Senior Data Scientist at the University of Chicago

 

Please apply at https://uchicago.wd5.myworkdayjobs.com/External/job/Chicago-IL/Sr-Data-Scientist_JR24587

 

About the Department
 

The TMW Center for Early Learning + Public Health (TMW Center) develops science-based interventions, tools, and technologies to help parents and caregivers interact with young children in ways that maximize brain development. A rich language environment is critical to healthy brain development, however few tools exist to measure the quality or quantity of these environments. Access to this type of data allows caregivers to enhance interactions in real-time and gives policy-makers insight in how to best build policies that have a population-level impact.

The wearable team within TMW Center is building a low-cost wearable device that can reliably and accurately measure a child’s early language environment vis-à-vis the conversational turns between a child and caregiver. The goal is to provide accurate, real-time feedback that empowers parents and caregivers to create the best language environment for their children.

Job Summary
 

The job works independently to perform a variety of activities relating to software support and/or development. Analyzes, designs, develops, debugs, and modifies computer code for end user applications, beta general releases, and production support. Guides development and implementation of applications, web pages, and user-interfaces using a variety of software applications, techniques, and tools. Solves complex problems in administration, maintenance, integration, and troubleshooting of code and application ecosystem currently in production.

We are searching for a strategic and inquisitive senior data scientist to develop and optimize innovative AI-based models focused on speech/audio processing. The senior data scientist is expected to outline requirements, brainstorm ideas and solutions with leadership, manage data integrity and conduct experiments, assign tasks to junior staff, and monitor performance of the team.

 

Responsibilities

• Formulates, suggests, and manages data-driven projects to support the development of audio algorithms and use cases.
• Analyzes data from various entities for later use by junior data scientists.
• Assesses scope and timelines, prioritize goals, and prepare project plans to meet product and research objectives.
• Delegates tasks to junior data scientists and provide coaching to improve quality of work.
• Continuously trains and nurtures data scientists to take on bigger assignments.
• Provides leadership in advancing the science of TMW Center interventions by generating new ideas and collaborating with the research analysis team.
• In collaboration with CTO, selects and guides decisions on statistical procedures and model selections, including conducting exploratory experiments to develop proof of concept.
• Cross-validate models to ensure generalization and predictability.
• Stays informed about developments in Data Science and adjacent fields to ensure most relevant methods and outputs are being leveraged.
• Ensures data governance is in place to comply with regulations and privacy standards and maintain documentation of methodologies, coding, and results.
• Designs new systems, features, and tools. Solves complex problems and identifies opportunities for technical improvement and performance optimization. Reviews and tests code to ensure appropriate standards are met.
• Utilizes technical knowledge of existing and emerging technologies, including public cloud offerings from Amazon Web Services, Microsoft Azure, and Google Cloud.
• Acts as a technical consultant and resource for faculty research, teaching, and/or administrative projects.
• Performs other related work as needed.

Minimum Qualifications
 

Education:

Minimum requirements include a college or university degree in related field.

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Work Experience:

Minimum requirements include knowledge and skills developed through 5-7 years of work experience in a related job discipline.

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Certifications:

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Preferred Qualifications

Education:

• Master’s degree in Computer Science, Statistics, Mathematics, or Economics with a focus on computer science.

Experience:

• Experience with Machine Learning and LLMs.
• Experience working on audio or speech data.
• Experience implementing edge models using TensorFlow micro, TensorFlow lite, and corresponding quantization techniques.
• Experience building audio classification models or speech to text models.
• Experience using the latest pre-trained models such as whisper and wav2vec.
• Proven experience taking an idea or user need and translating it into fully realized applications.
• Ability to relay insights in layman’s terms to inform business decisions. 
• 3+ years leading and managing junior data scientists.

Technical Skills or Knowledge:

• Proficiency in Python, Pytorch, Tensorflow, TinyML, Pandas and Numpy.
• Experience with cloud environments such as AWS, Azure or GCloud.
• Experience with command line interfaces (Linux, SSH).
• Experience processing large datasets with Spark, Dask or Ray.

Application Documents

• Resume (required)
• Cover Letter (preferred)

When applying, the document(s) MUST be uploaded via the My Experience page, in the section titled Application Documents of the application.

Title: Predictive Modeling of Subjective Disagreement in Speech Annotation/Evaluation Host laboratory : LIUM

