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Pooneh Mousavi

dblp:332/5839 · DBLP profile ↗
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5ranked-venue papers
2as first author
5since 2021 · last 2025
0009-0007-5761-5988ORCID · corroborated

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 4 · 2 first-author · 4 since 2021Graphics, computer vision, multimedia, augmented reality and games · 3 · 2 first-author · 3 since 2021

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Artificial intelligence
2 papers
Speech recognition and synthesis · 79% Learning paradigms · 16% Language models and text generation · 5%

Topics — the 6 heaviest of 7, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Natural language and speech › Speech recognition and synthesis
automatic speech recognition
0.812024
CL-MASR: A Continual Learning Benchmark for Multilingual ASR · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Machine learning › Learning paradigms
continual learning
0.812024
CL-MASR: A Continual Learning Benchmark for Multilingual ASR · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Natural language and speech › Speech recognition and synthesis › automatic speech recognition
multilingual speech recognition
0.812024
CL-MASR: A Continual Learning Benchmark for Multilingual ASR · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Natural language and speech › Speech recognition and synthesis
speaker recognition
0.812024
Open-Source Conversational AI with SpeechBrain 1.0 · J. Mach. Learn. Res. 2024
Natural language and speech › Speech recognition and synthesis
speech enhancement
0.812024
Open-Source Conversational AI with SpeechBrain 1.0 · J. Mach. Learn. Res. 2024
Natural language and speech › Language models and text generation › large language model
large language model integration
0.212024
Open-Source Conversational AI with SpeechBrain 1.0 · J. Mach. Learn. Res. 2024

Methods — techniques the papers use, named apart from their topics

pytorch · 0.8decoding strategies · 0.8continual learning methods · 0.8catastrophic forgetting mitigation · 0.8
YearPublicationVenuePosition
2025 What Are They Doing? Joint Audio-Speech Co-Reasoning
abstract
In audio and speech processing, tasks usually focus on either the audio or speech modality, even when both sounds and human speech are present in the same audio clip. Recent Auditory Large Language Models (ALLMs) have made it possible to process audio and speech simultaneously within a single model, leading to further considerations of joint audio-speech tasks.In this paper, we establish a novel benchmark to investigate how well ALLMs can perform joint audio-speech processing. Specifically, we introduce Joint Audio-Speech Co-Reasoning (JASCO), a novel task that unifies audio and speech processing, strictly requiring co-reasoning across both modalities. We also release a scene-reasoning dataset called "What Are They Doing". Additionally, we provide deeper insights into the models’ behaviors by analyzing their dependence on each modality.
Yingzhi Wang 0002, Pooneh Mousavi, Artem Ploujnikov, Mirco Ravanelli
ICASSP2
2025 LiSTEN: Learning Soft Token Embeddings for Neural Audio LLMs
Pooneh Mousavi, Cem Subakan, Mirco Ravanelli
INTERSPEECH1
2024 How Should We Extract Discrete Audio Tokens from Self-Supervised Models?
abstract
Discrete audio tokens have recently gained attention for their potential to bridge the gap between audio and language processing.Ideal audio tokens must preserve content, paralinguistic elements, speaker identity, and many other audio details.Current audio tokenization methods fall into two categories: Semantic tokens, acquired through quantization of Self-Supervised Learning (SSL) models, and Neural compression-based tokens (codecs).Although previous studies have benchmarked codec models to identify optimal configurations, the ideal setup for quantizing pretrained SSL models remains unclear.This paper explores the optimal configuration of semantic tokens across discriminative and generative tasks.We propose a scalable solution to train a universal vocoder across multiple SSL layers.Furthermore, an attention mechanism is employed to identify task-specific influential layers, enhancing the adaptability and performance of semantic tokens in diverse audio applications.
Pooneh Mousavi, Jarod Duret, Mohamed Salah Zaïem, Luca Della Libera, Artem Ploujnikov, Cem Subakan, Mirco Ravanelli
INTERSPEECH1
2024 Open-Source Conversational AI with SpeechBrain 1.0
abstract
SpeechBrain is an open-source Conversational AI toolkit based on PyTorch, focused particularly on speech processing tasks such as speech recognition, speech enhancement, speaker recognition, text-to-speech, and much more. It promotes transparency and replicability by releasing both the pre-trained models and the complete recipes of code and algorithms required for training them. This paper presents SpeechBrain 1.0, a significant milestone in the evolution of the toolkit, which now has over 200 recipes for speech, audio, and language processing tasks, and more than 100 models available on Hugging Face. SpeechBrain 1.0 introduces new technologies to support diverse learning modalities, Large Language Model (LLM) integration, and advanced decoding strategies, along with novel models, tasks, and modalities. It also includes a new benchmark repository, offering researchers a unified platform for evaluating models across diverse tasks.
Mirco Ravanelli, Titouan Parcollet, Adel Moumen, Sylvain de Langen, Cem Subakan, Peter Plantinga, Yingzhi Wang 0002, Pooneh Mousavi, Luca Della Libera, Artem Ploujnikov, Francesco Paissan, Davide Borra, Mohamed Salah Zaïem, Zeyu Zhao 0004, Shucong Zhang, Georgios Karakasidis, Sung-Lin Yeh, Pierre Champion, Aku Rouhe, Rudolf Braun, Florian Mai, Juan Zuluaga-Gomez, Seyed Mahed Mousavi, Andreas Nautsch, Xuechen Liu 0001, Sangeet Sagar, Jarod Duret, Salima Mdhaffar, Gaëlle Laperrière, Mickael Rouvier, Renato De Mori, Yannick Estève
J. Mach. Learn. Res.8
2024 CL-MASR: A Continual Learning Benchmark for Multilingual ASR
abstract
Modern multilingual automatic speech recognition (ASR) systems like Whisper have made it possible to transcribe audio in multiple languages with a single model. However, current state-of-the-art ASR models are typically evaluated on individual languages or in a multi-task setting, overlooking the challenge of continually learning new languages. There is insufficient research on how to add new languages without losing valuable information from previous data. Furthermore, existing continual learning benchmarks focus mostly on vision and language tasks, leaving continual learning for multilingual ASR largely unexplored. To bridge this gap, we propose CL-MASR, a benchmark designed for studying multilingual ASR in a continual learning setting. CL-MASR provides a diverse set of continual learning methods implemented on top of large-scale pretrained ASR models, along with common metrics to assess the effectiveness of learning new languages while addressing the issue of catastrophic forgetting. To the best of our knowledge, CL-MASR is the first continual learning benchmark for the multilingual ASR task.
Luca Della Libera, Pooneh Mousavi, Mohamed Salah Zaïem, Cem Subakan, Mirco Ravanelli
IEEE ACM Trans. Audio Speech Lang. Process.2