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

Taihui Wang

dblp:314/1380 · DBLP profile ↗
← Back
3ranked-venue papers
3as first author
3since 2021 · last 2024
0000-0003-3324-6878ORCID · corroborated

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

Artificial intelligence and machine learning · 3 · 3 first-author · 3 since 2021Graphics, computer vision, multimedia, augmented reality and games · 1 · 1 first-author · 1 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 · 96% Image and video processing · 4%

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

TopicWeightPapersLastEvidence papers
Audio and music processing › speech production
source-filter model
1.322024
Multichannel Linear Prediction-Based Speech Dereverberation Considering Sparse and Low-Rank Priors · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Convolutive Transfer Function-Based Multichannel Nonnegative Matrix Factorization for Overdetermined Blind Source Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2022
Audio and music processing › speech enhancement
dereverberation
0.812024
Multichannel Linear Prediction-Based Speech Dereverberation Considering Sparse and Low-Rank Priors · IEEE ACM Trans. Audio Speech Lang. Process. 2024
Audio and music processing › source separation
blind source separation
0.612022
Convolutive Transfer Function-Based Multichannel Nonnegative Matrix Factorization for Overdetermined Blind Source Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2022
Audio and music processing › source separation › blind source separation
multichannel blind source separation
0.612022
Convolutive Transfer Function-Based Multichannel Nonnegative Matrix Factorization for Overdetermined Blind Source Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2022
Audio and music processing › source separation
nonnegative matrix factorization for source separation
0.612022
Convolutive Transfer Function-Based Multichannel Nonnegative Matrix Factorization for Overdetermined Blind Source Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2022
Image and video processing
spatial modeling
0.212022
Convolutive Transfer Function-Based Multichannel Nonnegative Matrix Factorization for Overdetermined Blind Source Separation · IEEE ACM Trans. Audio Speech Lang. Process. 2022

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

maximum likelihood estimation · 0.8majorization-minimization · 0.8multiplicative update · 0.6multichannel wiener filter · 0.6iterative projection · 0.6
YearPublicationVenuePosition
2024 Multichannel Linear Prediction-Based Speech Dereverberation Considering Sparse and Low-Rank Priors
abstract
This article addresses the multi-channel linear prediction (MCLP)-based speech dereverberation problem by jointly considering the sparsity and low-rank priors of speech spectrograms. We utilize the complex generalized Gaussian (CGG) distribution as the source model and the generalized nonnegative matrix factorization (NMF) as the spectral model. The difference between the presented model and existing ones for MCLP is twofold. First, we adopt the CGG distribution with a time-frequency-variant scale parameter instead of that with a time-frequency-invariant scale parameter. Second, the time-frequency-varying scale parameter is approximated by NMF in a low-rank manner. Based on the maximum-likelihood criterion, speech dereverberation is formulated as an optimization problem that minimizes the prediction error weighted by the reciprocal of sparse and low-rank parameters. A convergence-guaranteed algorithm is derived to estimate the parameters using the majorization-minimization technology. The WPE, NMF-based WPE and CGG-based WPE can be treated as special cases of the proposed method with different shape and domain parameters. As a byproduct, the proposed method provides a simple and elegant way to derive the CGG-based WPE algorithm. A series of experiments show the superiority of the proposed method over WPE, NMF-based WPE and CGG-based WPE methods.
Taihui Wang, Feiran Yang 0001, Jun Yang 0004
IEEE ACM Trans. Audio Speech Lang. Process.1
2022 Convolutive Transfer Function-Based Multichannel Nonnegative Matrix Factorization for Overdetermined Blind Source Separation
abstract
Most multichannel blind source separation (BSS) approaches rely on a spatial model to encode the transfer functions from sources to microphones and a source model to encode the source power spectral density. The rank-1 spatial model has been widely exploited in independent component analysis (ICA), independent vector analysis (IVA), and independent low-rank matrix analysis (ILRMA). The full-rank spatial model is also considered in many BSS approaches, such as full-rank spatial covariance matrix analysis (FCA), multichannel nonnegative matrix factorization (MNMF), and FastMNMF, which can improve the separation performance in the case of long reverberation times. This paper proposes a new MNMF framework based on the convolutive transfer function (CTF) for overdetermined BSS. The time-domain convolutive mixture model is approximated by a frequency-wise convolutive mixture model instead of the widely adopted frequency-wise instantaneous mixture model. The iterative projection algorithm is adopted to estimate the demixing matrix, and the multiplicative update rule is employed to estimate nonnegative matrix factorization (NMF) parameters. Finally, the source image is reconstructed using a multichannel Wiener filter. The advantages of the proposed method are twofold. First, the CTF approximation enables us to use a short window to represent long impulse responses. Second, the full-rank spatial model can be derived based on the CTF approximation and slowly time-variant source variances, and close relationships between the proposed method and ILRMA, FCA, MNMF and FastMNMF are revealed. Extensive experiments show that the proposed algorithm achieves a higher separation performance than ILRMA and FastMNMF in reverberant environments.
Taihui Wang, Feiran Yang 0001, Jun Yang 0004
IEEE ACM Trans. Audio Speech Lang. Process.1
2021 Real-Time Independent Vector Analysis Using Semi-Supervised Nonnegative Matrix Factorization as a Source Model
Taihui Wang, Feiran Yang 0001, Jun Yang 0004
Interspeech1