何宏森｜西南科技大学｜学者主页

2026

Gudermannian-based Active Impulsive Noise Control Algorithm: Design and performance analysis

Yongcong Yuan, Yi Yu, Hongsen He, Ke Li, Rodrigo C. de Lamare

Mechanical Systems and Signal Processing

DOI

2026

Dynamic Step-Size FxNLMS Algorithm for Active Noise Control and Its DSP Implementation

Zhengwei Kong, Guofeng He, Yi Yu, Hongsen He, Jian Zhou

Circuits Systems and Signal Processing

DOI

2026

A Third-Order Cumulant Based Adaptive Beamforming Algorithm for Speech Enhancement

Juping Wang, Hongsen He, Yi Yu, Xiaoxia Li, Limin Zhang

Circuits Systems and Signal Processing

DOI

2026

An improved step-size converter based multi-sampled adaptive filter algorithm for vehicular handsfree echo cancellation

Zongxin Huang, Yi Yu, Yingsong Li, Hongsen He, Jingdong Chen, Jacob Benesty

Signal Processing

DOI

2026

Distributed Normalized Maximum Correntropy Algorithm with Step-Size Control

Hongshun Yu, Yi Yu, Hongsen He, Yuriy Zakharov, Rodrigo C. de Lamare

Proceedings of the IEEE Sensor Array and Multichannel Signal Processing Workshop

DOI

2026

Combined Step-Size Subspace Tracking Algorithm Based on Maximum Correntropy Criterion

Jin Wu, Yi Yu, Hongsen He, Rodrigo C. de Lamare, Ke Li

Proceedings of the IEEE Sensor Array and Multichannel Signal Processing Workshop

DOI

2026

Low-complexity decorrelation affine projection algorithms: Formulations and applications

Lin Luo, Yi Yu, Yuyu Zhu, Hongsen He, Rodrigo C. de Lamare

Journal of the Franklin Institute

DOI

2026

A Robust Low-Complexity Adaptive Filtering Algorithm with Dichotomous Coordinate Descent Scheme for Acoustic Echo Cancellation

Tingting Tao, Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

Circuits Systems and Signal Processing

DOI

2026

Transform-Domain Filtered-x Least Mean Square Algorithms with Switched Step-Size Mechanism for Feedforward ANC Systems

Zhiyuan Li, Yi Yu, Yuyu Zhu, Hongsen He, Rodrigo C. de Lamare

Circuits Systems and Signal Processing

DOI

2026

Switched Step-Size-Based Filtered-x NLMS Algorithm for Active Noise Control

Zhiyuan Li, Yi Yu, Hongsen He, Yuyu Zhu, Rodrigo C. de Lamare

IEEE Access

DOI

2026

A DCT-LMS Algorithm With Per-Coefficient Variable Step-Sizes

Zhiyuan Li, Yi Yu, Hongsen He, Yuyu Zhu, Rodrigo C. de Lamare

IEEE Signal Processing Letters

DOI

2026

Generalized Correntropy Subspace Tracking for Robust DoA Estimation

Jin Wu, Yi Yu, Hongsen He, Rodrigo C. de Lamare, Ke Li

IEEE Signal Processing Letters

DOI

2025

A third-order tensor decomposition approach for adaptive speech dereverberation

Zhiheng Wang, Zhu Zhen, Xiaojin Zeng, Hongsen He, Yi Yu, Tao Yang

NOISE-CON proceedings

DOI

2025

A trade-off pre-whitening method based on piecewise bilinear prediction for speaker localization

Jing Liu, Hongsen He, Yi Yu, Xueyuan Wang, Zhiheng Wang

NOISE-CON proceedings

DOI

2025

Robust regularized blind system identification with application to adaptive speech dereverberation

Zhimin Qiu, Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

The Journal of the Acoustical Society of America

DOI

2025

Widely Linear Complex-Valued Affine Projection Algorithm With a Sliding-Window Step-Size

Lin Luo, Yi Yu, Hongsen He, Tao Yu, Rodrigo C. de Lamare

IEEE Signal Processing Letters

DOI

2024

Novel normalized subband adaptive filtering algorithms with weights-dependent variable step-size

Ke Li, Yi Yu, Hongsen He, Tao Yu, Rodrigo C. de Lamare

Digital Signal Processing

DOI

2024

A Newton-Raphson solution to the exponentially weighted least M-estimate formulation for acoustic system identification