Location : Le Mans

Supervisors : Meysam Shamsi, Anthony Larcher

Beginning of internship : February 2024

Application deadline : 10/01/2024

Keywords: Subjective Disagreement Modeling, Synthetic Speech Quality Evaluation, Speech Emotion Recognition In the context of modeling subjective tasks, where diverse opinions, perceptions, and judgments exist among individuals, such as in speech quality or speech emotion recognition, addressing the challenge of defining ground truth and annotating a training set becomes crucial. The current practice of aggregating all annotations into a single label for modeling a subjective task is neither fair nor efficient [1]. The variability in annotations or evaluations can stem from various factors [2], broadly categorized into those associated with corpus quality and those intrinsic to the samples themselves. In the first case, the delicate definition of a subjective task introduces sensitivity into the annotation process, potentially leading to more errors, especially where the annotation tools and platform lack precision or annotators experience fatigue. In the second case, the inherent ambiguity in defining a subjective task and different perception may result in varying annotations and disagreements. Developing a predictive model to understand annotator/evaluator disagreement is crucial for engaging in discussions related to ambiguous samples and refining the definition of subjective concepts. Furthermore, this model can serve as a valuable tool for assessing the confidence of automatic evaluations [3,4]. This modeling approach will contribute to the automatic evaluation of corpus annotations, identification of ambiguous samples for reconsideration or re-annotation, automatic assessment of subjective models, and the detection of underrepresented samples and biases in the dataset. The proposed research involves utilizing a speech dataset such as MS-Podcast [5], SOMOS [6], VoiceMOS [7], for a subjective task with multiple annotations per sample. The primary objective is to predict the variation in assigned labels, measured through disagreement scores, entropy, or distribution.

Reference: [1]. Davani, A. M., Díaz, M., & Prabhakaran, V. (2022). Dealing with disagreements: Looking beyond the majority vote in subjective annotations. Transactions of the Association for Computational Linguistics, 10, 92-110.

[2]. Kreiman, J., Gerratt, B. R., & Ito, M. (2007). When and why listeners disagree in voice quality assessment tasks. The Journal of the Acoustical Society of America, 122(4), 2354-2364.

[3]. Wu, W., Chen, W., Zhang, C., & Woodland, P. C. (2023). It HAS to be Subjective: Human Annotator Simulation via Zero-shot Density Estimation. arXiv preprint arXiv:2310.00486.

[4]. Han, J., Zhang, Z., Schmitt, M., Pantic, M., & Schuller, B. (2017, October). From hard to soft: Towards more human-like emotion recognition by modelling the perception uncertainty. In Proceedings of the 25th ACM international conference on Multimedia (pp. 890-897).

[5]. Lotfian, R., & Busso, C. (2017). Building naturalistic emotionally balanced speech corpus by retrieving emotional speech from existing podcast recordings. IEEE Transactions on Affective Computing, 10(4), 471-483.

[6]. Maniati, G., Vioni, A., Ellinas, N., Nikitaras, K., Klapsas, K., Sung, J.S., Jho, G., Chalamandaris, A., Tsiakoulis, P. (2022) SOMOS: The Samsung Open MOS Dataset for the Evaluation of Neural Text-to-Speech Synthesis. Proc. Interspeech 2022, 2388-2392 [7]. Cooper, E., Huang, W. C., Tsao, Y., Wang, H. M., Toda, T., & Yamagishi, J. (2023). The VoiceMOS Challenge 2023: Zero-shot Subjective Speech Quality Prediction for Multiple Domains. arXiv preprint arXiv:2310.02640.

Applicant profile : Candidate motivated by artificial intelligence, enrolled in a Master's degree in Computer Science or related fields

For application: Send CV + cover letter to : meysam.shamsi@univ-lemans.fr or anthony.larcher@univ-lemans.fr before 10/01/2024

ANR Project «REVITALISE»

Automatic speech analysis of public talks. 

Description. Today, humanity has reached a stage at which extremely important aspects (such as information exchange) are tied not only to actual so-called hard skills, but also to soft skills. One such important skill is public speaking. Like many forms of interaction between people, the assessment of public speaking depends on many factors (often subjectively perceived). The goal of our project is to create an automatic system which can take into account these different factors and evaluate the quality of the performance. This requires understanding which elements can be assessed objectively and which vary depending on the listener [Hemamou, Wortwein, Chollet21]. For such an analysis, it is necessary to analyze public speaking at various levels: high-level (audio, video, text), intermediate (voice monotony, auto-gestures, speech structure, and etc.) and low-level (fundamental frequency, action units, POS / tags, and etc.) [Barkar]. This internship offers an opportunity to analyze the audio component of a public speech. The student is asked to solve two main problems. The engineering task is to create an automatic speech transcription system that detects speech disfluency. To do this, it is proposed to collect a bibliography on this topic and come up with an engineering solution. The second, research task, is to use audio cues to automatically analyze the success of a performance of a talk. This internship will give you an opportunity to solve an engineering problem as well as learn more about research approaches. By the end you will have expertise in audio processing as well and machine learning methods for multimodal analysis. If the internship is successfully completed, an article may be published. PhD position fundings on Social Computing will be accessible in the team at the end of the internship (at INRIA).

