何宏森教授介绍
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个人简历：（包括教育情况、工作经历）
何宏森，毕业于南京大学电子科学与工程学院暨声学研究所，博士、教授、博士研究生导师，现任职于西南科技大学beat365正版唯一网址；科研方面，面向临境语音通信、人机语音交互、声学监控、语音智能家居、助听等应用，致力于信号处理和语音通信领域的研究工作，主要包括统计与自适应信号处理、声信号处理、麦克风阵列信号处理、声源定位与跟踪、声通道辨识与均衡、语音分离与去混响、声回声抵消、语音降噪、自适应噪声控制等主题的研究；在声信号处理领域国际著名学术期刊和国际会议发表30余篇学术论文，先后承担各类科研项目20余项，其中主持国家自然科学基金面上项目2项、四川省杰出青年科技基金1项，2014年入选四川省杰出青年学术技术带头人(培育)。
论文著作
1. 部分期刊论文
[1] Yi Yu, Hongsen He, Tao Yang, Xueyuan Wang, and Rodrigo C. de Lamare, “Diffusion normalized least mean M-estimate algorithms: Design and performance analysis,” IEEE Transactions on Signal Processing, vol. 68, pp. 2199–2214, 2020.
[2] Hongsen He, Jingdong Chen, Jacob Benesty, Wenxing Zhang, and Tao Yang, “A class of multichannel sparse linear prediction algorithms for time delay estimation of speech sources,” Signal Processing (Elsevier), vol. 169, no. 4, pp. 1–14, Apr. 2020.
[3] Yi Yu, Hongsen He, Badong Chen, Jianghui Li, Youwen Zhang, and Lu Lu, “M-estimate based normalized subband adaptive filter algorithm: Performance analysis and improvements,” IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 28, no. 1, pp. 225–239, Jan. 2020.
[4] Hongsen He, Jingdong Chen, Jacob Benesty, and Tao Yang, “Noise robust frequency-domain adaptive blind multichannel identification with lp-norm constraint,” IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 26, no. 9, pp. 1608–1619, Sept. 2018.
[5] Hongsen He, Xueyuan Wang, Yingyue Zhou, and Tao Yang, “A steered response power approach with trade-off prewhitening for acoustic source localization,” Journal of the Acoustical Society of America, vol. 143, no. 2, pp. 1003–1007, Feb. 2018.
[6] Hongsen He, Tao Yang, and Jingdong Chen, “On time delay estimation from a sparse linear prediction perspective,” Journal of the Acoustical Society of America, vol. 137, no. 2, pp. 1044–1047, Feb. 2015.
[7] Hongsen He, Jing Lu, Jingdong Chen, Xiaojun Qiu, and Jacob Benesty, “Robust blind identification of room acoustic channels in symmetric alpha-stable distributed noise environments,” Journal of the Acoustical Society of America, vol. 136, no. 8, pp. 693–704, Aug. 2014.
[8] Hongsen He, Lifu Wu, Jing Lu, Xiaojun Qiu, and Jingdong Chen, “Time difference of arrival estimation exploiting multichannel spatio-temporal prediction,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 21, no. 3, pp. 463–475, Mar. 2013.
[9] Hongsen He, Jing Lu, Lifu Wu, and Xiaojun Qiu, “Time delay estimation via non-mutual information among multiple microphones,” Elsevier Applied Acoustics, vol. 74, no. 8, pp. 1033–1036, Aug. 2013.
[10] Lifu Wu, Hongsen He, and Xiaojun Qiu, “An active impulsive noise control algorithm with logarithmic transformation,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 19, no. 4, pp. 1041–1044, May 2011.
2. 部分国际会议论文
[1] Hongsen He, Jingdong Chen, Jacob Benesty, and Yi Yu, “Robust recursive least M-estimate adaptive filter for the identification of low-rank acoustic systems,” in Proc. 46th IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2021, pp. 940–944.
[2] Hongsen He, Jingdong Chen, Jacob Benesty, and Yi Yu, “Robust frequency-domain recursive least M-estimate adaptive filter for acoustic system identification,” in Proc. 45th IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2020, pp. 471–475.
