activity
20232026
most citedZipformer: A faster and better encoder for automatic speech recognition

28 citations · 49 across the 34 of their papers we have counts for

collaborators

35 papers

eess.AS2026

GigaSpeechBench: A Real-World Multilingual Speech-to-Text Benchmark

Yujie Tu, Yifan Yang, Tianrui Wang +35

While modern ASR systems achieve low error rates on high-resource benchmarks, such performance often overestimates real-world robustness. Existing evaluations address challenges in…

cs.SD2026

MMAE: A Massive Multitask Audio Editing Benchmark

Ziyang Ma, Ruiqi Yan, Ruiyang Xu +35

We introduce MMAE, a Massive Multitask Audio Editing benchmark, serving as the first comprehensive evaluation testbed designed for general-purpose instruction-based audio editing.…

eess.AS2026

UAT: Unified Audio-Text Diffusion for Audio Generation, Editing, and Captioning

Hui Wang, Yifan Yang, Zeyue Tian +8

Audio generation and audio-to-text understanding remain largely separate, with diffusion models dominating high-fidelity synthesis and autoregressive (AR) language models driving c…

eess.AS2026

WavCube: Unifying Speech Representation for Understanding and Generation via Semantic-Acoustic Joint Modeling

Guanrou Yang, Tian Tan, Qian Chen +12

Integrating speech understanding and generation is a pivotal step toward building unified speech models. However, the different representations required for these two tasks current…

eess.AS2026

Representation-Regularized Convolutional Audio Transformer for Audio Understanding

Bing Han, Chushu Zhou, Yifan Yang +4

Bootstrap-based Self-Supervised Learning (SSL) has achieved remarkable progress in audio understanding. However, existing methods typically operate at a single level of granularity…

cs.SD2026★ 3 cited

SLAM-LLM: A Modular, Open-Source Multimodal Large Language Model Framework and Best Practice for Speech, Language, Audio and Music Processing

Ziyang Ma, Guanrou Yang, Wenxi Chen +19

The recent surge in open-source Multimodal Large Language Models (MLLM) frameworks, such as LLaVA, provides a convenient kickoff for artificial intelligence developers and research…

eess.AS2026

Towards Fine-Grained and Multi-Granular Contrastive Language-Speech Pre-training

Yifan Yang, Bing Han, Hui Wang +8

Modeling fine-grained speaking styles remains challenging for language-speech representation pre-training, as existing speech-text models are typically trained with coarse captions…

eess.AS2025★ 1 cited

SPEAR: A Unified SSL Framework for Learning Speech and Audio Representations

Xiaoyu Yang, Yifan Yang, Zengrui Jin +5

Self-supervised learning (SSL) has significantly advanced acoustic representation learning. However, most existing models are optimised for either speech or audio event understandi…

cs.SD2025

SpeechLLM-as-Judges: Towards General and Interpretable Speech Quality Evaluation

Hui Wang, Jinghua Zhao, Yifan Yang +9

Generative speech technologies are progressing rapidly, but evaluating the perceptual quality of synthetic speech remains a core challenge. Existing methods typically rely on scala…

eess.AS2025

Position: Towards Responsible Evaluation for Text-to-Speech

Yifan Yang, Hui Wang, Bing Han +4

Recent advances in text-to-speech (TTS) technology have enabled systems to generate speech that is often indistinguishable from human speech, bringing benefits to accessibility, co…

eess.AS2025

Measuring Prosody Diversity in Zero-Shot TTS: A New Metric, Benchmark, and Exploration

Yifan Yang, Bing Han, Hui Wang +5

Prosody diversity is essential for achieving naturalness and expressiveness in zero-shot text-to-speech (TTS). However, frequently used acoustic metrics capture only partial views…

cs.SD2025

StreamMel: Real-Time Zero-shot Text-to-Speech via Interleaved Continuous Autoregressive Modeling

Hui Wang, Yifan Yang, Shujie Liu +7

Recent advances in zero-shot text-to-speech (TTS) synthesis have achieved high-quality speech generation for unseen speakers, but most systems remain unsuitable for real-time appli…

cs.LG2025

Zero-Shot Streaming Text to Speech Synthesis with Transducer and Auto-Regressive Modeling

