toolbox is intended to speed up the conception and to automate the implementation of new model-based audio source separation algorithms. It has the following additions compared to version 1: * Core in C++ * User scripts in MATLAB or python * Speedup * Multichannel audio input We provide 2 examples: 1. two-channel instantaneous NMF 2. real-world speech enhancement (2nd CHiME Challenge, Track 1)
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5-3-2Cantor Digitalis, an open-source real-time singing synthesizer controlled by hand gestures.

We are glad to announce the public realease of the Cantor Digitalis, an open-source real-time singing synthesizer controlled by hand gestures.


It can be used e.g. for making music or for singing voice pedagogy.

A wide variety of voices are available, from the classic vocal quartet (soprano, alto, tenor, bass), to the extreme colors of childish, breathy, roaring, etc. voices.  All the features of vocal sounds are entirely under control, as the synthesis method is based on a mathematic model of voice production, without prerecording segments.

The instrument is controlled using chironomy, i.e. hand gestures, with the help of interfaces like stylus or fingers on a graphic tablet, or computer mouse. Vocal dimensions such as the melody, vocal effort, vowel, voice tension, vocal tract size, breathiness etc. can easily and continuously be controlled during performance, and special voices can be prepared in advance or using presets.

Check out the capabilities of Cantor Digitalis, through performances extracts from the ensemble Chorus Digitalis:
http://youtu.be/_LTjM3Lihis?t=13s.

In pratice, this release provides:
  • the synthesizer application
  • the source code in the form of a Max package (GPL-like license)
  • a documentation for the musician and another for the developper
What do you need ?
  • a Mac OSX
  • ideally a Wacom graphic tablet, but it also works with your computer mouse
  • for the developers, the Max software
Interested ?
  • To download the Cantor Digitalis, click here
  • To subscribe to the Cantor Digitalisnewsletter and/or the forum list, or to contact the developers, click here
  • To learn about the Chorus Digitalis, ensemble of Cantor Digitalisand watch videos of performances, click here
  • For more details about the Cantor Digitalis, click here
 
Regards,
 
The Cantor Digitalis team (who loves feedback — cantordigitalis@limsi.fr)
Christophe d'Alessandro, Lionel Feugère, Olivier Perrotin
http://cantordigitalis.limsi.fr/
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5-3-3MultiVec: a Multilingual and MultiLevel Representation Learning Toolkit for NLP

 

We are happy to announce the release of our new toolkit “MultiVec” for computing continuous representations for text at different granularity levels (word-level or sequences of words). MultiVec includes Mikolov et al. [2013b]’s word2vec features, Le and Mikolov [2014]’s paragraph vector (batch and online) and Luong et al. [2015]’s model for bilingual distributed representations. MultiVec also includes different distance measures between words and sequences of words. The toolkit is written in C++ and is aimed at being fast (in the same order of magnitude as word2vec), easy to use, and easy to extend. It has been evaluated on several NLP tasks: the analogical reasoning task, sentiment analysis, and crosslingual document classification. The toolkit also includes C++ and Python libraries, that you can use to query bilingual and monolingual models.

 

The project is fully open to future contributions. The code is provided on the project webpage (https://github.com/eske/multivec) with installation instructions and command-line usage examples.

 

When you use this toolkit, please cite:

 

@InProceedings{MultiVecLREC2016,

Title                    = {{MultiVec: a Multilingual and MultiLevel Representation Learning Toolkit for NLP}},

Author                   = {Alexandre Bérard and Christophe Servan and Olivier Pietquin and Laurent Besacier},

Booktitle                = {The 10th edition of the Language Resources and Evaluation Conference (LREC 2016)},

Year                     = {2016},

Month                    = {May}

}

 

The paper is available here: https://github.com/eske/multivec/raw/master/docs/Berard_and_al-MultiVec_a_Multilingual_and_Multilevel_Representation_Learning_Toolkit_for_NLP-LREC2016.pdf

 

Best regards,

 

Alexandre Bérard, Christophe Servan, Olivier Pietquin and Laurent Besacier

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5-3-4An android application for speech data collection LIG_AIKUMA
We are pleased to announce the release of LIG_AIKUMA, an android application for speech data collection, specially dedicated to language documentation. LIG_AIKUMA is an improved version of the Android application (AIKUMA) initially developed by Steven Bird and colleagues. Features were added to the app in order to facilitate the collection of parallel speech data in line with the requirements of a French-German project (ANR/DFG BULB - Breaking the Unwritten Language Barrier). 
 
The resulting app, called LIG-AIKUMA, runs on various mobile phones and tablets and proposes a range of different speech collection modes (recording, respeaking, translation and elicitation). It was used for field data collections in Congo-Brazzaville resulting in a total of over 80 hours of speech.
 
Users who just want to use the app without access to the code can download it directly from the forge direct link: https://forge.imag.fr/frs/download.php/706/MainActivity.apk 
Code is also available on demand (contact elodie.gauthier@imag.fr and laurent.besacier@imag.fr).
 
More details on LIG_AIKUMA can be found on the following paper: http://www.sciencedirect.com/science/article/pii/S1877050916300448
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5-3-5Web services via ALL GO from IRISA-CNRS

It is our pleasure to introduce A||GO (https://allgo.inria.fr/ or http://allgo.irisa.fr/), a platform providing a collection of web-services for the automatic analysis of various data, including multimedia content across modalities. The platform builds on the back-end web service deployment infrastructure developed and maintained by  Inria?s  Service for Experimentation and Development (SED). Originally dedicated to multimedia content, A||GO progressively broadened to other fields such as computational biology, networks and telecommunications, computational graphics or computational physics.

