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ISCApad Archive  »  2018  »  ISCApad #235  »  Resources  »  Software

ISCApad #235

Wednesday, January 10, 2018 by Chris Wellekens

5-3 Software
5-3-1ROCme!: a free tool for audio corpora recording and management

ROCme!: nouveau logiciel gratuit pour l'enregistrement et la gestion de corpus audio.

Le logiciel ROCme! permet une gestion rationalisée, autonome et dématérialisée de l’enregistrement de corpus lus.

Caractéristiques clés :
- gratuit
- compatible Windows et Mac
- interface paramétrable pour le recueil de métadonnées sur les locuteurs
- le locuteur fait défiler les phrases à l'écran et les enregistre de façon autonome
- format audio paramétrable

Téléchargeable à cette adresse :


5-3-2VocalTractLab 2.0 : A tool for articulatory speech synthesis

VocalTractLab 2.0 : A tool for articulatory speech synthesis

It is my pleasure to announce the release of the new major version 2.0 of VocalTractLab. VocalTractLab is an articulatory speech synthesizer and a tool to visualize and explore the mechanism of speech production with regard to articulation, acoustics, and control. It is available from .
Compared to version 1.0, the new version brings many improvements in terms of the implemented models of the vocal tract, the vocal folds, the acoustic simulation, and articulatory control, as well as in terms of the user interface. Most importantly, the new version comes together with a manual.

If you like, give it a try. Reports on bugs and any other feedback are welcome.

Peter Birkholz


5-3-3Bob signal-processing and machine learning toolbox (v.1.2..0)

    The release 1.2.0 of the Bob signal-processing and machine learning toolbox is available .
    Bob provides both efficient implementations of several machine     learning algorithms as well as a framework to help researchers to     publish reproducible research.

It is developed by the Biometrics Group at Idiap in  Switzerland.

    The previous release of Bob was providing:
    * image, video and audio IO interfaces such as jpg, avi, wav, 
    * database accessors such as FRGC, Labelled Face in the Wild, and many     others,
    *mage processing: Local Binary Patterns (LBPs), Gabor Jets,  SIFT,
    * machines  and trainers such as Support Vector Machines (SVMs), k-Means,     Gaussian Mixture Models (GMMs), Inter-Session Variability modeling     (ISV), Joint Factor Analysis (JFA), Probabilistic Linear     Discriminant Analysis (PLDA), Bayesian intra/extra (personal)     classifier,
    The new release of Bob has brought the following features and/or improvements, such as:
    * Unified implementation of Local Binary Patterns (LBPs),
    * Histograms of Oriented Gradients (HOG) implementation,
    * Total variability (i-vector) implementation,
    * Conjugate gradient based-implementation for logistic regression,
    * Improved multi-layer perceptrons implementation (Back-propagation can now be easily used in combination with any optimizer -- i.e     L-BFGS),
    * Pseudo-inverse-based method for Linear Discriminant Analysis,
    * Covariance-based method for Principal Component Analysis,
    * Whitening and within-class covariance normalization techniques,
    * Module for object detection and keypoint localization     (bob.visioner),
    * Module for audio processing including feature extraction such as LFCC and     MFCC,
    * Improved extensions (satellite packages), that now support both     Python and C++ code, within an easy to use framework,
    * Improved documentation and add new tutorials,
    * Support for Intel's MKL (in addition to ATLAS),
    * Extend supported platforms (Arch Linux).
    This release represents a major milestone in Bob with plenty of  functionality improvements (>640 commits in total) and plenty of bug fixes.
    • Sources and Documentation
    • Binary packages:
    •     Ubuntu: 10.04, 12.04, 12.10 and 13.04
    • For     Mac OSX: works with 10.6 (Snow Leopard), 10.7 (Lion) and 10.8     (Mountain Lion)
    For instructions on how to install pre-packaged version on Ubuntu or     OSX, consult our quick       installation instructions  (N.B. OS X macport has not yet been     upgraded. This will be done very soon. cf. ).
    Best regards,
    Elie Khoury (on Behalf of the Biometric Group at Idiap lead by Sebastien Marcel)

 Dr. Elie Khoury Post Doctorant Biometric Person Recognition Group 
IDIAP Research Institute (Switzerland) Tel : +41 27 721 77 23