Registration & Organisation. Name of organization: Institut Polytechnique de Paris, Telecom-Paris Website of organization: https://www.telecom-paris.fr Department: IDS/LTCI/ Address: Palaiseau, France

Supervision. Supervision will include weekly meetings with the main supervisor and regular meetings (every 2-3 weeks) with co-supervisors. Telecom-Paris, 2023-2024 ANR Project «REVITALISE» Name of supervisor: Alisa BARKAR Name of co-supervisor: Chloe Clavel, Mathieu Chollet, Béatrice BIANCARDI Contact details: alisa.barkar@telecom-paris.fr

Duration & Planning. The internship is planned as a 5-6 month full-time internship for the spring semester 2024. 6 months considers 24 weeks within which it will be covered following list of activities:

● ACTIVITY 1(A1): Problem description and integration to the working environment

● ACTIVITY 2(A2): Bibliography overview

● ACTIVITY 3(A3): Implementation of the automatic transcription with detected discrepancies

● ACTIVITY 4(A4): Evaluation of the automatic transcription

● ACTIVITY 5(A5): Application of the developed methods to the existing data

● ACTIVITY 6(A6): Analysis of the importance of para-verbal features for the performance perception

● ACTIVITY 7(A7): Writing the report

Selected references of the team.

1. [Hemamou] L. Hemamou, G. Felhi, V. Vandenbussche, J.-C. Martin, C. Clavel, HireNet: a Hierarchical Attention Model for the Automatic Analysis of Asynchronous Video Job Interviews. in AAAI 2019, to appear

2. [Ben-Youssef] Atef Ben-Youssef, Chloé Clavel, Slim Essid, Miriam Bilac, Marine Chamoux, and Angelica Lim. Ue-hri: a new dataset for the study of user engagement in spontaneous human-robot interactions. In Proceedings of the 19th ACM International Conference on Multimodal Interaction, pages 464–472. ACM, 2017.

3. [Wortwein] Torsten Wörtwein, Mathieu Chollet, Boris Schauerte, Louis-Philippe Morency, Rainer Stiefelhagen, and Stefan Scherer. 2015. Multimodal Public Speaking Performance Assessment. In Proceedings of the 2015 ACM on International Conference on Multimodal Interaction (ICMI '15). Association for Computing Machinery, New York, NY, USA, 43–50.

4. [Chollet21] Chollet, M., Marsella, S., & Scherer, S. (2021). Training public speaking with virtual social interactions: effectiveness of real-time feedback and delayed feedback. Journal on Multimodal User Interfaces, 1-13.

5. [Barkar] Alisa Barkar, Mathieu Chollet, Beatrice Biancardi, and Chloe Clavel. 2023. Insights Into the Importance of Linguistic Textual Features on the Persuasiveness of Public Speaking. In Companion Publication of the 25th International Conference on Multimodal Interaction (ICMI '23 Companion). Association for Computing Machinery, New York, NY, USA, 51–55. https://doi.org/10.1145/3610661.3617161 Telecom-Paris, 2023-2024 ANR Project «REVITALISE»

Other references. 

1. Dinkar, T., Vasilescu, I., Pelachaud, C. and Clavel, C., 2020, May. How confident are you? Exploring the role of fillers in the automatic prediction of a speaker’s confidence. In ICASSP 2020-2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) (pp. 8104-8108). IEEE.

2. Whisper: Robust Speech Recognition via Large-Scale Weak Supervision, Radford A. et al., 2022, url: https://arxiv.org/abs/2212.04356

3. Romana, Amrit and Kazuhito Koishida. “Toward A Multimodal Approach for Disfluency Detection and Categorization.” ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) (2023): 1-5.

4. Radhakrishnan, Srijith et al. “Whispering LLaMA: A Cross-Modal Generative Error Correction Framework for Speech Recognition.” ArXiv abs/2310.06434 (2023): n. pag.

5. Wu, Xiao-lan et al. “Explanations for Automatic Speech Recognition.” ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) (2023): 1-5.

6. Min, Zeping and Jinbo Wang. “Exploring the Integration of Large Language Models into Automatic Speech Recognition Systems: An Empirical Study.” ArXiv abs/2307.06530 (2023): n. pag.

7. Ouhnini, Ahmed et al. “Towards an Automatic Speech-to-Text Transcription System: Amazigh Language.” International Journal of Advanced Computer Science and Applications (2023): n. pag.

8. Bigi, Brigitte. “SPPAS: a tool for the phonetic segmentations of Speech.” (2023).

9. Rekesh, Dima et al. “Fast Conformer with Linearly Scalable Attention for Efficient Speech Recognition.” ArXiv abs/2305.05084 (2023): n. pag.

10. Arisoy, Ebru et al. “Bidirectional recurrent neural network language models for automatic speech recognition.” 2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) (2015): 5421-5425.

11. Padmanabhan, Jayashree and Melvin Johnson. “Machine Learning in Automatic Speech Recognition: A Survey.” IETE Technical Review 32 (2015): 240 - 251.

12. Berard, Alexandre et al. “End-to-End Automatic Speech Translation of Audiobooks.” 2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) (2018): 6224-6228.

