Demonstration venue · read-only. Every page can be browsed; the buttons that would change it are switched off. Create an account to run TaxoReview on your own data. A simulation from public publication records: the people listed were not invited and did not agree to review.
? Tutorial for this page

Changsheng Quan

dblp:237/3163 · DBLP profile ↗
← Back
8ranked-venue papers
5as first author
7since 2021 · last 2024
0000-0002-8864-3814ORCID · corroborated

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

Artificial intelligence and machine learning · 5 · 3 first-author · 4 since 2021Graphics, computer vision, multimedia, augmented reality and games · 4 · 3 first-author · 4 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.

Computer graphics and multimedia
2 papers
Audio and music processing · 100%
Artificial intelligence
2 papers
Speech recognition and synthesis · 50% Deep learning architectures and training · 50%

Topics — the 8 heaviest of 8, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing
speech enhancement
1.522024
SpatialNet: Extensively Learning Spatial Information for Multichannel Joint Speech Separation, Denoising and Dereverberation · IEEE ACM Trans. Audio Speech Lang. Process. 2024
RealMAN: A Real-Recorded and Annotated Microphone Array Dataset for Dynamic Speech Enhancement and Localization · NeurIPS 2024
Audio and music processing › speech enhancement
dereverberation
0.812024
SpatialNet: Extensively Learning Spatial Information for Multichannel Joint Speech Separation, Denoising and Dereverberation · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Audio and music processing › speech enhancement
multichannel speech enhancement
0.812024
RealMAN: A Real-Recorded and Annotated Microphone Array Dataset for Dynamic Speech Enhancement and Localization · NeurIPS 2024
Audio and music processing › source separation › speech separation
multichannel speech separation
0.812024
SpatialNet: Extensively Learning Spatial Information for Multichannel Joint Speech Separation, Denoising and Dereverberation · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Audio and music processing › speech enhancement
noise reduction
0.812024
SpatialNet: Extensively Learning Spatial Information for Multichannel Joint Speech Separation, Denoising and Dereverberation · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Audio and music processing
sound source localization
0.812024
RealMAN: A Real-Recorded and Annotated Microphone Array Dataset for Dynamic Speech Enhancement and Localization · NeurIPS 2024
Machine learning › Deep learning architectures and training
attention mechanism
0.212024
SpatialNet: Extensively Learning Spatial Information for Multichannel Joint Speech Separation, Denoising and Dereverberation · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Natural language and speech › Speech recognition and synthesis
speech corpus
0.212024
RealMAN: A Real-Recorded and Annotated Microphone Array Dataset for Dynamic Speech Enhancement and Localization · NeurIPS 2024

