Microphone Arrays

Benesty, Jacob, Huang, Gongping, Chen, Jingdong

  • 出版商: Springer
  • 出版日期: 2024-08-11
  • 售價: $5,690
  • 貴賓價: 9.5$5,406
  • 語言: 英文
  • 頁數: 228
  • 裝訂: Quality Paper - also called trade paper
  • ISBN: 3031369769
  • ISBN-13: 9783031369766
  • 海外代購書籍(需單獨結帳)

相關主題

商品描述

This book explains the motivation for using microphone arrays as opposed to using a single sensor for sound acquisition. The book then goes on to summarize the most useful ideas, concepts, results, and new algorithms therein. The material presented in this work includes analysis of the advantages of using microphone arrays, including dimensionality reduction to remove the redundancy while preserving the variability of the array signals using the principal component analysis (PCA). The authors also discuss benefits such as beamforming with low-rank approximations, fixed, adaptive, and robust distortionless beamforming, differential beamforming, and a new form of binaural beamforming that takes advantage of both beamforming and human binaural hearing properties to improve speech intelligibility. The book makes the microphone array signal processing theory and applications available in a complete and self-contained text. The authors attempt to explain the main ideas in a clear and rigorous way so that the reader can easily capture the potentials, opportunities, challenges, and limitations of microphone array signal processing. This book is written for those who work on the topics of microphone arrays, noise reduction, speech enhancement, speech communication, and human-machine speech interfaces.
  • Presents the benefits of microphone arrays over single sensors from a signal enhancement / spatial filtering perspective;
  • Includes algorithms for dimensionality reduction and fixed and adaptive beamforming with microphone arrays;
  • Posits a framework that takes advantage of microphone array and human binaural hearing to improve speech intelligibility.

商品描述(中文翻譯)

本書解釋了使用麥克風陣列而非單一感測器進行聲音獲取的動機。接著,本書總結了其中最有用的觀念、概念、結果和新算法。本書所呈現的材料包括對使用麥克風陣列的優勢進行分析,包括利用主成分分析(PCA)進行降維,以去除冗餘,同時保留陣列信號的變異性。作者還討論了如低秩近似的波束形成、固定的、自適應的和穩健的無失真波束形成、差分波束形成,以及一種新型的雙耳波束形成,該方法利用波束形成和人類雙耳聽覺特性來提高語音可懂度。本書將麥克風陣列信號處理的理論和應用以完整且自成一體的文本形式呈現。作者試圖以清晰且嚴謹的方式解釋主要觀念,使讀者能夠輕鬆掌握麥克風陣列信號處理的潛力、機會、挑戰和限制。本書適合從事麥克風陣列、噪音減少、語音增強、語音通信和人機語音介面的相關工作的人士。

- 從信號增強/空間過濾的角度介紹麥克風陣列相對於單一感測器的優勢;
- 包含麥克風陣列的降維算法以及固定和自適應波束形成算法;
- 提出一個框架,利用麥克風陣列和人類雙耳聽覺來提高語音可懂度。

作者簡介

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 on robust speech recognition and speech enhancement. From 2001 to 2009, he was a Member of Technical Staff at Bell Laboratories, Murray Hill, New Jersey, working on acoustic signal processing for telecommunications. He subsequently joined WeVoice Inc. in New Jersey, serving as the Chief Scientist. He is currently a professor at the Northwestern Polytechnical University in Xi'an, China. His research interests include array signal processing, adaptive signal processing, speech enhancement, adaptive noise/echo control, signal separation, speech communication, and artificial intelligence. Dr. Chen has coauthored 12 monograph books. He received the 2008 Best Paper Award from the IEEE Signal Processing Society (with Benesty, Huang, and Doclo), the Best Paper Award from the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics in 2011 (with Benesty), the Bell Labs Role Model Teamwork Award twice, respectively, in 2009 and 2007, the NASA Tech Brief Award twice, respectively, in 2010 and 2009, and the Young Author Best Paper Award from the 5th National Conference on Man-Machine Speech Communications in 1998. He is a co-author of a paper for which C. Pan received the IEEE R10 (Asia-Pacific Region) Distinguished Student Paper Award (First Prize) in 2016. He was also a recipient of the Japan Trust International Research Grant from the Japan Key Technology Center in 1998 and the "Distinguished Young Scientists Fund" from the National Natural Science Foundation of China (NSFC) in 2014.