13. Kheir, Yassine El et al. “Automatic Pronunciation Assessment - A Review.” ArXiv abs/2310.13974 (2023): n. pag. Telecom-Paris, 2023-2024

Evaluation des systèmes de synthèse de la parole dans un environnement bruyant 

 Sujet L’´evaluation perceptive est capitale dans de nombreux domaines li´es au technologie de la parole dont la synth`ese de la parole. Elle permet d’´evaluer la qualit´e de la synth`ese de mani`ere subjective en demandant `a un jury[5] de noter la qualit´e d’un stimuli de parole synth´etis´ee[1, 2]. De r´ecent travaux ont permis de d´evelopper un mod`ele d’intelligence artificielle[3, 4] qui permet de pr´edire l’´evaluation subjective d’un segment de parole synth´etis´ee, ainsi permettant de s’affranchir d’un test par jury. Le probl`eme majeur de cette ´evaluation est l’interpr´etation du mot “qualit´e”. Certains peuvent baser leur jugement sur les caract´eristiques intrins`eques de la parole (tel que le timbre, le d´ebit de parole, la ponctuation, etc) alors que d’autres peuvent baser leur jugement sur les caract´eristiques li´es au signal audio (comme la pr´esence ou non de distorsion). Ainsi, l’´evaluation subjective de la parole peut ˆetre biais´ee par l’interpr´etation de la consigne par les auditeurs. Par cons´equent, le mod`ele d’intelligence artificielle mentionn´e ci-dessus peut ˆetre ainsi bas´e sur des mesures biais´ees. Le projet a pour but de r´ealiser un travail exploratoire pour ´evaluer la qualit´e de la synth`ese de la parole d’une mani`ere plus robuste que celle ayant ´et´e propos´e jusqu’ici. Pour ceci, nous partons de l’hypoth`ese que la qualit´e de la synth`ese de la parole peut ˆetre estim´ee par le biais de sa d´etection dans un environnement r´eel. En d’autre termes, un signal synth´etis´e parfaitement pour reproduire un signal de parole humaine ne devrait pas ˆetre d´etect´e dans un environnement de la vie quotidienne. Bas´e sur cette hypoth`ese, nous proposons donc de monter une exp´erience de perception de la parole en milieu bruyant. Il existe des m´ethodes de reproduction de milieu sonore qui permettent de simuler un environnement existant au casque. L’avantage de ces m´ethodes c’est qu’il est ´egalement possible de jouer un enregistrement d’un milieu r´eel au casque tout en ajoutant des signaux comme s’il avait ´et´e pr´esent dans la sc`ene sonore enregistr´ee. Ceci implique d’une part une campagne de mesure acoustique dans des environnement bruyant de la vie quotidienne (transport, open space, cantine, etc). Ensuite, une g´en´eration de parole synth´etis´ee sera n´ecessaire tout en prenant en compte le contexte des enregistrements. Il sera ´egalement pertinent de faire varier les param`etres de la parole synth´etis´ee tout en gardant la mˆeme s´emantique. Les enregistrements de la vie quotidienne seront ensuite mix´es aux signaux de parole synth´etis´ee pour ´evaluer la d´etection de cette derni`ere. Nous utiliserons le pourcentage de fois que la parole synth´etis´ee sera d´etect´ee comme indicateur de qualit´e. Ces pourcentages de d´etection seront ensuite compar´es au pr´ediction du mod`ele d’intelligence artificielle mentionn´e ci-dessus. Ainsi, nous pourrons conclure (1) si les m´ethodes sont ´equivalentes ou compl´ementaires et (2) quel(s) param`etre(s) de la parole synth´etis´ee engendre une d´etection de cette derni`ere en milieu bruyant.

Informations compl´ementaires:

• Encadrement: Le stage sera co-encadr´e par Aghilas Sini, maˆıtre de conf´erence au Laboratoire d’Informatique de l’Universit´e du Mans (aghilas.sini@univ-lemans.fr) et Thibault Vicente, maˆıtre de conf´erence au Laboratoire d’Acoustique de l’Universit´e du Mans (thibault.vicente@univ-lemans.fr)

• Niveau requis: Stage de M2 recherche

• P´eriode envisag´ee: 6 mois (F´evrier `a Juillet 2024)

• Lieu: Le Mans Universit´e

• mots-cl´es: parole synth´etis´ee, synth`ese sonore binaurale, test par jury

References

[1] Y.-Y. Chang. Evaluation of tts systems in intelligibility and comprehension tasks. In Proceedings of the 23rd Conference on Computational Linguistics and Speech Processing (ROCLING 2011), pages 64–78, 2011.

[2] J. Chevelu, D. Lolive, S. Le Maguer, and D. Guennec. Se concentrer sur les diff´erences: une m´ethode d’´evaluation subjective efficace pour la comparaison de syst`emes de synth`ese (focus on differences: a subjective evaluation method to efficiently compare tts systems*). In Actes de la conf´erence conjointe JEP-TALN-RECITAL 2016. volume 1: JEP, pages 137–145, 2016.

