Portrait de Cem Subakan

Cem Subakan

Membre académique associé

Professeur adjoint, Université Laval, Département d'informatique et de génie logiciel

Professeur associé, Concordia University, École de génie et d'informatique Gina-Cody

Sujets de recherche

Apprentissage multimodal

Biographie

Cem Subakan est professeur adjoint à l'Université Laval, au sein du Département d'informatique et de génie logiciel. Il est également professeur adjoint affilié au Département d'informatique et de génie logiciel de l'Université Concordia, ainsi que membre académique associé à Mila – Institut québécois d'intelligence artificielle. Il a obtenu un doctorat en informatique de l'Université de l'Illinois à Urbana-Champaign (UIUC) et a effectué un postdoctorat à Mila. Il agit en tant que relecteur pour plusieurs conférences, notamment NeurIPS, ICML, ICLR, ICASSP et MLSP, ainsi que pour des revues telles que IEEE Signal Processing Letters (SPL) et IEEE Transactions on Audio, Speech, and Language Processing (TASL). Ses recherches portent principalement sur l'apprentissage automatique appliqué à la parole et à l'audio. Plus précisément, il travaille sur l'apprentissage profond pour la séparation de sources et l'amélioration de la parole dans des conditions réalistes, l'interprétabilité des réseaux neuronaux, l'apprentissage continu et l'apprentissage multimodal. Il a reçu le Prix du meilleur article étudiant lors de la conférence IEEE Machine Learning for Signal Processing (MLSP) en 2017, ainsi que la bourse Sabura Muroga du Département d'informatique de l'UIUC. Il est également un contributeur clé au projet SpeechBrain, où il dirige la partie consacrée à la séparation de la parole.

Étudiants actuels

Matthieu Cervera

Co-superviseur⋅e :

Mirco Ravanelli

Maîtrise recherche - Université Laval

Luca Della Libera

Doctorat - Concordia

Superviseur⋅e principal⋅e :

Mirco Ravanelli

Maab Elrashid Ahmed Mohamed

Doctorat - Concordia

Superviseur⋅e principal⋅e :

Mirco Ravanelli

Doctorat - Université Laval

Co-superviseur⋅e :

Laurent Charlin

Doctorat - Université Laval

Co-superviseur⋅e :

Mirco Ravanelli

Aravind Krishnan

Collaborateur·rice alumni - Saarland University

Eleonora Mancini

Collaborateur·rice alumni - UdeM

Co-superviseur⋅e :

Mirco Ravanelli

Francesco Paissan

Doctorat - Université Laval

Co-superviseur⋅e :

Mirco Ravanelli

Billets de blogue

Visual of FocalCodec,nouvelle méthode pour compresser la parole sans sacrifier la qualité, en vue d'obtenir des LLM multimodaux plus efficaces.

23 janvier 2026

FocalCodec : donner l’ouïe et la parole aux LLM à débit ultra-faible

par

Luca Della Libera

Francesco Paissan

Mirco Ravanelli

Publications

Beyond Fixed Frames: Dynamic Character-Aligned Speech Tokenization

Luca Della Libera

Mirco Ravanelli

Neural audio codecs are at the core of modern conversational speech technologies, converting continuous speech into sequences of discrete to… (voir plus)kens that can be processed by LLMs. However, existing codecs typically operate at fixed frame rates, allocating tokens uniformly in time and producing unnecessarily long sequences. In this work, we introduce DyCAST, a Dynamic Character-Aligned Speech Tokenizer that enables variable-frame-rate tokenization through soft character-level alignment and explicit duration modeling. DyCAST learns to associate tokens with character-level linguistic units during training and supports alignment-free inference with direct control over token durations at decoding time. To improve speech resynthesis quality at low frame rates, we further introduce a retrieval-augmented decoding mechanism that enhances reconstruction fidelity without increasing bitrate. Experiments show that DyCAST achieves competitive speech resynthesis quality and downstream performance while using significantly fewer tokens than fixed-frame-rate codecs. Code and checkpoints will be released publicly at https://github.com/lucadellalib/dycast.

2026-01-29

ArXiv (prépublication)

Toward Faithful Explanations in Acoustic Anomaly Detection

Maab Elrashid

Anthony Deschênes

Mirco Ravanelli

Rémi Georges

Interpretability is essential for user trust in real-world anomaly detection applications. However, deep learning models, despite their stro… (voir plus)ng performance, often lack transparency. In this work, we study the interpretability of autoencoder-based models for audio anomaly detection, by comparing a standard autoencoder (AE) with a mask autoencoder (MAE) in terms of detection performance and interpretability. We applied several attribution methods, including error maps, saliency maps, SmoothGrad, Integrated Gradients, GradSHAP, and Grad-CAM. Although MAE shows a slightly lower detection, it consistently provides more faithful and temporally precise explanations, suggesting a better alignment with true anomalies. To assess the relevance of the regions highlighted by the explanation method, we propose a perturbation-based faithfulness metric that replaces them with their reconstructions to simulate normal input. Our findings, based on experiments in a real industrial scenario, highlight the importance of incorporating interpretability into anomaly detection pipelines and show that masked training improves explanation quality without compromising performance.