Ningning Pan received a bachelor's degree in Electronics and Information Engineering in 2014 and a Master's degree in Signal and Information Processing in 2017 all from the Northwestern Polytechnical University (NPU). She is expected to receive a Ph.D. degree in Information and Communication also from NPU in 2023. Between 2018 and 2020, she was a visiting scholar at Columbia University. She will be soon joining the Southwestern University of Finance and Economics asan associate professor. Her research interests include speech enhancement, artificial intelligence, binaural hearing, and acoustic signal processing.


作者簡介(中文翻譯)

雅各布·本斯提(Jacob Benesty)於1987年在法國皮埃爾與瑪麗·居里大學獲得微波碩士學位,並於1991年4月在法國巴黎薩克雷大學獲得控制與信號處理的博士學位。在他的博士期間(1989年11月至1991年4月),他在法國巴黎的國家電信研究中心(CNET)從事自適應濾波器和快速算法的研究。1994年1月至1995年7月,他在巴黎電信大學研究多通道自適應濾波器和聲學回聲消除。1995年10月至2003年5月,他先是擔任貝爾實驗室(Bell Laboratories)顧問,後來成為技術人員,該實驗室位於美國新澤西州的穆雷山。2003年5月,他加入加拿大魁北克省蒙特利爾的魁北克大學INRS-EMT,擔任教授。他同時也是丹麥奧爾堡大學的兼任教授,以及中國西安的西北工業大學的客座教授。他的研究興趣包括信號處理、聲學信號處理和多媒體通信。他是許多重要技術的發明者,特別是他是貝爾實驗室的首席研究員,構思並設計了全球首個實時免提全雙工立體聲視頻會議系統。此外,他還構思並設計了全球首個基於PC的多方免提全雙工立體聲會議系統,該系統可通過IP網絡運行。他是《Springer Topics in Signal Processing》系列書籍的編輯。他曾擔任多個國際會議的總主席和技術主席,並且是幾個IEEE技術委員會的成員。他的四篇期刊論文曾獲得IEEE信號處理學會的獎項,2010年他獲得羅馬尼亞科學院的Gheorghe Cartianu獎,2023年他因在音頻和聲學信號處理方面的卓越貢獻,獲得丹麥奧爾堡大學的榮譽博士學位(Doctor Technices Honoris Causa)。他在聲學信號處理領域共同撰寫和編輯了多本書籍。

黃公平(Gongping Huang)於2012年和2019年分別在中國西安的西北工業大學獲得電子與信息工程學士學位和信息與通信工程博士學位。2015年至2017年間,他在加拿大魁北克省蒙特利爾的魁北克大學INRS-EMT擔任訪問研究員。隨後,他於2019年至2021年在以色列海法的以色列理工學院(Technion)擔任博士後研究員。他是德國埃爾朗根-紐倫堡大學的洪堡研究員,該項目由亞歷山大·洪堡基金會支持。他目前是武漢大學的教授。他的研究興趣包括麥克風陣列、聲學信號處理和語音增強。黃博士獲得了洪堡研究獎學金(2021年)、安德魯與厄爾娜·芬奇·維特比博士後獎(2019年)、中國電子學會最佳博士論文獎(2021年)以及山西省最佳博士論文獎(2021年)。他目前擔任《Circuits Systems and Signal Processing Journal》的副編輯,以及IEEE Open Journal of Signal Processing的顧問副編輯。他還擔任超過30本科學期刊和國際會議的活躍審稿人。

陳京東(Jingdong Chen)於1998年在中國科學院獲得模式識別與智能控制的博士學位。1998年至1999年,他在日本京都的ATR解釋電信研究實驗室工作,進行語音合成、語音分析以及評估語音合成的客觀測量研究。隨後,他加入澳大利亞布里斯班的格里菲斯大學,從事穩健語音識別和信號處理的研究。2000年至2001年,他在ATR口語語言翻譯研究實驗室工作。