Multilingual language models (MLMs), such as XLM-R or BLOOM, are pretrained on data covering many languages and share their parameters across all languages. This modeling approach has several powerful advantages, such as allowing similar languages to exert positive influence on each other, and enabling cross-lingual task transfer (i.e., fine-tuning on some source language(s), then using the model on different target languages). The success of such transfer, however, depends on the model's ability to effectively share information between different languages in its parameter space. Yet, the cross-lingual information sharing mechanisms within MLMs are still not fully understood. In this talk, I will present our recent research that investigates this question from three different perspectives: encoding of typological relationships between languages within MLMs, language-wise modularity of MLMs and the influence of training examples in specific languages on predictions made in others.

Ekaterina Shutova is an Associate Professor at the ILLC, University of Amsterdam, where she leads the Amsterdam Natural Language Understanding Lab and the Natural Language Processing & Digital Humanities research unit. She received her PhD from the University of Cambridge, and then worked as a research scientist at the University of California, Berkeley. Ekaterina’s current research focuses on few-shot learning for language interpretation tasks, multilingual NLP, generalisability and robustness of NLP models and interpretability in deep learning. Her prominent service roles include Program Chair of ACL 2025, Senior Action Editor of ACL Rolling Review, Action Editor of Computational Linguistics and Demonstrations chair at EMNLP 2022. She is also an ELLIS scholar.

Abstract: Large language models (LLMs) constitute a paradigm shift in Natural Language Processing (NLP) and its applications across all domains. Models such as ChatGPT seem to possess human-like abilities --- reasoning about problems, passing bar exams, writing stories. But do they? In trying to answer this question, I will discuss three main desiderata for building human-centric NLP systems: knowledge-aware models, human-AI collaboration frameworks, and theoretically-grounded evaluation protocols. In this talk, I will use argumentation and creativity as two case studies. I will cover knowledge-aware models for implicit premise generation, human-AI collaboration framework for high-quality datasets creation (e.g., visual metaphors) and helping human solve tasks (e.g., writing short stories), and last but not least a novel evaluation protocol for assessing the creative capabilities of LLMs in both producing as well as assessing creative text.

Smaranda Muresan is a Research Scientist at the Data Science Institute at Columbia University, a Visiting Associate Professor at Barnard College and an Amazon Scholar. Her research focuses on human-centric Natural Language Processing for social good and responsible computing. She develops theory-guided and knowledge-aware computational models for understanding and generating language in context (e.g., visual, social, multilingual, multicultural) with applications to computational social science, education, and public health. Research topics that she worked on over the years include: argument mining and generation, fact-checking and misinformation detection, figurative language understanding and generation (e.g., sarcasm, metaphor, idioms), and multilingual language processing for low-resource and endangered languages. Recently, her research interests include explainable models and human-AI collaboration frameworks for high-quality datasets creation. She received best papers awards at SIGDIAL 2017 and ACL 2018 (short paper). She served as a board member for the North American Chapter of the Association for Computational Linguistics

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Humans make sense of language in context, bringing to bear their own understanding of the world including their model of their interlocutor's understanding of the world. In this talk, I will explore various potential risks that arise when we as humans bring this sense-making capacity to interactions with artificial interlocutors. That is, I will ask what happens in conversations where one party has no (or extremely limited) access to meaning and all of the interpretative work rests with the other, and briefly explore what this entails for the design of language technology.

Emily M. Bender is a Professor of Linguistics and an Adjunct Professor in the School of Computer Science and the Information School at the University of Washington, where she has been on the faculty since 2003. Her research interests include multilingual grammar engineering, computational semantics, and the societal impacts of language technology. In 2022 she was elected as a Fellow of the American Association for the Advancement of Science (AAAS).