[3] C.-C. Lo, S.-W. Fu, W.-C. Huang, X. Wang, J. Yamagishi, Y. Tsao, and H.-M. Wang. MOSNet: Deep Learning-Based Objective Assessment for Voice Conversion. In Proc. Interspeech 2019, pages 1541–1545, 2019.

[4] G. Mittag and S. M¨oller. Deep learning based assessment of synthetic speech naturalness. arXiv preprint arXiv:2104.11673, 2021.

[5] M. Wester, C. Valentini-Botinhao, and G. E. Henter. Are we using enough listeners? no!—an empirically-supported critique of interspeech 2014 tts evaluations. In 16th Annu. Conf. Int. Speech Commun. Assoc., 2015.

M2 Master Internship 

Automatic Alsatian speech recognition 

1 Supervisors

Name: Emmanuel Vincent

Team and lab: Multispeech team, Inria research center at Université de Lorraine, Nancy

Email: emmanuel.vincent@inria.fr

Name: Pascale Erhart

Team and lab: Language/s and Society team, LiLPa, Strasbourg

Email: pascale.erhart@unistra.fr

2 Motivation and context This internship is part of the Inria COLaF project (Corpora and tools for the languages of France), whose objective is to develop and disseminate inclusive language corpora and technologies for regional languages (Alsatian, Breton, Corsican, Occitan, Picard, etc.), overseas languages and non-territorial immigration languages of France. With few exceptions, these languages are largely ignored by language technology providers [1]. However, such technologies are keys to the protection, promotion and teaching of these languages. Alsatian is the second regional language spoken in France in terms of number of speakers, with 46% of Alsace residents saying they speak Alsatian fairly well or very well [2]. However, it remains an underresourced language in terms of data and language technologies. Attempts at machine translation have been made as well as data collection [3].

3 Objectives The objective of the internship is to design an automatic speech recognition system for Alsatian based on sound archives (radio, television, web, etc.). This raises two challenges: i) Alsatian is not a homogeneous language but a continuum of dialectal varieties which are not always written in a standardized way, ii) the textual transcription is often unavailable or differs from the pronounced words (transcription errors , subtitles, etc.). Solutions will be based on i) finding a suitable methodology for choosing and preparing data, ii) designing an automatic speech recognition system using end-to-end neural networks which can rely on the adaptation of an existing multilingual system like Whisper [4] in a self-supervised manner from a number of untranscribed recordings [5] and in a supervised manner from a smaller number of transcribed recordings, or even from text-only data [6]. The work will be based on datasets collected by LiLPa and the COLaF project’s engineers, which include the television shows Sunndi's Kater [7] and Kùmme Mit [8] whose dialogues are scripted, some radio broadcasts from the 1950s–1970s with their typescripts [9], as well as untranscribed radio broadcasts of France Bleu Elsass. Dictionaries of Alsatian such as the Wörterbuch der elsässischen Mundarten which can be consulted via the Woerterbuchnetz portal [10] or phonetization initiatives [11] could be exploited, for example using Orthal spelling [12]. The internship opens the possibility of pursuing a PhD thesis funded by the COLaF project.

4 Bibliography

[1] DGLFLF, Rapport au Parlement sur la langue française 2023, https://www.culture.gouv.fr/Media/Presse/Rapport-au-Parlement-surla-langue-francaise-2023

[2] https://www.alsace.eu/media/5491/cea-rapport-esl-francais.pdf

[3] D. Bernhard, A-L Ligozat, M. Bras, F. Martin, M. Vergez-Couret, P. Erhart, J. Sibille, A. Todirascu, P. Boula de Mareüil, D. Huck, “Collecting and annotating corpora for three under-resourced languages of France: Methodological issues”, Language Documentation & Conservation, 2021, 15, pp.316-357.

[4] A. Radford, J.W. Kim, T. Xu, G. Brockman, C. McLeavey, I. Sutskever, “Robust speech recognition via large-scale weak supervision”, in 40th International Conference on Machine Learning, 2023, pp. 28492-28518.

[5] A. Bhatia, S. Sinha, S. Dingliwal, K. Gopalakrishnan, S. Bodapati, K. Kirchhoff, “Don't stop selfsupervision: Accent adaptation of speech representations via residual Adapters”, in Interspeech, 2023, pp. 3362-3366.

[6] N. San, M. Bartelds, B. Billings, E. de Falco, H. Feriza, J. Safri, W. Sahrozi, B. Foley, B. McDonnell, D. Jurafsky, “Leveraging supplementary text data to kick-start automatic speech recognition system development with limited transcriptions”, in 6th Workshop on Computational Methods for Endangered Languages, 2023, pp. 1-6.