As part of the CNRS PlaSciDo initiative [1], the Linkmedia team at IRISA / Inria Rennes is making available via A||GO a number of web services devoted to multimedia content analysis across modalities (language, audio, image, video). The web services provided currently include research results from the Linkmedia team as well as contribution from a number of partners. A list of the services available by the date is given below and the current state is available at
https://www-linkmedia.irisa.fr/software along with demo videos. Most web services are interoperable, facilitating the implementation of a multimedia content analysis processing chain, and are free to use for trial, prototyping or lab work. A brief and free account creation step will allow you to execute the web-services using either the graphical interface or a command line via a dedicated API.

We expect the number of web services to grow over time and invite interested parties to contact us should they wish to contribute the multimedia web service offer of A||GO.

List of multimedia content analysis tools currently available on A||GO:
- Audio Processing
        SaMuSa: music/speech segmentation
        SilAD: silence detection
        Radi.sh: repeated audio motif discovery
        LORIA STS v2: speech transcription for the French language from LORIA
        Multi channel BSS locate: audio source localization toolbox from IRISA-PANAMA
        A-spade: audio declipper from IRISA-PANAMA
        Transvox: voice faker from LORIA
- Natural Language Processing
        NERO: name entity recognition
        TermEx: keywords/indexing terms detection
        Otis!: topic segmentation
        Hi-tost: hierarchical topic structuring
- Video Processing
        Vidseg: video shot segmentation
        HUFA: face detection and tracking
Shortcuts to Linkmedia services are also available here:
https://www-linkmedia.irisa.fr/software/
 
For more information don't hesitate to contact us (
contact-multimedia-allgo@irisa.fr
). 

 
Gabriel Sargent and Guillaume Gravier
--
Linkmedia
IRISA - CNRS
Rennes, France

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5-3-6Clickable map - Illustrations of the IPA

Clickable map - Illustrations of the IPA


We have produced a clickable map showing the Illustrations of the International Phonetic
Alphabet.

The map is being updated with each new issue of the Journal of the International Phonetic
Association.

https://richardbeare.github.io/marijatabain/ipa_illustrations_all.html

Marija Tabain - La Trobe University, Australia
Richard Beare - Monash University & MCRI, Australia

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5-3-7LIG-Aikuma running on mobile phones and tablets

 


Dear all,

LIG is pleased to inform you that the website for the app Lig-Aikuma is online: https://lig-aikuma.imag.fr/
In the same time, an update of Lig-Aikuma (V3) was made available (see website).      

LIG-AIKUMA is a free Android app running on various mobile phones and tablets. The app proposes a range of different speech collection modes (recording, respeaking, translation and elicitation) and offers the possibility to share recordings between users. LIG-AIKUMA is built upon the initial AIKUMA app developed by S. Bird & F. Hanke (see https://en.wikipedia.org/wiki/Aikuma  for more information)

Improvements of the app:

  • Visual upgrade:
    + Waveform visualizer on the Respeaking and Translation modes (possibility to zoom in/out the audio signal)
    + File explorer included in all modes, to facilitate the navigation between files
    + New Share mode to share recordings between devices (by Bluetooth, Mail, NFC if available)
    + French and German languages available. In addition to English, the application now supports French and German languages. Lig-Aikuma uses by default the language of the phone/tablet.
    + New icons, more consistent to discriminate all type of files (audio, text, image, video)
  • Conceptual upgrade:
    + New name for the root project: ligaikuma ?> /! Henceforth, all data will be stored into this directory instead of ?aikuma? (in the previous versions of the app). This change doesn?t have compatibility issues. In the file explorer of the mode, the default position is this root directory. Just go back once with the left grey arrow (on the lower left of the screen) and select the ?aikuma? directory to access to your old recordings
    + Generation of a PDF consent form (from informations filled in the metadata form) that can be signed by linguist and speaker thanks to a pdf annotation tool (like Adobe Fill & Sign mobile app)
    + Generation of a CSV file which can be imported in Elan software: it will automatically create segmented tier, as it was done during a respeaking or a translation session. It will also mention by a ?non-speech? label that a segment has no speech.
    + Géolocalisation of the recordings
    + Respeak an elicit file: it is now possible to use in Respeaking or Translation mode an audio file initially recorded in Elicitation mode
  • Structural upgrade:
    + Undo button on Elicitation to erase/redo the current recording
    + Improvement session backup on Elicitation
    + Non-speech button in Respeaking and Translation modes to indicate by a comment that the segment does not contain speech (but noise or silent for instance)
    + Automatic speaker profile creation to quickly fill in the metadata infos if several sessions with a same speaker
Best regards,

Elodie Gauthier & Laurent Besacier
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5-3-8Python Library
Nous sommes heureux d'annoncer la mise à disposition du public de la
première bibliothèque en langage Python pour convertir des nombres écrits en
français en leur représentation en chiffres.
 
L'analyseur est robuste et est capable de segmenter et substituer les expressions
de nombre dans un flux de mots, comme une conversation par exemple. Il reconnaît les différentes
variantes de la langue (quantre-vingt-dix / nonante?) et traduit aussi bien les
ordinaux que les entiers, les nombres décimaux et les séquences formelles (n° de téléphone, CB?).
 
Nous espérons que cet outil sera utile à celles et ceux qui, comme nous, font du traitment
du langage naturel en français.
 
Cette bibliothèque est diffusée sous license MIT qui permet une utilisation très libre.
 
 
-- 
Romuald Texier-Marcadé
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