[3] Hongsen He, Jingdong Chen, Jacob Benesty, Yingyue Zhou, and Tao Yang, “Robust multichannel TDOA estimation for speaker localization using the impulsive characteristics of speech spectrum,” in Proc. 42nd IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2017, pp. 6130–6134.
[4] Hongsen He, Jingdong Chen, Jacob Benesty, and Tao Yang, “Multichannel time delay estimation for acoustic source localization via robust adaptive blind system identification,” in Proc. 15th International Workshop on Acoustic Signal Enhancement (IWAENC, Biennial), 2016 (5 pages) .
[5] Hongsen He, Xiaojun Qiu, and Tao Yang, “On directivity of a circular array with directional microphones,” in Proc. 15th International Workshop on Acoustic Signal Enhancement (IWAENC, Biennial), 2016 (5 pages).
[6] Hongsen He, Jingdong Chen, Jacob Benesty, and Tao Yang, “On time delay estimation based on multichannel spatiotemporal sparse linear prediction,” in Proc. 41st IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2016, pp. 390–394.
3. 部分授权发明专利
[1] 何宏森，陈景东，杨涛，基于鲁棒自适应盲系统辨识的多通道时延估计方法，国家发明专利，ZL201610814013.1，授权日期：2018.05.29.
[2] 何宏森，陈景东，杨涛，多通道群稀疏线性预测时延估计方法，国家发明专利，ZL201610142030.5，授权日期：2017.12.05.
[3] 何宏森，黄志宏，邱小军，袁浩，基于传声器阵列的语音信号处理方法及装置，国家发明专利，ZL201010197159.9，授权日期：2014.11.05.
[1] Yi Yu, Hongsen He, Tao Yang, Xueyuan Wang, and Rodrigo C. de Lamare, “Diffusion normalized least mean M-estimate algorithms: Design and performance analysis,” IEEE Transactions on Signal Processing, vol. 68, pp. 2199–2214, 2020.
[2] Hongsen He, Jingdong Chen, Jacob Benesty, Wenxing Zhang, and Tao Yang, “A class of multichannel sparse linear prediction algorithms for time delay estimation of speech sources,” Signal Processing (Elsevier), vol. 169, no. 4, pp. 1–14, Apr. 2020.
[3] Yi Yu, Hongsen He, Badong Chen, Jianghui Li, Youwen Zhang, and Lu Lu, “M-estimate based normalized subband adaptive filter algorithm: Performance analysis and improvements,” IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 28, no. 1, pp. 225–239, Jan. 2020.
[4] Hongsen He, Jingdong Chen, Jacob Benesty, and Tao Yang, “Noise robust frequency-domain adaptive blind multichannel identification with lp-norm constraint,” IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 26, no. 9, pp. 1608–1619, Sept. 2018.
[5] Hongsen He, Xueyuan Wang, Yingyue Zhou, and Tao Yang, “A steered response power approach with trade-off prewhitening for acoustic source localization,” Journal of the Acoustical Society of America, vol. 143, no. 2, pp. 1003–1007, Feb. 2018.
[6] Hongsen He, Tao Yang, and Jingdong Chen, “On time delay estimation from a sparse linear prediction perspective,” Journal of the Acoustical Society of America, vol. 137, no. 2, pp. 1044–1047, Feb. 2015.
[7] Hongsen He, Jing Lu, Jingdong Chen, Xiaojun Qiu, and Jacob Benesty, “Robust blind identification of room acoustic channels in symmetric alpha-stable distributed noise environments,” Journal of the Acoustical Society of America, vol. 136, no. 8, pp. 693–704, Aug. 2014.
[8] Hongsen He, Lifu Wu, Jing Lu, Xiaojun Qiu, and Jingdong Chen, “Time difference of arrival estimation exploiting multichannel spatio-temporal prediction,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 21, no. 3, pp. 463–475, Mar. 2013.
[9] Hongsen He, Jing Lu, Lifu Wu, and Xiaojun Qiu, “Time delay estimation via non-mutual information among multiple microphones,” Elsevier Applied Acoustics, vol. 74, no. 8, pp. 1033–1036, Aug. 2013.