2025-12-31

arXiv (publié)

Investigating Faithfulness in Large Audio Language Models

Lovenya Jain

Mirco Ravanelli

Faithfulness measures whether chain-of-thought (CoT) representations accurately reflect a model's decision process and can be used as reliab… (voir plus)le explanations. Prior work has shown that CoTs from text-based LLMs are often unfaithful. This question has not been explored for large audio-language models (LALMs), where faithfulness is critical for safety-sensitive applications. Reasoning in LALMs is also more challenging, as models must first extract relevant clues from audio before reasoning over them. In this paper, we investigate the faithfulness of CoTs produced by several LALMs by applying targeted interventions, including paraphrasing, filler token injection, early answering, and introducing mistakes, on two challenging reasoning datasets: SAKURA and MMAR. After going through the aforementioned interventions across several datasets and tasks, our experiments suggest that, LALMs generally produce CoTs that appear to be faithful to their underlying decision processes.

2025-09-26

ArXiv (prépublication)

Investigating Faithfulness in Large Audio Language Models

Lovenya Jain

Mirco Ravanelli

Faithfulness measures whether chain-of-thought (CoT) representations accurately reflect a model's decision process and can be used as reliab… (voir plus)le explanations. Prior work has shown that CoTs from text-based LLMs are often unfaithful. This question has not been explored for large audio-language models (LALMs), where faithfulness is critical for safety-sensitive applications. Reasoning in LALMs is also more challenging, as models must first extract relevant clues from audio before reasoning over them. In this paper, we investigate the faithfulness of CoTs produced by several LALMs by applying targeted interventions, including paraphrasing, filler token injection, early answering, and introducing mistakes, on two challenging reasoning datasets: SAKURA and MMAR. After going through the aforementioned interventions across several datasets and tasks, our experiments suggest that, LALMs generally produce CoTs that appear to be faithful to their underlying decision processes.

2025-09-26

ArXiv (prépublication)

Virtual Consistency for Audio Editing

Matthieu Cervera

Francesco Paissan

Mirco Ravanelli

Free-form, text-based audio editing remains a persistent challenge, despite progress in inversion-based neural methods. Current approaches r… (voir plus)ely on slow inversion procedures, limiting their practicality. We present a virtual-consistency based audio editing system that bypasses inversion by adapting the sampling process of diffusion models. Our pipeline is model-agnostic, requiring no fine-tuning or architectural changes, and achieves substantial speed-ups over recent neural editing baselines. Crucially, it achieves this efficiency without compromising quality, as demonstrated by quantitative benchmarks and a user study involving 16 participants.

2025-09-21

ArXiv (prépublication)

Virtual Consistency for Audio Editing

Matthieu Cervera

Francesco Paissan

Mirco Ravanelli

Free-form, text-based audio editing remains a persistent challenge, despite progress in inversion-based neural methods. Current approaches r… (voir plus)ely on slow inversion procedures, limiting their practicality. We present a virtual-consistency based audio editing system that bypasses inversion by adapting the sampling process of diffusion models. Our pipeline is model-agnostic, requiring no fine-tuning or architectural changes, and achieves substantial speed-ups over recent neural editing baselines. Crucially, it achieves this efficiency without compromising quality, as demonstrated by quantitative benchmarks and a user study involving 16 participants.

2025-09-21

ArXiv (prépublication)

FocalCodec: Low-Bitrate Speech Coding via Focal Modulation Networks

Luca Della Libera

Francesco Paissan

Mirco Ravanelli

Large language models have revolutionized natural language processing through self-supervised pretraining on massive datasets. Inspired by t… (voir plus)his success, researchers have explored adapting these methods to speech by discretizing continuous audio into tokens using neural audio codecs. However, existing approaches face limitations, including high bitrates, the loss of either semantic or acoustic information, and the reliance on multi-codebook designs when trying to capture both, which increases architectural complexity for downstream tasks. To address these challenges, we introduce FocalCodec, an efficient low-bitrate codec based on focal modulation that utilizes a single binary codebook to compress speech between 0.16 and 0.65 kbps. FocalCodec delivers competitive performance in speech resynthesis and voice conversion at lower bitrates than the current state-of-the-art, while effectively handling multilingual speech and noisy environments. Evaluation on downstream tasks shows that FocalCodec successfully preserves sufficient semantic and acoustic information, while also being well-suited for generative modeling. Demo samples and code are available at https://lucadellalib.github.io/focalcodec-web/.