Li-Min Zhang, Hongsen He, Jingdong Chen, Yi Yu, Jacob Benesty

Applied Acoustics

DOI

2024

A low-complexity adaptive speech dereverberation approach based on multichannel linear prediction

Zhu Zhen, Hongsen He, Zhimin QIU, Yi YU

NOISE-CON proceedings

DOI

2024

Affine Projection Algorithms with Novel Schemes of Variable Projection Order

Lin Luo, Yi Yu, Tao Yang, Ke Li, Hongsen He

Circuits Systems and Signal Processing

DOI

2024

Normalized Multichannel Frequency-Domain LMS Filter With Nearest Kronecker Product Decomposition for Blind Identification of Low-Rank Acoustic Systems

Zhimin Qiu, Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

European Signal Processing Conference (EUSIPCO)

DOI

2024

Robust Widely linear complex-valued combined step-size algorithm for beamforming

Lin Luo, Peng Guo, Yi Yu, Hongsen He, Rodrigo C. de Lamare

International Symposium on Wireless Communication Systems

DOI

2024

A Steered Response Power Approach with Bilinear Prediction-Based Trade-Off Prewhitening for Speaker Localization

Zhiheng Wang, Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

IEEE International Conference on Acoustics Speech and Signal Processing

DOI

2024

Adaptive Subband Forward Blind Source Separation Algorithms Based on Kalman Mechanism

Jianhong Ye, Yi Yu, Yuriy V. Zakharov, Zhigang Liu, Hongsen He

IEEE Transactions on Instrumentation and Measurement

DOI

2023

An adaptive multichannel speech dereverberation approach based on robust estimation

Xiaojin Zeng, Hongsen He, Zhu Zhen

NOISE-CON proceedings

DOI

2023

Robust Proportionate Subband Adaptive Filter Algorithms With Optimal Variable Step-Size

Yi Yu, Jianhong Ye, Yuriy V. Zakharov, Hongsen He

IEEE Transactions on Circuits & Systems II Express Briefs

DOI

2023

A tensor decomposition based multichannel linear prediction approach to speech dereverberation

Xiaojin Zeng, Hongsen He, Jingdong Chen, Jacob Benesty

Applied Acoustics

DOI

2023

A Recursive Least M-Estimate Adaptive Algorithm with Low Complexity for Active Control of Impulsive Noises

Miaomiao Wang, Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

European Signal Processing Conference (EUSIPCO)

DOI

2023

Robust Time-Delay Estimation for Speaker Localization Using Mutual Information Among Multiple Microphone Signals

Juping Wang, Hongsen He, Yi Yu, Yingyue Zhou, Limin Zhang

IEEE Sensors Journal

DOI

2023

A Frequency-Domain Recursive Least-Squares Adaptive Filtering Algorithm Based On A Kronecker Product Decomposition

Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

IEEE International Conference on Acoustics Speech and Signal Processing

DOI

2023

A time delay estimation approach with low computational complexity for speaker localization

Bangguo Song, Hongsen He, Tao Yang

NOISE-CON proceedings

DOI

2022

General Robust Proportionate NSAF Algorithm With a Step-Size Converter

Zongxin Huang, Yi Yu, Rodrigo C. de Lamare, Hongsen He, Ke Li

IEEE Transactions on Circuits & Systems II Express Briefs

DOI

2022

General Robust Subband Adaptive Filtering for Echo Cancellation

Yi Yu, Hongsen He, Rodrigo C. de Lamare, Badong Chen

IEEE International Workshop on Machine Learning for Signal Processing (MLSP)

DOI

2022

A Step size Converter for Normalized Subband Adaptive Filtering Algorithm

Zongxin Huang, Yi Yu, Ke Li, Hongsen He

IEEE International Conference on Electronic Information and Communication Technology (ICEICT)

DOI

2022

Normalized Least Mean M-Estimate Algorithm with Switching Step-Sizes Against Impulsive Noises

Peng Guo, Yi Yu, Hongsen He, Ke Li, Tao Yu

Circuits Systems and Signal Processing

DOI

2022

Robust NLMS algorithms with combined step-size against impulsive noises

Peng Guo, Yi Yu, Tao Yang, Hongsen He, Rodrigo C. de Lamare

Digital Signal Processing

DOI

2022

Sparsity-Aware Robust Normalized Subband Adaptive Filtering Algorithms With Alternating Optimization of Parameters