[7] https://www.france.tv/france-3/grand-est/sunndi-s-kater/ [8] https://www.france.tv/france-3/grand-est/kumme-mit/toutes-les-videos/ [9] https://www.ouvroir.fr/cpe/index.php?id=1511 [10] https://woerterbuchnetz.de/?sigle=ElsWB#0 [11] 10.5281/zenodo.1174213 [12] https://orthal.fr/ 

5 Profile MSc in speech processing, natural language processing, computational linguistics, or computer science. Strong programming skills in Python/Pytorch. Knowledge of Alsatian and/or German is a plus, but in no way a prerequisite

-- Post-doctoral research position - L3i - La Rochelle France

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Title : Emotion detection by semantic analysis of the text in comics speech balloons

 

The L3i laboratory has one open post-doc position in computer science, in the specific field of natural language processing in the context of digitised documents.

 

Duration: 12 months (an extension of 12 months will be possible)

Position available from: as soon as possible

Salary: approximately 2100 € / month (net)

Place: L3i lab, University of La Rochelle, France

Specialty: Computer Science/ Document Analysis/ Natural Language Processing

Contact: Jean-Christophe BURIE (jcburie [at] univ-lr.fr) / Antoine Doucet (antoine.doucet [at] univ-lr.fr)

 

Position Description

The L3i is a research lab of the University of La Rochelle. La Rochelle is a city in the south west of France on the Atlantic coast and is one of the most attractive and dynamic cities in France. The L3i works since several years on document analysis and has developed a well-known expertise in ‘Bande dessinée”, manga and comics analysis, indexing and understanding.

The work done by the post-doc will take part in the context of SAiL (Sequential Art Image Laboratory) a joint laboratory involving L3i and a private company. The objective is to create innovative tools to index and interact with digitised comics. The work will be done in a team of 10 researchers and engineers.

The team has developed different methods to extract and recognise the text of the speech balloons. The specific task of the recruited researcher will be to use Natural Language Processing strategies to analyse the text in order to identify emotions expressed by a character (reacting to the utterance of another speaking character) or caused by it (talking to another character). The datasets will be collections of comics in French and English.

 

Qualifications

Candidates must have a completed PhD and a research experience in natural language processing. Some knowledge and experience in deep learning is also recommended.

 

General Qualifications

• Good programming skills mastering at least one programming language like Python, Java, C/C++

• Good teamwork skills

• Good writing skills and proficiency in written and spoken English or French

 

Applications

Candidates should send a CV and a motivation letter to jcburie [at] univ-lr.fr and antoine.doucet [at] univ-lr.fr.

Master 2 Internship Proposal 

Advisors: Jules Cauzinille, Benoˆıt Favre, Arnaud Rey

November 2023

Deep transfer knowledge from speech to primate vocalizations

Keywords: Computational bioacoustics, deep learning, self-supervised learning, transfer knowledge, efficient fine-tuning, primate vocalizations

1 Context This internship takes part in a multidisciplinary research project aimed at bridging the gap between state of the art deep leaning methods developed for speech processing and computational bioacoustics. Computational bioacoustics is a relatively new research filed which proposes to tackle the study of animal acoustic communication with computational approaches Stowell [2022]. Recently, bioacousticians are showing increasing interest for the deep learning revolution embodied in transformer architectures and self-supervised pre-trained models, but much investigation still needs to be carried out. We propose to test the viability of self-supervision and knowledge transfer as a bioacoustic tool by pre-training models on speech and using them for primate vocalisation analysis.

2 Problem Statement Speech based models are able to reach convincing performance on primate-related tasks including segmentation, individual identification or call type classification Sarkar and Doss [2023] as they are with many different downstream tasks (such as vocal emotion recognition Wang et al. [2021]). We have tested publicly available models such as HuBERT Hsu et al. [2021] and Wav2Vec2 [Schneider et al., 2019], two self-supervised speech-based architectures, on some of these tasks with Gibbon vocalizations. Our method involves probing and traditional fine-tuning of these models.

As to ensure true knowledge transfer from pre-training speech datasets to the downstream classification tasks, the goal of this internship will be to implement efficient fine-tuning methods in a similar fashion. These will allow to limit and control the amount of information lost in the finetuning process. Depending on the interests of the candidate, the methods can include prompt tuning Lester et al. [2021], attention prompting Gao et al. [2023], low rank adaptation Hu et al. [2021] or adversarial reprogramming Elsayed et al. [2018]. The candidate will also be free to explore other methods relevant to the question at hand, either on Gibbons or other species data-sets currently being collected.

3 Profile The intern will propose and implement the efficient fine-tuning solutions on an array of (preferably self-supervised) acoustic models pre-trained on speech or general sound such as HuBERT, Wav2vec, WavLM, VGGish, etc. Exploring adversial re-programming of models pre-trained on other modalities (images, videos, etc.) could also be carried out. The work will be implemented using pytorch.The candidate must have the following qualities :

• Excellent knowledge of deep learning methods

• Extensive experience with PyTorch models

• An interest in processing bioacoustic data

• An interest in reading and writing scientific papers as well as some curiosity for

research challenges

The internship will last 6 months at the LIS and LPC laboratories in Marseille during spring 2024.

The candidate will work in close collaboration with Jules Cauzinille as part of his thesis on “Self-supervised learning for primate vocalization analysis”. The candidate will also be in contact with the researchers community of the ILCB.

