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1 Constant Directivity Beamforming
1.1 Introduction
1.2 Problem Formulation
1.3 Theoretical Solution
1.3.1 Continuous sensor
1.3.2 Beam-shaping function
1.4 Practical Implementation
1.4.1 Dimension-reducing parameterization
1.4.2 Reference beam-shaping filter
1.4.3 Sensor placelllent
1.4.4 SUllllllary of illlplelllentation
1.5 Examples
1.6 Conclusions
2 Superdirective Microphone Arrays
2.1 Introduction
2.2 Evaluation of Beamformers
2.2.1 Array-Gain
2.2.2 Beampattern
2.2.3 Directivity
2.2.4 Front-to-Back Ratio
2.2.5 White Noise Gain
2.3 Design of Superdirective Beamformers
2.3.1 Delay-and-Sum Beamformer
2.3.2 Design for spherical isotropic noise
2.3.3 Design for Cylindrical Isotropic Noise
2.3.4 Design for an Optimal Front-to-Back Ratio
2.3.5 Design for Measured Noise Fields
2.4 Extensions and Details
2.4.1 Alternative Form
2.4.2 Comparison with Gradient Microphones
2.5 Conclusion
3 Post-Filtering Techniques
3.1 Introduction
3.2 Multi-channel Wiener Filtering in Subbands
3.2.1 Derivation of the Optimum Solution
3.2.2 Factorization of the Wiener Solution
3.2.3 Interpretation
3.3 Algorithms for Post-Filter Estimation
3.3.1 Analysis of Post-Filter Algorithms
3.3.2 Properties of Post-Filter Algorithms
3.3.3 A New Post-Filter Algorithm
3.4 Performance Evaluation
3.4.1 Simulation System
3.4.2 Objective Measures
3.4.3 Simulation Results
3.5 Conclusion
4 Spatial Coherence Functions for Differential Microphones in Isotropic NoiseFields
4.1 Introduction
4.2 Adaptive Noise Cancellation
4.3 Spherically Isotropic Coherence
4.4 Cylindrically Isotropic Fields
4.5 Conclusions
5 Robust Adaptive Beamforming
5.1 Introduction
5.2 Adaptive Beamformers
5.3 Robustness Problem in the GJBF
5.4 Robust Adaptive Microphone Arrays - Solutionsto Steering-Vector Errors
5.4.1 LAF-LAF Structure
5.4.2 CCAF-LAF Structure
5.4.3 CCAF-NCAF Structure
5.4.4 CCAF-NCAF Structure with an AMC
5.5 Software Evaluation of a Robust AdaptiveMicrophone Array
5.5.1 Simulated Anechoic Environment
5.5.2 Reverberant Environment
5.6 Hardware Evaluation of a Robust AdaptiveMicrophone Array
5.6.1 Implementation
5.6.2 Evaluation in a Real Environment
5.7 Conclusion
6 GSVD-Based Optimal Filtering forMulti-Microphone Speech Enhancement
6.1 Introduction
6.2 GSVD-Based Optimal Filtering Technique
6.2.1 Optimal Filter Theory
6.2.2 General Class of Estimators
6.2.3 Symmetry Properties for Time-Series Filtering
6.3 Performance of GSVD-Based Optimal Filtering
6.3.1 Simulation Environment
6.3.2 Spatial Directivity Pattern
6.3.3 Noise Reduction Performance
6.3.4 Robustness Issues
6.4 Complexity Reduction
6.4.1 Linear Algebra Techniques for Computing GSVD
6.4.2 Recursive and Approximate GSVD-Updating Algorithms
6.4.3 Downsampling Techniques
6.4.4 Simulations
6.4.5 Computational Complexity
6.5 Combination with ANC Postprocessing Stage
6.5.1 Creation of Speech and Noise References
6.5.2 Noise Reduction Performance of ANC PostprocessingStage
6.5.3 Comparison with Standard Beamforming Techniques
6.6 Conclusion
7 Explicit Speech Modeling for MicrophoneArray Applications
7.1 Introduction
7.2 Model-Based Strategies
7.2.1 Example 1: A Frequency-Domain Model-Based Algorithm
7.2.2 Example 2: A Time-Domain Model-Based Algorithm
Michael Brandstein . Darren Ward (Eds.) Microphone Arrays Signal Processing Techniques and Applications With 149 Figures Springer
Series Editors Prof. Dr.-Ing. ARILD LACROIX Johann-Wolfgang-Goethe-Universitiit Institut ftir angewandte Physik Robert-Mayer-Str. 2-4 D-60325 Frankfurt Prof. Dr.-Ing. ANASTAS lOS VENETSANOPOULOS University of Toronto Dept. of Electrical and Computer Engineering 10 King's College Road M5S 3G4 Toronto, Ontario Canada Editors Prof. MICHAEL BRANDSTEIN Harvard University, Div. of Eng. and Applied Scciences 33 Oxford Street MA 02138 Cambridge USA e-mail: msb@hrl.harvard.edu Dr. DARREN WARD Imperial College, Dept. of Electrical Engineering Exhibition Road SW7 2AZ London GB e-mail: d.ward@ic.ac.uk ISBN 978-3-642-07547-6 DOl 10.1007/978-3-662-04619-7 ISBN 978-3-662-04619-7 (eBook) Cip data applied for This work is subject to copyright. All rights are reserved, whether the whole or part of the material is concerned, specifically the rights of translation, reprinting, reuse of illustrations, recitation, broadcasting, reproduction on microfilm or in other ways, and storage in data banks. Duplication of this publication or parts thereof is permitted only under the provisions of the German Copyright Law of September 9, 1965, in its current version, and permission for use must always be obtained from Springer-Verlag Berlin Heidelberg GmbH. Violations are liable for prosecution act under German Copyright Law. http://www.springer.de © Springer-Verlag Berlin Heidelberg 2001 Originally published by Springer-Verlag Berlin Heidelberg New York in 2001 Softcover reprint of tbe hardcover 1st edition 2001 The use of general descriptive names, registered names, trademarks, etc. in this publication does not imply, even in the absence of a specific statement, that such names are exempt from the relevant protective laws and regulations and therefore free for general use. Typesetting: Camera-ready copy by authors Cover-Design: de'blik, Berlin SPIN: 10836055 62/3020 543 2 1 0 Printed on acid-free paper