Yi Yu, Zongxin Huang, Hongsen He, Yuriy V. Zakharov, Rodrigo C. de Lamare

IEEE Transactions on Circuits & Systems II Express Briefs

DOI

2022

General Robust Subband Adaptive Filtering: Algorithms and Applications

Yi Yu, Hongsen He, Rodrigo C. de Lamare, Badong Chen

IEEE/ACM Transactions on Audio Speech and Language Processing

DOI

2021

基于多分辨率特征和时频注意力的环境声音分类

刘慧，李小霞，何宏森

《计算机应用研究》

DOI

2021

Robust Recursive Least M-Estimate Adaptive Filter for the Identification of Low-Rank Acoustic Systems

Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

IEEE International Conference on Acoustics Speech and Signal Processing

DOI

2020

Robust Frequency-Domain Recursive Least M-Estimate Adaptive Filter For Acoustic System Identification

Hongsen He, Jingdong Chen, Jacob Benesty, Yi Yu

IEEE International Conference on Acoustics Speech and Signal Processing

DOI

2020

Diffusion Normalized Least Mean M-estimate Algorithms: Design and Performance Analysis

Yi Yu, Hongsen He, Tao Yang, Xueyuan Wang, Rodrigo C. de Lamare

IEEE Transactions on Signal Processing

DOI

2019

基于改进稀疏线性预测的时延估计算法

贺良，郭海燕，何宏森，王学渊

《传感器与微系统》

2019

A class of multichannel sparse linear prediction algorithms for time delay estimation of speech sources

Hongsen He, Jingdong Chen, Jacob Benesty, Wenxing Zhang, Tao Yang

Signal Processing

DOI

2019

M-Estimate Based Normalized Subband Adaptive Filter Algorithm: Performance Analysis and Improvements

Yi Yu, Hongsen He, Badong Chen, Jianghui Li, Youwen Zhang, Lu Lu

IEEE/ACM Transactions on Audio Speech and Language Processing

DOI

2018

Noise Robust Frequency-Domain Adaptive Blind Multichannel Identification With $\ell _p$-Norm Constraint

Hongsen He, Jingdong Chen, Jacob Benesty, Tao Yang

IEEE/ACM Transactions on Audio Speech and Language Processing

DOI

2017

An ℓ1/ℓ1-norm based linear prediction algorithm to time delay estimation via alternating direction method of multipliers

Hongsen He, Xueyuan Wang, Yingyue Zhou, Tao Yang

IEEE International Conference on Signal Processing, Communications and Computing (ICSPCC)

DOI

2016

Multichannel time delay estimation for acoustic source localization via robust adaptive blind system identification

Hongsen He, Jingdong Chen, Jacob Benesty, Tao Yang

DOI

2016

On directivity of a circular array with directional microphones

Hongsen He, Xiaojun Qiu, Tao Yang

DOI

2016

On time delay estimation based on multichannel spatiotemporal sparse linear prediction

Hongsen He, Jingdong Chen, Jacob Benesty, Tao Yang

IEEE International Conference on Acoustics Speech and Signal Processing

DOI

2013

基于传声器阵列的建模和定位算法研究

何宏森

《南京大学》

2012

Time Difference of Arrival Estimation Exploiting Multichannel Spatio-Temporal Prediction

Hongsen He, Lifu Wu, Jing Zhou Lu, Xiaojun Qiu, Jingdong Chen

IEEE Transactions on Audio Speech and Language Processing

DOI

2010

An Active Impulsive Noise Control Algorithm With Logarithmic Transformation

Lifu Wu, Hongsen He, Xiaojun Qiu

IEEE Transactions on Audio Speech and Language Processing

DOI

精选作品

期刊论文

Gudermannian-based Active Impulsive Noise Control Algorithm: Design and performance analysis

Dynamic Step-Size FxNLMS Algorithm for Active Noise Control and Its DSP Implementation

A Third-Order Cumulant Based Adaptive Beamforming Algorithm for Speech Enhancement

An improved step-size converter based multi-sampled adaptive filter algorithm for vehicular handsfree echo cancellation