No theory of speech perception can be considered complete without an explanation of how listeners are able to extract meaning from severely degraded forms of speech. Starting with a brief overview of a century of research which has seen the development of many types of distorted speech, followed by some anecdotal evidence that automatic speech recognisers still have some way to go to match listeners' performance in this area, I will describe the outcome of one recent [1] and several ongoing studies into the detailed time course of a listener's response to distorted speech. These studies variously consider the rapidity of adaptation, whether adaptation can only proceed if words are recognised, the degree to which the response to one form of distortion is conditioned on prior experience with other forms, and the nature of adaptation in a language other than one's own native tongue. Taken together, findings from these experiments suggest that listeners are capable of continuous and extremely rapid adaptation to novel forms of speech that differ greatly from the type of input that makes up the vast bulk of their listening experience. It is an open question as to whether big-data-based automatic speech recognition can offer a similar degree of flexibility. [1] Cooke, M, Scharenborg, O and Meyer, B (2022). The time course of adaptation to distorted speech. J. Acoust. Soc. Am. 151, 2636-2646. 10.1121/10.0010235

Martin Cooke is Ikerbasque Research Professor. After starting his career in the UK National Physical Laboratory, he worked at the University of Sheffield for 26 years before taking up his current position. His research has focused on analysing the computational auditory scene, devising algorithms for robust automatic speech recognition and investigating human speech perception. His interests also include the effects of noise on talkers as well as listeners, and second language listening in noise.

Speech production is a complex motor process involving several physiological phenomena, such as the neural, nervous and muscular activities that drive our respiratory, laryngeal and articulatory movements. Modeling speech production, in particular the relationship between articulatory gestures (tongue, lips, jaw, velum) and acoustic realizations of speech, is a challenging, and still evolving, research question. From an applicative point of view, such models could be embedded into assistive devices able to restore oral communication when part of the speech production chain is damaged (articulatory synthesis, silent speech interface). They could also help rehabilitate speech sound disorders using a therapy based on biofeedback (and articulatory inversion). From a more fundamental research perspective, such models can also be used to question the cognitive mechanisms underlying speech learning, perception and motor control. In this talk, I will present three recent studies conducted in our group to address some of these fundamental questions. In the first one, we quantified the benefit of relying on lip movement when learning speech representations in a self-supervised manner using predictive coding techniques. In the second one, we integrated articulatory priors into the latent space of a variational auto-encoder, with potential application to speech enhancement. In the third one, I will describe a first attempt toward a computational model of speech learning, based on deep learning, which can be used to understand how a child learns the acoustic-to-articulatory inverse mapping in a self-supervised manner.

Thomas Hueber is a senior research scientist at CNRS (« Directeur de recherche ») working at GIPSA-lab in Grenoble, France. He is head of the CRISSP research team (cognitive robotics, interactive systems and speech processing). He holds a Ph.D. in Computer Science from Pierre and Marie Curie University (Paris) in 2009. His research activities focus on automatic speech processing, with a particular interest in (1) the capture, analysis and modeling of articulatory gestures and electrophysiological signals involved in its production, (2) the development of speech technologies that exploit these different signals, for speech recognition and synthesis, for people with a spoken communication disorder, and (3) the study, through modeling and simulation, of the cognitive mechanisms underlying speech perception and production. He received in 2011 the 6th Christian Benoit award (ISCA/AFCP/ACB) and in 2015 the ISCA Award for the best paper published in Speech Communication. In 2017, he co-edited in IEEE/ACM Trans. Audio Speech and Language Processing, a special issue on Biosignal-based speech processing. He is also associate editor of EURASIP Journal on Audio, Speech, and Music Processing.

Recent advances in machine translation quality have improved its usefulness as a tool to satisfy the demand for multilingual information and communication. Machine translation is nowadays a common part of professional translation workflows, but it is not a tool exclusive to translators. Users of machine translation can be found, for example, in public service institutions and newsrooms looking to produce and disseminate information in multiple languages. At the same time, machine translation can also offer a way for people to access information that may not otherwise be available in their language. Effective and responsible use of machine translation, however, requires a clear understanding of the potential risks as well as potential benefits. In this talk, I discuss how machine translation is used for producing and accessing information and how various situational factors affect its use in different scenarios.

Dr Maarit Koponen currently works as Professor of Translation Studies at the University of Eastern Finland. She has previously worked as a post-doctoral researcher at the University of Helsinki and as a lecturer at the University of Turku after receiving her PhD in Language Technology at the University of Helsinki in 2016. Her research focuses on translation technology, particularly machine translation, and the effect of technology on translation both in professional and non-professional settings. Starting in October 2022, Koponen leads a work package focusing on linguistic barriers to information accessibility and technological solutions as part of the research project DECA (Democratic epistemic capacities in the age of algorithms), funded by the Academy of Finland Strategic Research Council. She chairs Working Group 7 “Language work, language professionals” of the EU COST Action “Language in the Human-Machine Era” (LITHME). She has also worked as a professional translator for several years.

