This book introduces readers to the novel concept of variable span speech enhancement filters, and demonstrates how it can be used for effective noise reduction in various ways. Further, the book provides the accompanying Matlab code, allowing readers to easily implement the main ideas discussed. Variable span filters combine the ideas of optimal linear filters with those of subspace methods, as they involve the joint diagonalization of the correlation matrices of the desired signal and the noise. The book shows how some well-known filter designs, e.g. the minimum distortion, maximum signal-to-noise ratio, Wiener, and tradeoff filters (including their new generalizations) can be obtained using the variable span filter framework. It then illustrates how the variable span filters can be applied in various contexts, namely in single-channel STFT-based enhancement, in multichannel enhancement in both the time and STFT domains, and, lastly, in time-domain binaural enhancement. In these contexts, the properties of these filters are analyzed in terms of their noise reduction capabilities and desired signal distortion, and the analyses are validated and further explored in simulations. **

Presents the novel concept of variable span speech enhancement filters Shows various ways in which the concept can be used to achieve noise reduction Includes Matlab code so that all of the main ideas can be easily implemented Includes supplementary material: sn.pub/extras

Autorentext

Jacob Benesty received a master's degree in microwaves from Pierre & Marie Curie University, France, in 1987, and a Ph.D. degree in control and signal processing from Paris-Saclay University, France, in April 1991. During his Ph.D. (from Nov. 1989 to Apr. 1991), he worked on adaptive filters and fast algorithms at the Centre National d'Etudes des Telecommunications (CNET), Paris, France. From January 1994 to July 1995, he worked at Telecom Paris University on multichannel adaptive filters and acoustic echo cancellation. From October 1995 to May 2003, he was first a consultant and then a Member of the Technical Staff at Bell Laboratories, Murray Hill, NJ, USA. In May 2003, he joined the University of Quebec, INRS-EMT, in Montreal, Quebec, Canada, as a professor. He is also an adjunct professor with Aalborg University, Denmark, and a guest professor with Northwestern Polytechnical University, Xi'an, China. His research interests are in signal processing, acoustic signal processing, and multimedia communications. He is the inventor of many important technologies. In particular, he was the lead researcher at Bell Labs who conceived and designed the world-first real-time hands-free full-duplex stereophonic teleconferencing system. Also, he conceived and designed the world-first PC-based multi-party hands-free full-duplex stereo conferencing system over IP networks. He is the editor of the book series Springer Topics in Signal Processing. He was the general chair and technical chair of many international conferences and a member of several IEEE technical committees. Four of his journal papers were awarded by the IEEE Signal Processing Society, in 2010 he received the Gheorghe Cartianu Award from the Romanian Academy, and in 2023 he received an Honorary Doctorate (Doctor Technices Honoris Causa) from Aalborg University, Denmark, for his distinguished efforts in audio and acoustic signal processing. He has co-authored and co-edited/co-authored numerous books in the areaof acoustic signal processing. Gongping Huang received his bachelor's degree in Electronics and Information Engineering and his Ph.D. degree in Information and Communication Engineering from Northwestern Polytechnical University (NPU) in Xian, China, in 2012 and 2019, respectively. Between 2015 and 2017, he worked as a visiting researcher at University of Quebec, INRS-EMT, Montreal, Quebec, Canada. Following this, he worked as a postdoctoral research fellow at the Technion-Israel Institute of Technology in Haifa, Israel, from 2019 to 2021. He was a Humboldt Research Fellow at the University of Erlangen-Nuremberg in Germany, supported by the Alexander von Humboldt Foundation. He is now a professor at Wuhan University. His research interests include microphone arrays, acoustic signal processing, and speech enhancement. Dr. Huang received the Humboldt Research Fellowship (2021), the Andrew and Erna Finci Viterbi Post-Doctoral Fellowship award (2019), the Best Ph.D. Thesis Award from the Chinese Institute of Electronics (2021), and the Best Ph.D. Thesis Award of Shanxi Province (2021). He is currently serving as an Associate Editor for the Circuits Systems and Signal Processing Journal and a Consulting Associate Editors for the IEEE Open Journal of Signal Processing. He also serves as an active reviewer for more than 30 scientific journals and international conferences. Jingdong Chen received the Ph.D. degree in pattern recognition and intelligence control from the Chinese Academy of Sciences in 1998. From 1998 to 1999, he was with ATR Interpreting Telecommunications Research Laboratories, Kyoto, Japan, where he conducted research on speech synthesis, speech analysis, as well as objective measurements for evaluating speech synthesis. He then joined the Griffith University, Brisbane, Australia, where he engaged in research on robust speech recognition and signal processing. From 2000 to 2001, he worked at ATR Spoken Language Translation Research Laboratories

Inhalt

Introduction.- General Concept with Filtering Vectors.- General Concept with Filtering Matrices.- Single-Channel Signal Enhancement in the STFT Domain.- Multichannel Signal Enhancement in the Time Domain.- Multichannel Signal Enhancement in the STFT Domain.- Binaural Signal Enhancement in the Time Domain.