Distributed Normalized Maximum Correntropy Algorithm with Step-Size Control

Combined Step-Size Subspace Tracking Algorithm Based on Maximum Correntropy Criterion

Low-complexity decorrelation affine projection algorithms: Formulations and applications

A Robust Low-Complexity Adaptive Filtering Algorithm with Dichotomous Coordinate Descent Scheme for Acoustic Echo Cancellation

Transform-Domain Filtered-x Least Mean Square Algorithms with Switched Step-Size Mechanism for Feedforward ANC Systems

Switched Step-Size-Based Filtered-x NLMS Algorithm for Active Noise Control

A DCT-LMS Algorithm With Per-Coefficient Variable Step-Sizes

Generalized Correntropy Subspace Tracking for Robust DoA Estimation

A third-order tensor decomposition approach for adaptive speech dereverberation

A trade-off pre-whitening method based on piecewise bilinear prediction for speaker localization

Robust regularized blind system identification with application to adaptive speech dereverberation

Widely Linear Complex-Valued Affine Projection Algorithm With a Sliding-Window Step-Size

Novel normalized subband adaptive filtering algorithms with weights-dependent variable step-size

A Newton-Raphson solution to the exponentially weighted least M-estimate formulation for acoustic system identification

A low-complexity adaptive speech dereverberation approach based on multichannel linear prediction

Affine Projection Algorithms with Novel Schemes of Variable Projection Order

Normalized Multichannel Frequency-Domain LMS Filter With Nearest Kronecker Product Decomposition for Blind Identification of Low-Rank Acoustic Systems

Robust Widely linear complex-valued combined step-size algorithm for beamforming

A Steered Response Power Approach with Bilinear Prediction-Based Trade-Off Prewhitening for Speaker Localization

Adaptive Subband Forward Blind Source Separation Algorithms Based on Kalman Mechanism

An adaptive multichannel speech dereverberation approach based on robust estimation

Robust Proportionate Subband Adaptive Filter Algorithms With Optimal Variable Step-Size

A tensor decomposition based multichannel linear prediction approach to speech dereverberation

A Recursive Least M-Estimate Adaptive Algorithm with Low Complexity for Active Control of Impulsive Noises

Robust Time-Delay Estimation for Speaker Localization Using Mutual Information Among Multiple Microphone Signals

A Frequency-Domain Recursive Least-Squares Adaptive Filtering Algorithm Based On A Kronecker Product Decomposition

A time delay estimation approach with low computational complexity for speaker localization

General Robust Proportionate NSAF Algorithm With a Step-Size Converter

General Robust Subband Adaptive Filtering for Echo Cancellation

A Step size Converter for Normalized Subband Adaptive Filtering Algorithm

Normalized Least Mean M-Estimate Algorithm with Switching Step-Sizes Against Impulsive Noises

Robust NLMS algorithms with combined step-size against impulsive noises

Sparsity-Aware Robust Normalized Subband Adaptive Filtering Algorithms With Alternating Optimization of Parameters

General Robust Subband Adaptive Filtering: Algorithms and Applications

基于多分辨率特征和时频注意力的环境声音分类

Robust Recursive Least M-Estimate Adaptive Filter for the Identification of Low-Rank Acoustic Systems

Robust Frequency-Domain Recursive Least M-Estimate Adaptive Filter For Acoustic System Identification

Diffusion Normalized Least Mean M-estimate Algorithms: Design and Performance Analysis

基于改进稀疏线性预测的时延估计算法

A class of multichannel sparse linear prediction algorithms for time delay estimation of speech sources

M-Estimate Based Normalized Subband Adaptive Filter Algorithm: Performance Analysis and Improvements

Noise Robust Frequency-Domain Adaptive Blind Multichannel Identification With $\ell _p$-Norm Constraint

An ℓ1/ℓ1-norm based linear prediction algorithm to time delay estimation via alternating direction method of multipliers

Multichannel time delay estimation for acoustic source localization via robust adaptive blind system identification

On directivity of a circular array with directional microphones

On time delay estimation based on multichannel spatiotemporal sparse linear prediction

基于传声器阵列的建模和定位算法研究

Time Difference of Arrival Estimation Exploiting Multichannel Spatio-Temporal Prediction

An Active Impulsive Noise Control Algorithm With Logarithmic Transformation

研究方向