[10] Lifu Wu, Hongsen He, and Xiaojun Qiu, “An active impulsive noise control algorithm with logarithmic transformation,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 19, no. 4, pp. 1041–1044, May 2011.
2. 部分国际会议论文
[1] Hongsen He, Jingdong Chen, Jacob Benesty, and Yi Yu, “Robust recursive least M-estimate adaptive filter for the identification of low-rank acoustic systems,” in Proc. 46th IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2021, pp. 940–944.
[2] Hongsen He, Jingdong Chen, Jacob Benesty, and Yi Yu, “Robust frequency-domain recursive least M-estimate adaptive filter for acoustic system identification,” in Proc. 45th IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2020, pp. 471–475.
[3] Hongsen He, Jingdong Chen, Jacob Benesty, Yingyue Zhou, and Tao Yang, “Robust multichannel TDOA estimation for speaker localization using the impulsive characteristics of speech spectrum,” in Proc. 42nd IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2017, pp. 6130–6134.
[4] Hongsen He, Jingdong Chen, Jacob Benesty, and Tao Yang, “Multichannel time delay estimation for acoustic source localization via robust adaptive blind system identification,” in Proc. 15th International Workshop on Acoustic Signal Enhancement (IWAENC, Biennial), 2016 (5 pages) .
[5] Hongsen He, Xiaojun Qiu, and Tao Yang, “On directivity of a circular array with directional microphones,” in Proc. 15th International Workshop on Acoustic Signal Enhancement (IWAENC, Biennial), 2016 (5 pages).
[6] Hongsen He, Jingdong Chen, Jacob Benesty, and Tao Yang, “On time delay estimation based on multichannel spatiotemporal sparse linear prediction,” in Proc. 41st IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2016, pp. 390–394.
3. 部分授权发明专利
[1] 何宏森，陈景东，杨涛，基于鲁棒自适应盲系统辨识的多通道时延估计方法，国家发明专利，ZL201610814013.1，授权日期：2018.05.29.
[2] 何宏森，陈景东，杨涛，多通道群稀疏线性预测时延估计方法，国家发明专利，ZL201610142030.5，授权日期：2017.12.05.
[3] 何宏森，黄志宏，邱小军，袁浩，基于传声器阵列的语音信号处理方法及装置，国家发明专利，ZL201010197159.9，授权日期：2014.11.05.
科研项目（项目名称、项目编号、项目来源、起止时间、经费、排名情况）：
代表性科研项目
[1] 主持国家自然科学基金面上项目：基于麦克风阵列的鲁棒语音去混响研究，No：62071399，2021.01.01-2024.12.31。
[2] 主持国家自然科学基金面上项目：复杂声环境下的鲁棒自适应滤波理论及算法研究，No：61571376，2016.01.01-2019.12.31。
[3] 主持四川省杰出青年科技基金项目：鲁棒频域自适应滤波及其在音频声信号处理中的应用研究，No：2014JQ0042，2014.01.01-2016.12.31。
[1] 主持国家自然科学基金面上项目：基于麦克风阵列的鲁棒语音去混响研究，No：62071399，2021.01.01-2024.12.31。
[2] 主持国家自然科学基金面上项目：复杂声环境下的鲁棒自适应滤波理论及算法研究，No：61571376，2016.01.01-2019.12.31。
[3] 主持四川省杰出青年科技基金项目：鲁棒频域自适应滤波及其在音频声信号处理中的应用研究，No：2014JQ0042，2014.01.01-2016.12.31。
教学工作（讲授过的课程、指导研究生的情况）：
为自动化、电气、电子、通信等本科专业讲授“自动控制理论”、为研究生/留学生讲授“Modern Signal Processing”等课程。
获奖及其他（获奖、学术交流、社会活动、社会兼职等）
美国电气与电子工程师协会(IEEE)和美国声学学会(ASA)会员，担任多个IEEE Transactions期刊、Journal of the Acoustical Society of America(美国声学学报)、Elsevier Signal Processing、IEEE Signal Processing Letters、IEEE Access等学术期刊及国际会议的审稿专家和国家自然科学基金项目的通讯评审专家。