4 Contact Please send a CV, transcripts and a letter of application to jules.cauzinille@lis- lab.fr, benoit.favre@lislab.fr, and arnaud.rey@cnrs.fr. Do not hesitate to contact us if you have any question (or if you want to hear what our primates sound like).

References

Gamaleldin F. Elsayed, Ian Goodfellow, and Jascha Sohl-Dickstein. Adversarial reprogramming of neural networks, 2018.

Peng Gao, Jiaming Han, Renrui Zhang, Ziyi Lin, Shijie Geng, Aojun Zhou, Wei Zhang, Pan Lu, Conghui He, Xiangyu Yue, Hongsheng Li, and Yu Qiao. Llama-adapter v2: Parameter-efficient visual instruction model, 2023.

Wei-Ning Hsu, Benjamin Bolte, Yao-Hung Tsai, Kushal Lakhotia, Ruslan Salakhutdinov, and Abdelrahman Mohamed. Hubert: Self-supervised speech representation learning by masked prediction of hidden units. IEEE/ACM Transactions on Audio, Speech, and Language Processing, PP:1–1, 2021. doi: 10.1109/TASLP.2021.3122291.

Edward J. Hu, Yelong Shen, Phillip Wallis, Zeyuan Allen-Zhu, Yuanzhi Li, Shean Wang, Lu Wang, and Weizhu Chen. Lora: Low-rank adaptation of large language models, 2021.

Brian Lester, Rami Al-Rfou, and Noah Constant. The power of scale for parameter-efficient prompt tuning, 2021.

Eklavya Sarkar and Mathew Magimai Doss. Can Self-Supervised Neural Networks Pre-Trained on Human Speech distinguish Animal Callers?, May 2023. arXiv:2305.14035 [cs, eess].

Steffen Schneider, Alexei Baevski, Ronan Collobert, and Michael Auli. wav2vec: Unsupervised Pre-Training for Speech Recognition. In Proc. Interspeech 2019, pages 3465–3469, 2019. doi: 10.21437/Interspeech.2019-1873.

Dan Stowell. Computational bioacoustics with deep learning: a review and roadmap. 10:e13152, 2022. ISSN 2167-8359. doi: 10.7717/peerj.13152. URL https://peerj.com/articles/13152.

Yingzhi Wang, Abdelmoumene Boumadane, and Abdelwahab Heba. A fine-tuned wav2vec 2.0/hubert benchmark for speech emotion recognition, speaker verification and spoken language understanding. CoRR, abs/2111.02735, 2021. doi: 10.48550/arXiv.2111.02735

PhD Title: SUMMA-Sound : SUMMarization of Activities of daily living using Sound-based activity recognition Partnership:

IMT Atlantique : Campus ☒ Brest ☐ Nantes ☐ Rennes Laboratory : Lab-STICC Doctoral school : ☒ SPIN ☐ 3MG Funding: IMT Atlantique, co-tutelle with Instituto Superior Técnico