2025-09-17

NeurIPS.cc/2025/Conference (poster)

FocalCodec-Stream: Streaming Low-Bitrate Speech Coding via Causal Distillation

Luca Della Libera

Mirco Ravanelli

Neural audio codecs are a fundamental component of modern generative audio pipelines. Although recent codecs achieve strong low-bitrate reco… (voir plus)nstruction and provide powerful representations for downstream tasks, most are non-streamable, limiting their use in real-time applications. We present FocalCodec-Stream, a hybrid codec based on focal modulation that compresses speech into a single binary codebook at 0.55 - 0.80 kbps with a theoretical latency of 80 ms. Our approach combines multi-stage causal distillation of WavLM with targeted architectural improvements, including a lightweight refiner module that enhances quality under latency constraints. Experiments show that FocalCodec-Stream outperforms existing streamable codecs at comparable bitrates, while preserving both semantic and acoustic information. The result is a favorable trade-off between reconstruction quality, downstream task performance, latency, and efficiency. Code and checkpoints will be released at https://github.com/lucadellalib/focalcodec.

2025-09-01

arXiv (publié)

Autoregressive Speech Enhancement via Acoustic Tokens

Luca Della Libera

Mirco Ravanelli

2025-07-01

arXiv (publié)

Discrete Audio Tokens: More Than a Survey!

Gallil Maimon

Adel Moumen

Darius Petermann

Jiatong Shi

Haibin Wu

Haici Yang

Anastasia Kuznetsova

Artem Ploujnikov

Bhuvana Ramabhadran

Benjamin Elizalde

Jinyu Li

Phil Woodland

Minje Kim

Hung-yi Lee

Shinji Watanabe

Yossi Adi … (voir 1 de plus)

Mirco Ravanelli

Discrete audio tokens are compact representations that aim to preserve perceptual quality, phonetic content, and speaker characteristics whi… (voir plus)le enabling efficient storage and inference, as well as competitive performance across diverse downstream tasks. They provide a practical alternative to continuous features, enabling the integration of speech and audio into modern large language models (LLMs). As interest in token-based audio processing grows, various tokenization methods have emerged, and several surveys have reviewed the latest progress in the field. However, existing studies often focus on specific domains or tasks and lack a unified comparison across various benchmarks. This paper presents a systematic review and benchmark of discrete audio tokenizers, covering three domains: speech, music, and general audio. We propose a taxonomy of tokenization approaches based on encoder-decoder, quantization techniques, training paradigm, streamability, and application domains. We evaluate tokenizers on multiple benchmarks for reconstruction, downstream performance, and acoustic language modeling, and analyze trade-offs through controlled ablation studies. Our findings highlight key limitations, practical considerations, and open challenges, providing insight and guidance for future research in this rapidly evolving area. For more information, including our main results and tokenizer database, please refer to our website: https://poonehmousavi.github.io/dates-website/.

2025-06-12

ArXiv (prépublication)

Discrete Audio Tokens: More Than a Survey!

Gallil Maimon

Adel Moumen

Darius Petermann

Jiatong Shi

Haibin Wu

Haici Yang

Anastasia Kuznetsova

Artem Ploujnikov

Bhuvana Ramabhadran

Benjamin Elizalde

Jinyu Li

Phil Woodland

Minje Kim

Hung-yi Lee

Shinji Watanabe

Yossi Adi … (voir 1 de plus)

Mirco Ravanelli

Discrete audio tokens are compact representations that aim to preserve perceptual quality, phonetic content, and speaker characteristics whi… (voir plus)le enabling efficient storage and inference, as well as competitive performance across diverse downstream tasks.They provide a practical alternative to continuous features, enabling the integration of speech and audio into modern large language models (LLMs). As interest in token-based audio processing grows, various tokenization methods have emerged, and several surveys have reviewed the latest progress in the field. However, existing studies often focus on specific domains or tasks and lack a unified comparison across various benchmarks. This paper presents a systematic review and benchmark of discrete audio tokenizers, covering three domains: speech, music, and general audio. We propose a taxonomy of tokenization approaches based on encoder-decoder, quantization techniques, training paradigm, streamability, and application domains. We evaluate tokenizers on multiple benchmarks for reconstruction, downstream performance, and acoustic language modeling, and analyze trade-offs through controlled ablation studies. Our findings highlight key limitations, practical considerations, and open challenges, providing insight and guidance for future research in this rapidly evolving area. For more information, including our main results and tokenizer database, please refer to our website: https://poonehmousavi.github.io/dates-website/.

2025-06-12

ArXiv (prépublication)

LiSTEN: Learning Soft Token Embeddings for Neural Audio LLMs

Mirco Ravanelli

Foundation models based on large language models (LLMs) have shown great success in handling various tasks and modalities. However, adapting… (voir plus) these models for general-purpose audio-language tasks is challenging due to differences in acoustic environments and task variations. In this work, we introduce LiSTEN Learning Soft Token Embeddings for Neural Audio LLMs), a framework for adapting LLMs to speech and audio tasks. LiSTEN uses a dynamic prompt selection strategy with learnable key-value pairs, allowing the model to balance general and task-specific knowledge while avoiding overfitting in a multitask setting. Our approach reduces dependence on large-scale ASR or captioning datasets, achieves competitive performance with fewer trainable parameters, and simplifies training by using a single-stage process. Additionally, LiSTEN enhances interpretability by analyzing the diversity and overlap of selected prompts across different tasks.

2025-05-24

ArXiv (prépublication)