Current NLP systems impress us by achieving close-to-human performance on benchmarks of answering commonsense questions or writing interesting stories. However, most of the progress is evaluated using static, closed-ended datasets created for individual tasks. To deploy commonsense reasoning services in the wild, we look to develop and evaluate systems that can generate answers in an open-ended way, perform robust logical reasoning, and generalize across diverse task formats, domains, and datasets. In this talk I will share our effort on introducing new formulations of commonsense reasoning challenges and novel evaluation protocols, towards broadening the scope in approaching machine common sense. We hope that such a shift of evaluation paradigm would encourage more research on externalizing the model reasoning process and improving model robustness and cross-task generalization.

Xiang Ren is an assistant professor and Viterbi Early Career Chair at the USC Computer Science Department, a Research Team Leader at USC ISI, and the director of the Intelligence and Knowledge Discovery (INK) Lab at USC. Priorly, he spent time as a research scholar at Stanford University and received his Ph.D. in Computer Science from the University of Illinois Urbana-Champaign. Ren's research seeks to build generalizable natural language processing (NLP) systems which can handle a wide variety of language tasks and situations. He works on new algorithms and datasets to make NLP systems cheaper to develop and maintain, arm machine models with common sense, and improve models’ transparency and reliability to build user trust. His research work has received several best paper awards in top NLP and AI conference venues. Ren has been awarded a NSF CAREER Award, multiple faculty research awards from Google, Facebook, Amazon, JP Morgan and Sony, and the 2018 ACM SIGKDD Doctoral Dissertation Award. He was named Forbes' Asia 30 Under 30 in 2019.

The development of advanced spoken language technologies based on automatic speech recognition (ASR) and text-to-speech synthesis (TTS) has enabled computers to either learn how to listen or speak. Many applications and services are now available but still support fewer than 100 languages. Nearly 7000 living languages that are spoken by 350 million people remain uncovered. This is because the construction is commonly done based on machine learning trained in a supervised fashion where a large amount of paired speech and corresponding transcription is required. In this talk, we will introduce a semi-supervised learning mechanism based on a machine speech chain framework. First, we describe the primary machine speech chain architecture that learns not only to listen or speak but also to listen while speaking. The framework enables ASR and TTS to teach each other given unpaired data. After that, we describe the use of machine speech chain for code-switching and cross-lingual ASR and TTS of several languages, including low-resourced ethnic languages. Finally, we describe the recent multimodal machine chain that mimics overall human communication to listen while speaking and visualizing. With the support of image captioning and production models, the framework enables ASR and TTS to improve their performance using an image-only dataset.