Preface The study and implementation of microphone arrays originated over 20 years ago. Thanks to the research and experimental developments pursued to the present day, the field has matured to the point that array-based technology now has immediate applicability to a number of current systems and a vast potential for the improvement of existing products and the creation of future devices. In putting this book together, our goal was to provide, for the first time, a single complete reference on microphone arrays. We invited the top re searchers in the field to contribute articles addressing their specific topic(s) of study. The reception we received from our colleagues was quite enthusi astic and very encouraging. There was the general consensus that a work of this kind was well overdue. The results provided in this collection cover the current state of the art in microphone array research, development, and technological application. This text is organized into four sections which roughly follow the major areas of microphone array research today. Parts I and II are primarily the oretical in nature and emphasize the use of microphone arrays for speech enhancement and source localization, respectively. Part III presents a num ber of specific applications of array-based technology. Part IV addresses some open questions and explores the future of the field. Part I concerns the problem of enhancing the speech signal acquired by an array of microphones. For a variety of applications, including human computer interaction and hands-free telephony, the goal is to allow users to roam unfettered in diverse environments while still providing a high quality speech signal and robustness against background noise, interfering sources, and reverberation effects. The use of microphone arrays gives one the oppor tunity to exploit the fact that the source of the desired speech signal and the noise sources are physically separated in space. Conventional array process ing techniques, typically developed for applications such as radar and sonar, were initially applied to the hands-free speech acquisition problem. However, the environment in which microphone arrays is used is significantly different from that of conventional array applications. Firstly, the desired speech signal has an extremely wide bandwidth relative to its center frequency, meaning that conventional narrowband techniques are not suitable. Secondly, there
VI Preface is significant multi path interference caused by room reverberation. Finally, the speech source and noise signals may located close to the array, meaning that the conventional far-field assumption is typically not valid. These dif ferences (amongst others) have meant that new array techniques have had to be formulated for microphone array applications. Chapter 1 describes the design of an array whose spatial response does not change appreciably over a wide bandwidth. Such a design ensures that the spatial filtering performed by the array is uniform across the entire bandwidth of the speech signal. The main problem with many array designs is that a very large physical array is required to obtain reasonable spatial resolution, especially at low frequencies. This problem is addressed in Chapter 2, which reviews so-called superdirec tive arrays. These arrays are designed to achieve spatial directivity that is significantly higher than a standard delay-and-sum beamformer. Chapter 3 describes the use of a single-channel noise suppression filter on the output of a microphone array. The design of such a post-filter typically requires in formation about the correlation of the noise between different microphones. The spatial correlation functions for various directional microphones are in vestigated in Chapter 4, which also describes the use of these functions in adaptive noise cancellation applications. Chapter 5 reviews adaptive tech niques for microphone arrays, focusing on algorithms that are robust and perform well in real environments. Chapter 6 presents optimal spatial filter ing algorithms based on the generalized singular-value decomposition. These techniques require a large number of computations, so the chapter presents techniques to reduce the computational complexity and thereby permit real time implementation. Chapter 7 advocates a new approach that combines explicit modeling of the speech signal (a technique which is well-known in single-channel speech enhancement applications) with the spatial filtering af forded by multi-channel array processing. Part II is devoted to the source localization problem. The ability to locate and track one or more speech sources is an essential requirement of micro phone array systems. For speech enhancement applications, an accurate fix on the primary talker, as well as knowledge of any interfering talkers or coher ent noise sources, is necessary to effectively steer the array, enhancing a given source while simultaneously attenuating those deemed undesirable. Location data may be used as a guide for discriminating individual speakers in a multi source scenario. With this information available, it would then be possible to automatically focus upon and follow a given source on an extended basis. Of particular interest lately, is the application of the speaker location estimates for aiming a camera or series of cameras in a video-conferencing system. In this regard, the automated localization information eliminates the need for a human or number of human camera operators. Several existing commercial products apply microphone-array technology in small-room environments to steer a robotic camera and frame active talkers. Chapter 8 summarizes the various approaches which have been explored to accurately locate an individ-