5-3-4COVAREP: A Cooperative Voice Analysis Repository for Speech Technologies
CALL for contributions
We are pleased to announce the creation of an open-source repository of advanced speech processing algorithms called COVAREP (A Cooperative Voice Analysis Repository for Speech Technologies). COVAREP has been created as a GitHub project ( where researchers in speech processing can store original implementations of published algorithms.
Over the past few decades a vast array of advanced speech processing algorithms have been developed, often offering significant improvements over the existing state-of-the-art. Such algorithms can have a reasonably high degree of complexity and, hence, can be difficult to accurately re-implement based on article descriptions. Another issue is the so-called 'bug magnet effect' with re-implementations frequently having significant differences from the original. The consequence of all this has been that many promising developments have been under-exploited or discarded, with researchers tending to stick to conventional analysis methods.
By developing the COVAREP repository we are hoping to address this by encouraging authors to include original implementations of their algorithms, thus resulting in a single de facto version for the speech community to refer to.
We envisage a range of benefits to the repository:
1) Reproducible research: COVAREP will allow fairer comparison of algorithms in published articles.
2) Encouraged usage: the free availability of these algorithms will encourage researchers from a wide range of speech-related disciplines (both in academia and industry) to exploit them for their own applications.
3) Feedback: as a GitHub project users will be able to offer comments on algorithms, report bugs, suggest improvements etc.
We welcome contributions from a wide range of speech processing areas, including (but not limited to): Speech analysis, synthesis, conversion, transformation, enhancement, speech quality, glottal source/voice quality analysis, etc.
In order to achieve a reasonable standard of consistency and homogeneity across algorithms we have compiled a list of requirements for prospective contributors to the repository. However, we intend the list of the requirements not to be so strict as to discourage contributions.
  • Only published work can be added to the   repository
  • The code must be available as open source
  • Algorithms should be coded in Matlab, however we   strongly encourage authors to make the code compatible with Octave in order to   maximize usability
  • Contributions have to comply with a Coding   Convention (see GitHub site for coding convention and template). However, only   for normalizing the inputs/outputs and the documentation. There is no   restriction for the content of the functions (though, comments are obviously   encouraged).
Getting contributing institutions to agree to a homogenous IP policy would be close to impossible. As a result COVAREP is a repository and not a toolbox, and each algorithm will have its own licence associated with it. Though flexible to different licence types, contributions will need to have a licence which is compatible with the repository, i.e. {GPL, LGPL, X11, Apache, MIT} or similar. We would encourage contributors to try to obtain LGPL licences from their institutions in order to be more industry friendly.
We believe that the COVAREP repository has a great potential benefit to the speech research community and we hope that you will consider contributing your published algorithms to it. If you have any questions, comments issues etc regarding COVAREP please contact us on one of the email addresses below. Please forward this email to others who may be interested.
Existing contributions include: algorithms for spectral envelope modelling, adaptive sinusoidal modelling, fundamental frequncy/voicing decision/glottal closure instant detection algorithms, methods for detecting non-modal phonation types etc.
Gilles Degottex <>, John Kane <>, Thomas Drugman <>, Tuomo Raitio <>, Stefan Scherer <>

5-3-5Release of the version 2 of FASST (Flexible Audio Source Separation Toolbox).
Release of the version 2 of FASST (Flexible Audio Source Separation Toolbox). 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)

5-3-6Cantor 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:

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
The Cantor Digitalis team (who loves feedback —
Christophe d'Alessandro, Lionel Feugère, Olivier Perrotin

5-3-7MultiVec: 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 ( with installation instructions and command-line usage examples.


When you use this toolkit, please cite:



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:


Best regards,


Alexandre Bérard, Christophe Servan, Olivier Pietquin and Laurent Besacier


5-3-8An 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: 
Code is also available on demand (contact and
More details on LIG_AIKUMA can be found on the following paper:

5-3-9(2017-09-20) 11th Oxford Dysfluency Conference , Oxford, UK

11th Oxford Dysfluency Conference 
Challenge and Change

20-23 September 2017 | St Catherine’s College, Oxford, UK

The 11th Oxford Dysfluency Conference (ODC), under the theme ‘Challenge and Change’, is to be held at St Catherine’s College Oxford from 20-23 September, 2017. ODC has a reputation as one of the leading international scientific conferences in the field of dysfluency.

Abstract submission deadline: 31 March 2017.

The conference brings together researchers and clinicians, providing a showcase and forum for discussion and collegial debate about the most current and innovative research and clinical practices. Throughout the history of ODC, the primary aim has been to bridge the gap between research and clinical practice.

The conference seeks to promote research that informs management, with interventions that are supported by sound theory and which inform future research.

In 2017, the goal is to encourage discussion and debate that will challenge and enhance our perspectives and understanding of research; the nature of stuttering and / or cluttering; and management across the ages.


Abstract submission deadline: 31 March 2017

Abstracts are now invited on the following topics. They should be submitted using the online abstract submission system.

  1. New perspectives on assessment and therapy in children
  2. New perspectives on assessment and therapy in adolescents
  3. New perspectives on assessment and therapy in adults
  4. Issues, variables, and controversies in assessment and outcome
  5. Supporting the next generation of clinicians and researchers
  6. Evidence in practice
  7. Neurological foundations

The 2017 conference will enable delegates to:

  • Present and learn from the latest research developments and findings
  • Explore issues relating to the nature of stuttering and / or cluttering and its treatment
  • Develop knowledge and clinical skills for working with children and adults who stutter and / or clutter
  • Advance research in the field of dysfluency
  • Consider ways to integrate research into clinical practice
  • Support and encourage new researchers in the field
  • Develop collaborations with researchers working in dysfluency
  • Provide informal opportunities to meet and discuss ideas with leading experts in the field in a friendly environment

5-3-10Web services via ALL GO from IRISA-CNRS

It is our pleasure to introduce A||GO ( or, 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 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 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:
For more information don't hesitate to contact us (

Gabriel Sargent and Guillaume Gravier
Rennes, France


5-3-11Clickable map - Illustrations of the IPA

Clickable map - Illustrations of the IPA

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

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

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


5-3-12LIG-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:
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  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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