activity
20182025
most citedQwen-Audio: Advancing Universal Audio Understanding via Unified Large-Scale Audio-Language Models

23 citations · 137 across the 28 of their papers we have counts for

collaborators

29 papers

eess.AS2025

OmniAudio: Generating Spatial Audio from 360-Degree Video

Huadai Liu, Tianyi Luo, Kaicheng Luo +11

Traditional video-to-audio generation techniques primarily focus on perspective video and non-spatial audio, often missing the spatial cues necessary for accurately representing so…

cs.SD2025

SyncSpeech: Efficient and Low-Latency Text-to-Speech based on Temporal Masked Transformer

Zhengyan Sheng, Zhihao Du, Shiliang Zhang +2

Current text-to-speech (TTS) models face a persistent limitation: autoregressive (AR) models suffer from low generation efficiency, while modern non-autoregressive (NAR) models exp…

cs.CL2025★ 3 cited

MinMo: A Multimodal Large Language Model for Seamless Voice Interaction

Qian Chen, Yafeng Chen, Yanni Chen +33

Recent advancements in large language models (LLMs) and multimodal speech-text models have laid the groundwork for seamless voice interactions, enabling real-time, natural, and hum…

cs.SD2024★ 4 cited

CosyVoice 2: Scalable Streaming Speech Synthesis with Large Language Models

Zhihao Du, Yuxuan Wang, Qian Chen +16

In our previous work, we introduced CosyVoice, a multilingual speech synthesis model based on supervised discrete speech tokens. By employing progressive semantic decoding with two…

eess.AS2024

Are Transformers in Pre-trained LM A Good ASR Encoder? An Empirical Study

Keyu An, Shiliang Zhang, Zhijie Yan

In this study, we delve into the efficacy of transformers within pre-trained language models (PLMs) when repurposed as encoders for Automatic Speech Recognition (ASR). Our underlyi…

cs.SD2024★ 9 cited

FunAudioLLM: Voice Understanding and Generation Foundation Models for Natural Interaction Between Humans and LLMs

Keyu An, Qian Chen, Chong Deng +30

This report introduces FunAudioLLM, a model family designed to enhance natural voice interactions between humans and large language models (LLMs). At its core are two innovative mo…

cs.SD2024★ 4 cited

CosyVoice: A Scalable Multilingual Zero-shot Text-to-speech Synthesizer based on Supervised Semantic Tokens

Zhihao Du, Qian Chen, Shiliang Zhang +9

Recent years have witnessed a trend that large language model (LLM) based text-to-speech (TTS) emerges into the mainstream due to their high naturalness and zero-shot capacity. In…

eess.AS2023★ 23 cited

Qwen-Audio: Advancing Universal Audio Understanding via Unified Large-Scale Audio-Language Models

Yunfei Chu, Jin Xu, Xiaohuan Zhou +5

Recently, instruction-following audio-language models have received broad attention for audio interaction with humans. However, the absence of pre-trained audio models capable of h…

cs.SD2023★ 1 cited

Advancing VAD Systems Based on Multi-Task Learning with Improved Model Structures

Lingyun Zuo, Keyu An, Shiliang Zhang +1

In a speech recognition system, voice activity detection (VAD) is a crucial frontend module. Addressing the issues of poor noise robustness in traditional binary VAD systems based…

cs.SD2023★ 1 cited

The second multi-channel multi-party meeting transcription challenge (M2MeT) 2.0): A benchmark for speaker-attributed ASR

Yuhao Liang, Mohan Shi, Fan Yu +11

With the success of the first Multi-channel Multi-party Meeting Transcription challenge (M2MeT), the second M2MeT challenge (M2MeT 2.0) held in ASRU2023 particularly aims to tackle…

cs.SD2023★ 18 cited

LauraGPT: Listen, Attend, Understand, and Regenerate Audio with GPT

Zhihao Du, Jiaming Wang, Qian Chen +12

Generative Pre-trained Transformer (GPT) models have achieved remarkable performance on various natural language processing tasks, and have shown great potential as backbones for a…

cs.SD2023

Accurate and Reliable Confidence Estimation Based on Non-Autoregressive End-to-End Speech Recognition System