Context : IMT Atlantique, internationally recognised for the quality of its research, is a leading general engineering school under the aegis of the French Ministry of Industry and Digital Technology, ranked in the three main international rankings (THE, SHANGHAI, QS). Located on three campuses, Brest, Nantes and Rennes, IMT Atlantique aims to combine digital technology and energy to transform society and industry through training, research and innovation. It aims to be the leading French higher education and research institution in this field on an international scale. With 290 researchers and permanent lecturers, 1000 publications and 18 M€ of contracts, it supervises 2300 students each year and its training courses are based on cutting-edge research carried out within 6 joint research units: GEPEA, IRISA, LATIM, LABSTICC, LS2N and SUBATECH. The proposed thesis is part of the research activities of the team RAMBO (Robot interaction, Ambient systems, Machine learning, Behaviour, Optimization) and of the laboratory Lab-STICC and the department of Computer Science of IMT Atlantique. Scientific context: The objective of this thesis is to develop a method for collecting and summarizing domestic health-related data relevant for medical diagnosis, in a non-intrusive manner using audio information. This research addresses the lack of existing practical tools for providing high-level succinct information to medical staff on the evolution of patients they follow for health diagnostic purposes. This research is based on the assumption that valuable diagnostic data can be collected by observing short- and long-term lifestyle changes and behavioural anomalies. It relies on the latest advances in the domains of audio-based activity recognition, summarization of human activity, and health diagnosis. Research on health diagnosis in domestic environments has already explored a variety of sensors and modalities for gathering data on human health indicators [5]. Nevertheless, audio-based activity recognition is notable for its less intrusive nature. Employing state-of-the-art sound-based activity recognition models [2] to monitor domestic human activity, the thesis will investigate and develop methods for summarization of human activity [3] in a human-understandable language, in order to produce easily interpretable data by doctors who, remotely, monitor their patients [4]. This work continues and fosters the research of the RAMBO team at IMT Atlantique on ambient systems, enabling well ageing at home for the elderly adults or dependent populations [1]. We expect this thesis to provide technology likely to relieve the burden on gerontologists and elderly-care facilities, and alleviate the caregiver shortage by offering some automatic support to the task of monitoring elderly or handicapped people, enabling them to age-at-home while still being followed by medical specialists using automated means. Expected contributions of the thesis Scientific goals: (1) Determine the set of human activities relevant for health diagnosis, (2) Implement a state-of-the-art model for audio-based activity recognition and validate its function by clinicians, (3) Develop a model for summarizing the evolution of human activity over time intervals of arbitrary duration (typically spanning from days to months and possibly years). Expected outcomes of the PhD: (1) A model for semantic summarization of human activity, based on sound recognition of activities of daily living. (2) A proof of concept for this model Candidate profile and required skills: • Master Degree in Computer Science (or equivalent) • Programming and Software Engineering skills (Python, Git, Software Architecture Design) • Data science skills • Machine learning skills • English speaking and writing skills References: [1] Damien Bouchabou. “Human activity recognition in smart homes : tackling data variability using context-dependent deep learning, transfer learning and data synthesis”. Theses. Ecole nationale supérieure Mines-Télécom Atlantique, May 2022. url: https://theses.hal.science/tel-03728064. [2] Detection and Classification of Acoustic Scenes and Events (DCASE). url: https://dcase.community/challenge2022/task-soundevent-detection-in-domestic-environments (visited on 07/01/2022). [3] P Durga et al. “When less is better: A summarization technique that enhances clinical effectiveness of data”. In: Proceedings of the 2018 International Conference on Digital Health. 2018, pp. 116–120. [4] Akshay Jain et al. “Linguistic summarization of in-home sensor data”. In: Journal of Biomedical Informatics 96 (2019), p. 103240. issn: 1532-0464. [5] Mostafa Haghi Kashani et al. “A systematic review of IoT in healthcare: Applications, techniques, and trends”. In: Journal of Network and Computer Applications 192 (2021), p. 103164. Work Plan: The thesis will be organised in the following steps: (1) Definition of pertinent sounds and activities for health diagnosis, (2) Hardware set-up, (3) Dataset constitution, (4) Activity recognition, (5) Diarization of activities, (6) Summarization, (7) Validation in a real environment. Application: To apply for this position, please send an email with your Curriculum Vitae, a document with your academic results (if possible), and a couple of lines describing your motivation to pursue a PhD to mihai[dot]andries[at]imt-atlantique[dot]fr before 16 May 2023. Additional Information :  Application deadline : 16 May 2023  Start date : Fall 2023  Contract duration: 36 months  Localisation - Location : Brest (France) and Lisbon (Portugal)  Contact(s) : Mihai ANDRIES (mihai[dot]andries[at]imt-atlantique.fr) Plinio Moreno (plinio[at]isr.tecnico.ulisboa.pt)

We’re happy to announce a new research position in the field of speech- and text anonymization at German Research Center for Artificial Intelligence, Berlin, Germany. We’re looking for a full time Researcher or Junior Researcher level, and offer a 2 years contract with optional prolongation and PhD perspective. 

The Speech and Language Technology Lab at DFKI Berlin is involved in numerous national as well as international research programmes and networks. We offer an interesting and flexible work environment as part of an innovative, international and enthusiastic team which coordinates and participates in national as well as European projects in the wider area of Language Technology. Your tasks - Research and development of methods for anonymization and fake-detection using text, speech, and semi-structural data from medical and fake-news application domains - Research and development of methods for explainability, fairness, trustworthiness for above methods - Support of industrial and public scientific, research and development projects on national and international level ( BMBF, EU, etc.) - Exchange and network with relevant stakeholder from science, industry, politics and civil society. Your qualifications - Masters degree (or equivalent qualification) in Computer Science, Computational Linguistics, Signal Processing or a related discipline - MA degree in the area of speech, NLP, AI, ML, xAI - Good English or German language skills - Confident appearance in an international working environment and high self-motivation - Ability to work in a team, problem-solving and results-oriented Your benefits - A wide and interesting range of tasks and extensive development opportunities in a young and international team - Contribution to trend-setting, innovative projects in the field of Language Technology and AI - An innovative, agile and professional work environment - An attractive workplace in the heart of Berlin with PhD perspective The German Research Center for Artificial Intelligence (DFKI) is Germany's leading business-oriented research institution in the field of innovative software technologies based on artificial intelligence methods. In the international scientific community, DFKI ranks among the most recognized "Centers of Excellence" and currently is the biggest research center worldwide in the area of Artificial Intelligence and its application in terms of number of employees and the volume of external funds. The DFKI cooperates closely with national and international companies. DFKI encourages applications from people with disability; DFKI intends to increase the proportion of female employees in the field of science and encourages women to apply for this position. Application deadline: Dec 23 More details and link: https://jobs.dfki.de/en/internal/vacancy/en-researcher-m-w-d-in-506968.html