Sakriani Sakti is currently an associate professor at Japan Advanced Institute of Science and Technology (JAIST) Japan, adjunct associate professor at Nara Institute of Science and Technology (NAIST) Japan, visiting research scientist at RIKEN Center for Advanced Intelligent Project (RIKEN AIP) Japan, and adjunct professor at the University of Indonesia. She received DAAD-Siemens Program Asia 21st Century Award in 2000 to study in Communication Technology, University of Ulm, Germany, and received her MSc degree in 2002. During her thesis work, she worked with the Speech Understanding Department, DaimlerChrysler Research Center, Ulm, Germany. She then worked as a researcher at ATR Spoken Language Communication (SLC) Laboratories Japan in 2003-2009, and NICT SLC Groups Japan in 2006-2011, which established multilingual speech recognition for speech-to-speech translation. While working with ATR and NICT, Japan, she continued her study (2005-2008) with Dialog Systems Group University of Ulm, Germany, and received her Ph.D. degree in 2008. She was actively involved in international collaboration activities such as Asian Pacific Telecommunity Project (2003-2007) and various speech-to-speech translation research projects, including A-STAR and U-STAR (2006-2011). In 2011-2017, she was an assistant professor at the Augmented Human Communication Laboratory, NAIST, Japan. She also served as a visiting scientific researcher of INRIA Paris-Rocquencourt, France, in 2015-2016, under JSPS Strategic Young Researcher Overseas Visits Program for Accelerating Brain Circulation. In 2018–2021, she was a research associate professor at NAIST and a research scientist at RIKEN, Center for Advanced Intelligent Project AIP, Japan. Currently, she is an associate professor at JAIST, adjunct associate professor at NAIST, visiting research scientist at RIKEN AIP, and adjunct professor at the University of Indonesia. She is a member of JNS, SFN, ASJ, ISCA, IEICE, and IEEE. Furthermore, she is currently a committee member of IEEE SLTC (2021-2023) and an associate editor of the IEEE/ACM Transactions on Audio, Speech, and Language Processing (2020-2023). She was a board member of Spoken Language Technologies for Under-resourced languages (SLTU) and the general chair of SLTU2016. She was also the general chair of the "Digital Revolution for Under-resourced Languages (DigRevURL)" Workshop as the Interspeech Special Session in 2017 and DigRevURL Asia in 2019. She was also the organizing committee of the Zero Resource Speech Challenge 2019 and 2020. She was also involved in creating joint ELRA and ISCA Special Interest Group on Under-resourced Languages (SIGUL) and served as SIGUL Board since 2018. Last year, in collaboration with UNESCO and ELRA, she was also the organizing committee of the International Conference of "Language Technologies for All (LT4All): Enabling Linguistic Diversity and Multilingualism Worldwide". Her research interests lie in deep learning & graphical model framework, statistical pattern recognition, zero-resourced speech technology, multilingual speech recognition and synthesis, spoken language translation, social-affective dialog system, and cognitive-communication.

The design of widespread vision-and-language datasets and pre-trained encoders directly adopts, or draws inspiration from, the concepts and images of ImageNet. While one can hardly overestimate how much this benchmark contributed to progress in computer vision, it is mostly derived from lexical databases and image queries in English, resulting in source material with a North American or Western European bias. Therefore, we devise a new protocol to construct an ImageNet-style hierarchy representative of more languages and cultures. In particular, we let the selection of both concepts and images be entirely driven by native speakers, rather than scraping them automatically. Specifically, we focus on a typologically diverse set of languages, namely, Indonesian, Mandarin Chinese, Swahili, Tamil, and Turkish. On top of the concepts and images obtained through this new protocol, we create a multilingual dataset for Multicultural Reasoning over Vision and Language (MaRVL) by eliciting statements from native speaker annotators about pairs of images. The task consists of discriminating whether each grounded statement is true or false. We establish a series of baselines using state-of-the-art models and find that their cross-lingual transfer performance lags dramatically behind supervised performance in English. These results invite us to reassess the robustness and accuracy of current state-of-the-art models beyond a narrow domain, but also open up new exciting challenges for the development of truly multilingual and multicultural systems.

Desmond is an Assistant Professor at the University of Copenhagen. His primary research interests are multimodal and multilingual machine learning and he was involved in the creation of the Multi30K, How2, and MaRVL datasets. His work received an Area Chair Favourite paper at COLING 2018 and the Best Long Paper Award at EMNLP 2021. He co-organised the Multimodal Machine Translation Shared Task from 2016–2018, the 2018 Frederick Jelinek Memorial Workshop on Grounded Sequence-to-Sequence Learning, the How2 Challenge Workshop at ICML 2019, and the Workshop on Multilingual Multimodal Learning at ACL 2022.

Abstract Speech recognition is the mapping of a continuous, highly variable speech signal onto discrete, abstract representations. The question of how speech is represented and processed in the human brain and in automatic speech recognition (ASR) systems, although crucial in both the field of human speech processing and the field of automatic speech processing, has historically been investigated in the two fields separately. This webinar will discuss how comparisons between humans and deep neural network (DNN)-based ASRs, and cross-fertilization of the two research fields, can provide valuable insights into the way humans process speech and improve ASR technology. Specifically, it will present results of several experiments carried out on both human listeners and DNN-based ASR systems on the representation of speech in human listeners and DNNs and on lexically-guided perceptual learning, i.e., the ability to adapt a sound category on the basis of new incoming information resulting in improved processing of subsequent information. It will explain how listeners adapt to the speech of new speakers, and will present the results of a lexically-guided perceptual study carried out on a DNN-based ASR system, similar to the human experiments. In order to investigate the speech representations and adaptation processes in the DNN-based ASR systems, activations in the hidden layers of the DNN were visualized. These visualizations revealed that DNNs use speech representations that are similar to those used by human listeners, without being explicitly taught to do so, and showed an adaptation of the phoneme categories similar to what is assumed happens in the human brain.