Xian Shi, Haoneng Luo, Zhifu Gao +2

Estimating confidence scores for recognition results is a classic task in ASR field and of vital importance for kinds of downstream tasks and training strategies. Previous end-to-e…

cs.CL2023★ 1 cited

MUG: A General Meeting Understanding and Generation Benchmark

Qinglin Zhang, Chong Deng, Jiaqing Liu +7

Listening to long video/audio recordings from video conferencing and online courses for acquiring information is extremely inefficient. Even after ASR systems transcribe recordings…

cs.CL2023

Overview of the ICASSP 2023 General Meeting Understanding and Generation Challenge (MUG)

Qinglin Zhang, Chong Deng, Jiaqing Liu +7

ICASSP2023 General Meeting Understanding and Generation Challenge (MUG) focuses on prompting a wide range of spoken language processing (SLP) research on meeting transcripts, as SL…

cs.SD2023

Achieving Timestamp Prediction While Recognizing with Non-Autoregressive End-to-End ASR Model

Xian Shi, Yanni Chen, Shiliang Zhang +1

Conventional ASR systems use frame-level phoneme posterior to conduct force-alignment~(FA) and provide timestamps, while end-to-end ASR systems especially AED based ones are short…

cs.MM2022★ 1 cited

MMSpeech: Multi-modal Multi-task Encoder-Decoder Pre-training for Speech Recognition

Xiaohuan Zhou, Jiaming Wang, Zeyu Cui +4

In this paper, we propose a novel multi-modal multi-task encoder-decoder pre-training framework (MMSpeech) for Mandarin automatic speech recognition (ASR), which employs both unlab…

cs.SD2022★ 1 cited

Speaker Overlap-aware Neural Diarization for Multi-party Meeting Analysis

Zhihao Du, Shiliang Zhang, Siqi Zheng +1

Recently, hybrid systems of clustering and neural diarization models have been successfully applied in multi-party meeting analysis. However, current models always treat overlapped…

cs.SD2022★ 2 cited

Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition

Zhifu Gao, Shiliang Zhang, Ian McLoughlin +1

Transformers have recently dominated the ASR field. Although able to yield good performance, they involve an autoregressive (AR) decoder to generate tokens one by one, which is com…

cs.SD2022★ 1 cited

Speaker Embedding-aware Neural Diarization: an Efficient Framework for Overlapping Speech Diarization in Meeting Scenarios

Zhihao Du, Shiliang Zhang, Siqi Zheng +1

Overlapping speech diarization has been traditionally treated as a multi-label classification problem. In this paper, we reformulate this task as a single-label prediction problem…

cs.SD2022

Summary On The ICASSP 2022 Multi-Channel Multi-Party Meeting Transcription Grand Challenge

Fan Yu, Shiliang Zhang, Pengcheng Guo +13

The ICASSP 2022 Multi-channel Multi-party Meeting Transcription Grand Challenge (M2MeT) focuses on one of the most valuable and the most challenging scenarios of speech technologie…

eess.AS2022

ProsoSpeech: Enhancing Prosody With Quantized Vector Pre-training in Text-to-Speech

Yi Ren, Ming Lei, Zhiying Huang +4

Expressive text-to-speech (TTS) has become a hot research topic recently, mainly focusing on modeling prosody in speech. Prosody modeling has several challenges: 1) the extracted p…

cs.SD2021★ 11 cited

M2MeT: The ICASSP 2022 Multi-Channel Multi-Party Meeting Transcription Challenge

Fan Yu, Shiliang Zhang, Yihui Fu +9

Recent development of speech processing, such as speech recognition, speaker diarization, etc., has inspired numerous applications of speech technologies. The meeting scenario is o…

cs.SD2021

BeamTransformer: Microphone Array-based Overlapping Speech Detection

Siqi Zheng, Shiliang Zhang, Weilong Huang +5

We propose BeamTransformer, an efficient architecture to leverage beamformer's edge in spatial filtering and transformer's capability in context sequence modeling. BeamTransformer…

cs.SD2021★ 17 cited

A Real-time Speaker Diarization System Based on Spatial Spectrum

Siqi Zheng, Weilong Huang, Xianliang Wang +3

In this paper we describe a speaker diarization system that enables localization and identification of all speakers present in a conversation or meeting. We propose a novel systema…