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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

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24 papers · 1 filter

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…

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…

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…

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…

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…

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…

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…

eess.AS2024

LoRA-Whisper: Parameter-Efficient and Extensible Multilingual ASR

Zheshu Song, Jianheng Zhuo, Yifan Yang +3

Recent years have witnessed significant progress in multilingual automatic speech recognition (ASR), driven by the emergence of end-to-end (E2E) models and the scaling of multiling…

eess.AS2024★ 2 cited

GigaSpeech 2: An Evolving, Large-Scale and Multi-domain ASR Corpus for Low-Resource Languages with Automated Crawling, Transcription and Refinement

Yifan Yang, Zheshu Song, Jianheng Zhuo +13

The evolution of speech technology has been spurred by the rapid increase in dataset sizes. Traditional speech models generally depend on a large amount of labeled training data, w…

eess.AS2024★ 10 cited

VALL-T: Decoder-Only Generative Transducer for Robust and Decoding-Controllable Text-to-Speech

Chenpeng Du, Yiwei Guo, Hankun Wang +6

Recent TTS models with decoder-only Transformer architecture, such as SPEAR-TTS and VALL-E, achieve impressive naturalness and demonstrate the ability for zero-shot adaptation give…

eess.AS2023★ 28 cited

Zipformer: A faster and better encoder for automatic speech recognition

Zengwei Yao, Liyong Guo, Xiaoyu Yang +6

The Conformer has become the most popular encoder model for automatic speech recognition (ASR). It adds convolution modules to a transformer to learn both local and global dependen…

eess.AS2023★ 1 cited

Libriheavy: a 50,000 hours ASR corpus with punctuation casing and context

Wei Kang, Xiaoyu Yang, Zengwei Yao +5

In this paper, we introduce Libriheavy, a large-scale ASR corpus consisting of 50,000 hours of read English speech derived from LibriVox. To the best of our knowledge, Libriheavy i…

eess.AS2023

Incorporating Class-based Language Model for Named Entity Recognition in Factorized Neural Transducer

Peng Wang, Yifan Yang, Zheng Liang +3

Despite advancements of end-to-end (E2E) models in speech recognition, named entity recognition (NER) is still challenging but critical for semantic understanding. Previous studies…

eess.AS2023

PromptASR for contextualized ASR with controllable style

Xiaoyu Yang, Wei Kang, Zengwei Yao +5

Prompts are crucial to large language models as they provide context information such as topic or logical relationships. Inspired by this, we propose PromptASR, a framework that in…

eess.AS2023

Towards Universal Speech Discrete Tokens: A Case Study for ASR and TTS

Yifan Yang, Feiyu Shen, Chenpeng Du +4

Self-supervised learning (SSL) proficiency in speech-related tasks has driven research into utilizing discrete tokens for speech tasks like recognition and translation, which offer…

eess.AS2023

Blank-regularized CTC for Frame Skipping in Neural Transducer

Yifan Yang, Xiaoyu Yang, Liyong Guo +6

Neural Transducer and connectionist temporal classification (CTC) are popular end-to-end automatic speech recognition systems. Due to their frame-synchronous design, blank symbols…

eess.AS2023

Delay-penalized CTC implemented based on Finite State Transducer

Zengwei Yao, Wei Kang, Fangjun Kuang +5

Connectionist Temporal Classification (CTC) suffers from the latency problem when applied to streaming models. We argue that in CTC lattice, the alignments that can access more fut…