Haiyang Sun, Shujie Hu, Shujie Liu +8

Zero-shot streaming text-to-speech is an important research topic in human-computer interaction. Existing methods primarily use a lookahead mechanism, relying on future text to ach…

eess.AS2025

VietASR: Achieving Industry-level Vietnamese ASR with 50-hour labeled data and Large-Scale Speech Pretraining

Jianheng Zhuo, Yifan Yang, Yiwen Shao +4

Automatic speech recognition (ASR) has made remarkable progress but heavily relies on large-scale labeled data, which is scarce for low-resource languages like Vietnamese. While ex…

eess.AS2025

EmoVoice: LLM-based Emotional Text-To-Speech Model with Freestyle Text Prompting

Guanrou Yang, Chen Yang, Qian Chen +12

Human speech goes beyond the mere transfer of information; it is a profound exchange of emotions and a connection between individuals. While Text-to-Speech (TTS) models have made h…

eess.AS2025

Pseudo-Autoregressive Neural Codec Language Models for Efficient Zero-Shot Text-to-Speech Synthesis

Yifan Yang, Shujie Liu, Jinyu Li +10

Recent zero-shot text-to-speech (TTS) systems face a common dilemma: autoregressive (AR) models suffer from slow generation and lack duration controllability, while non-autoregress…

cs.CL2025

FELLE: Autoregressive Speech Synthesis with Token-Wise Coarse-to-Fine Flow Matching

Hui Wang, Shujie Liu, Lingwei Meng +9

To advance continuous-valued token modeling and temporal-coherence enforcement, we propose FELLE, an autoregressive model that integrates language modeling with token-wise flow mat…

eess.AS2024

SLAM-Omni: Timbre-Controllable Voice Interaction System with Single-Stage Training

Wenxi Chen, Ziyang Ma, Ruiqi Yan +13

Recent advancements highlight the potential of end-to-end real-time spoken dialogue systems, showcasing their low latency and high quality. In this paper, we introduce SLAM-Omni, a…

cs.AI2024

A Comparative Study of LLM-based ASR and Whisper in Low Resource and Code Switching Scenario

Zheshu Song, Ziyang Ma, Yifan Yang +2

Large Language Models (LLMs) have showcased exceptional performance across diverse NLP tasks, and their integration with speech encoder is rapidly emerging as a dominant trend in t…

eess.AS2024

Interleaved Speech-Text Language Models for Simple Streaming Text-to-Speech Synthesis

Yifan Yang, Shujie Liu, Jinyu Li +10

This paper introduces Interleaved Speech-Text Language Model (IST-LM) for zero-shot streaming Text-to-Speech (TTS). Unlike many previous approaches, IST-LM is directly trained on i…

eess.AS2024

k2SSL: A Faster and Better Framework for Self-Supervised Speech Representation Learning

Yifan Yang, Jianheng Zhuo, Zengrui Jin +9

Self-supervised learning (SSL) has achieved great success in speech-related tasks. While Transformer and Conformer architectures have dominated SSL backbones, encoders like Zipform…

cs.CL2024

Making LLMs Better Many-to-Many Speech-to-Text Translators with Curriculum Learning

Yexing Du, Youcheng Pan, Ziyang Ma +7

Multimodal Large Language Models (MLLMs) have achieved significant success in Speech-to-Text Translation (S2TT) tasks. While most existing research has focused on English-centric t…

cs.CL2024

Exploring SSL Discrete Speech Features for Zipformer-based Contextual ASR

Mingyu Cui, Yifan Yang, Jiajun Deng +7

Self-supervised learning (SSL) based discrete speech representations are highly compact and domain adaptable. In this paper, SSL discrete speech features extracted from WavLM model…

cs.SD2024

LibriheavyMix: A 20,000-Hour Dataset for Single-Channel Reverberant Multi-Talker Speech Separation, ASR and Speaker Diarization

Zengrui Jin, Yifan Yang, Mohan Shi +10

The evolving speech processing landscape is increasingly focused on complex scenarios like meetings or cocktail parties with multiple simultaneous speakers and far-field conditions…