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

temporal convolutional network · 1.5self-attention · 1.5narrow-band and cross-band processing · 1.5microphone array processing · 1.5deep learning · 1.5STFT · 1.5
YearPublicationVenuePosition
2024 RealMAN: A Real-Recorded and Annotated Microphone Array Dataset for Dynamic Speech Enhancement and Localization
abstract
The training of deep learning-based multichannel speech enhancement and source localization systems relies heavily on the simulation of room impulse response and multichannel diffuse noise, due to the lack of large-scale real-recorded datasets. However, the acoustic mismatch between simulated and real-world data could degrade the model performance when applying in real-world scenarios. To bridge this simulation-to-real gap, this paper presents a new relatively large-scale Real-recorded and annotated Microphone Array speech&Noise (RealMAN) dataset. The proposed dataset is valuable in two aspects: 1) benchmarking speech enhancement and localization algorithms in real scenarios; 2) offering a substantial amount of real-world training data for potentially improving the performance of real-world applications. Specifically, a 32-channel array with high-fidelity microphones is used for recording. A loudspeaker is used for playing source speech signals (about 35 hours of Mandarin speech). A total of 83.7 hours of speech signals (about 48.3 hours for static speaker and 35.4 hours for moving speaker) are recorded in 32 different scenes, and 144.5 hours of background noise are recorded in 31 different scenes. Both speech and noise recording scenes cover various common indoor, outdoor, semi-outdoor and transportation environments, which enables the training of general-purpose speech enhancement and source localization networks. To obtain the task-specific annotations, speaker location is annotated with an omni-directional fisheye camera by automatically detecting the loudspeaker. The direct-path signal is set as the target clean speech for speech enhancement, which is obtained by filtering the source speech signal with an estimated direct-path propagation filter. Baseline experiments demonstrate that i) compared to using simulated data, the proposed dataset is indeed able to train better speech enhancement and source localization networks; ii) using various sub-arrays of the proposed 32-channel microphone array can successfully train variable-array networks that can be directly used to unseen arrays.
Changsheng Quan, Nian Shao, Hui Bu
NeurIPS2
2024 Multichannel Long-Term Streaming Neural Speech Enhancement for Static and Moving Speakers
abstract
In this work, we extend our previously proposed offline SpatialNet for long-term streaming multichannel speech enhancement in both static and moving speaker scenarios. SpatialNet exploits spatial information, such as the spatial/steering direction of speech, for discriminating between target speech and interferences, and achieved outstanding performance. The core of SpatialNet is a narrow-band self-attention module used for learning the temporal dynamic of spatial vectors. Towards long-term streaming speech enhancement, we propose to replace the offline self-attention network with online networks that have linear inference complexity w.r.t signal length and meanwhile maintain the capability of learning long-term information. Three variants are developed based on i) masked self-attention, ii) Retention, a self-attention variant with linear inference complexity, and iii) Mamba, a structured-state-space-based RNN-like network. Moreover, we investigate the length extrapolation ability of different networks, namely test on signals that are much longer than training signals, and propose a short-signal training plus long-signal fine-tuning strategy, which largely improves the length extrapolation ability of the networks within limited training time. Overall, the proposed online SpatialNet achieves outstanding speech enhancement performance for long audio streams, and for both static and moving speakers.
Changsheng Quan, Xiaofei Li 0001
IEEE Signal Process. Lett.1
2024 SpatialNet: Extensively Learning Spatial Information for Multichannel Joint Speech Separation, Denoising and Dereverberation
abstract
This work proposes a neural network to extensively exploit spatial information for multichannel joint speech separation, denoising and dereverberation, named SpatialNet. In the short-time Fourier transform (STFT) domain, the proposed network performs end-to-end speech enhancement. It is mainly composed of interleaved narrow-band and cross-band blocks to respectively exploit narrow-band and cross-band spatial information. The narrow-band blocks process frequencies independently, and use self-attention mechanism and temporal convolutional layers to respectively perform spatial-feature-based speaker clustering and temporal smoothing/filtering. The cross-band blocks process frames independently, and use full-band linear layer and frequency convolutional layers to respectively learn the correlation between all frequencies and adjacent frequencies. Experiments are conducted on various simulated and real datasets, and the results show that 1) the proposed network achieves the state-of-the-art performance on almost all tasks; 2) the proposed network suffers little from the spectral generalization problem; and 3) the proposed network is indeed performing speaker clustering (demonstrated by attention maps).
Changsheng Quan, Xiaofei Li 0001
IEEE ACM Trans. Audio Speech Lang. Process.1
2023 MCNET: Fuse Multiple Cues for Multichannel Speech Enhancement
abstract
In multichannel speech enhancement, both spectral and spatial information are vital for discriminating between speech and noise. How to fully exploit these two types of information and their temporal dynamics remains an interesting research problem. As a solution to this problem, this paper proposes a multi-cue fusion network named McNet, which cascades four modules to respectively exploit the full-band spatial, narrowband spatial, sub-band spectral, and full-band spectral information. Experiments show that each module in the proposed network has its unique contribution and, as a whole, notably outperforms other state-of-the-art methods.
Changsheng Quan
ICASSP2
2022 Multi-Channel Narrow-Band Deep Speech Separation with Full-Band Permutation Invariant Training
abstract
This paper addresses the problem of multi-channel multi-speech separation based on deep learning techniques. In the short time Fourier transform domain, we propose an end-to-end narrow-band network that directly takes as input the multi-channel mixture signals of one frequency, and outputs the separated signals of this frequency. In narrow-band, the spatial information (or inter-channel difference) can well discriminate between speakers at different positions. This information is intensively used in many narrow-band speech separation methods, such as beamforming and clustering of spatial vectors. The proposed network is trained to learn a rule to automatically exploit this information and perform speech separation. Such a rule should be valid for any frequency, thence the network is shared by all frequencies. In addition, a full-band permutation invariant training criterion is proposed to solve the frequency permutation problem encountered by most narrow-band methods. Experiments show that, by focusing on deeply learning the narrow-band information, the proposed method outperforms the oracle beamforming method and the state-of-the-art deep learning based method.
Changsheng Quan
ICASSP1
2022 Multichannel Speech Separation with Narrow-band Conformer
abstract
This work proposes a multichannel speech separation method with narrow-band Conformer (named NBC).The network is trained to learn to automatically exploit narrow-band speech separation information, such as spatial vector clustering of multiple speakers.Specifically, in the short-time Fourier transform (STFT) domain, the network processes each frequency independently, and is shared by all frequencies.For one frequency, the network inputs the STFT coefficients of multichannel mixture signals, and predicts the STFT coefficients of separated speech signals.Clustering of spatial vectors shares a similar principle with the self-attention mechanism in the sense of computing the similarity of vectors and then aggregating similar vectors.Therefore, Conformer would be especially suitable for the present problem.Experiments show that the proposed narrowband Conformer achieves better speech separation performance than other state-of-the-art methods by a large margin.
Changsheng Quan
INTERSPEECH1
2021 A local search method based on edge age strategy for minimum vertex cover problem in massive graphs
Changsheng Quan, Ping Guo 0003
Expert Syst. Appl.1
2019 UPSimulator: A general P system simulator
abstract
In membrane computing, researchers have designed various P systems to solve problems, for example, NP-hard problems. The research of verification methods and simulation tools for these P systems is one of the main topics in membrane computing. In this paper, we present a general-purpose P system simulator named UPSimulator for cell-like, tissue-like and neural-like P systems. UPSimulator can serve as both a simulator and a development library for P system models, which processes objects in the regions defined by membranes. UPSimulator provides a feasible solution for the simulation of hybrid P systems by supporting the complex evolutionary rules and structures in different types of P systems.
Ping Guo 0003, Changsheng Quan, Lian Ye
Knowl. Based Syst.2