Odette Scharenborg is an Associate Professor and Delft Technology Fellow at Delft University of Technology working on automatic speech processing. She has an interdisciplinary background in automatic speech recognition and psycholinguistics, and uses knowledge from how humans process speech in order to develop inclusive automatic speech recognition systems that are able to recognise speech from everyone, irrespective of how they speak or the language they speak. Since 2017, she is on the Board of the International Speech Communication Association, and currently serves as Vice-President. Since 2018, she is on the IEEE Speech and Language Processing Technical Committee, and she is a Senior Associate Editor of IEEE Signal Processing Letters.

Researchers in NLP increasingly frame and discuss research results in ways that serve to deemphasize the field's successes, at least in part in an effort to combat the field's widespread hype. Though well-meaning, this often yields misleading or even false claims about the limits of our best technology. This is a problem, and it may be more serious than it looks: It harms our credibility in ways that can make it harder to mitigate present-day harms, from NLP deployments, like those involving discriminatory systems for content moderation or resume screening. It also limits our ability to prepare for the potentially enormous impacts of more distant future advances. This talk urges researchers to be careful about these claims and suggests some research directions and communication strategies that will make it easier to avoid or rebut them.

Sam Bowman has been on the faculty at NYU since 2016, when he completed PhD with Chris Manning and Chris Potts at Stanford. At NYU, he is a member of the Center for Data Science, the Department of Linguistics, and Courant Institute's Department of Computer Science. His research focuses on data, evaluation techniques, and modeling techniques for sentence and paragraph understanding in natural language processing, and on applications of machine learning to scientific questions in linguistic syntax and semantics. He is the senior organizer behind the GLUE and SuperGLUE benchmark competitions and he has received a 2015 EMNLP Best Resource Paper Award, a 2019 *SEM Best Paper Award, a 2017 Google Faculty Research Award, and a 2021 NSF CAREER award.

As speech technologies mature and become ever more pervasive, the opportunities for real impact for people increases. This talk will outline the major challenges faced by researchers in porting mainstream speech technology to the domain of healthcare applications; in particular, the need for personalised systems and the challenge of working in an inherently sparse data domain. Three areas in automatic processing of pathological speech will be covered: i) detection, ii) therapy/treatment and iii) facilitating communication. The talk will give an overview of recent state-of-the-art results and specific experiences from current projects at the University of Sheffield (UK)'s Speech and Hearing (SPandH) & Healthcare lab.

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People love to argue. In recent years, Artificial Intelligence has achieved great advances in modelling natural language argumentation. While analysing and creating arguments is a highly complex (and enjoyable!) task at which even humans are not good, let alone perfect, we describe our natural language processing (NLP) research to identify arguments, their stance and aspects, aggregate arguments into topically coherent clusters, and finally, even to generate new arguments, given their desired topic, aspect and stance. The talk will tell you the story how the ArgumenText project has been conceptualized into a set of novel NLP tasks and highlight their main research outcomes. Argument mining has a tremendous number of possible applications, of which the talk discusses a few selected ones.

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The talk will be about some gotcha's in Deep Learning.

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Natural Language Processing (NLP) is at a very exciting time in its history. In the last 5 years a new technology has revolutionized the way we do our work. Even without special adaptation it tends to work better than almost every prior method, and yet we still don't really know how it works! So this is also a dangerous time: how can you trust a system that might (and sometimes does) do very strange things for which you can find no explanation or correction? In such a situation it is not a bad idea to look at the history of NLP, what NLP is at its core, and how the new technology fits into the NLP landscape. And, most importantly, where NLP is going (with or without this new technology) and how we can best prepare for it. The HiTZ Centre has a wonderful opportunity to help shape a future in which NLP will be as ubiquitous and as useful as the cellphone.

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