Preface VII ual in a practical acoustic environment. The emphasis is on precision in the face of adverse conditions, with an appropriate method presented in detail. Chapter 9 extends the problem to the case of multiple active sources. While again considering realistic environments, the issue is complicated by the pres ence of several talkers. Chapter 10 further generalizes the source localization scenario to include knowledge derived from non-acoustic sensor modalities. In this case both audio and video signals are effectively combined to track the motion of a talker. Part III of this text details some specific applications of microphone array technology available today. Microphone arrays have been deployed for a vari ety of practical applications thus far and their utility and presence in our daily lives is increasing rapidly. At one extreme are large aperture arrays with tens to hundreds of elements designed for large rooms, distant talkers, and adverse acoustic conditions. Examples include the two-dimensional, harmonic array installed in the main auditorium of Bell Laboratories, Murray Hill and the 512-element Huge Microphone Array (HMA) developed at Brown University. While these systems provide tremendous functionality in the environments for which they are intended, small arrays consisting of just a handful (usu ally 2 to 8) of microphones and encompassing only a few centimeters of space have become far more common and affordable. These systems are intended for sound capture in close-talking, low to moderate noise conditions (such as an individual dictating at a workstation or using a hands-free telephone in an automobile) and have exhibited a degree of effectiveness, especially when compared to their single microphone counterparts. The technology has developed to the point that microphone arrays are now available in off-the shelf consumer electronic devices available for under $150. Because of their growing popularity and feasibility we have chosen to focus primarily on the issues associated with small-aperture devices. Chapter 11 addresses the in corporation of multiple microphones into hearing aid devices. The ability of beamforming methods to reduce background noise and interference has been shown to dramatically improve the speech understanding of the hearing im paired and to increase their overall satisfaction with the device. Chapter 12 focuses on the case of a simple two-element array combined with postfiltering to achieve noise and echo reduction. The performance of this configuration is analyzed under realistic acoustic conditions and its utility is demonstrated for desktop conferencing and intercom applications. Chapter 13 is concerned with the problem of acoustic feedback inherent in full-duplex communica tions involving loudspeakers and microphones. Existing single-channel echo cancellation methods are integrated within a beamforming context to achieve enhanced echo suppression. These results are applied to single- and multi channel conferencing scenarios. Chapter 14 explores the use of microphone arrays for sound capture in automobiles. The issues of noise, interference, and echo cancellation specifically within the car environment are addressed and a particularly effective approach is detailed. Chapter 15 discusses the applica-
VIII Preface tion of microphone arrays to improve the performance of speech recognition systems in adverse conditions. Strategies for effectively coupling the acous tic signal enhancements afforded through beamforming with existing speech recognition techniques are presented. A specific adaptation of a recognizer to function with an array is presented. Finally, Chapter 16 presents an overview of the problem of separating blind mixtures of acoustic signals recorded at a microphone array. This represents a very new application for microphone ar rays, and is a technique that is fundamentally different to the spatial filtering approaches detailed in earlier chapters. In the final section of the book, Part IV presents expert summaries of current open problems in the field, as well as personal views of what the future of microphone array processing might hold. These summaries, presented in Chapters 17 and 18, describe both academically-oriented research problems, as well as industry-focused areas where microphone array research may be headed. The individual chapters that we selected for .the book were designed to be tutorial in nature with a specific emphasis on recent important results. We hope the result is a text that will be of utility to a large audience, from the student Or practicing engineer just approaching the field to the advanced researcher with multi-channel signal processing experience. Cambridge MA, USA London, UK January 2001 Michael Brandstein Darren Ward
Contents Part I. Speech Enhancement 1 Constant Directivity Beamforming Darren B. Ward, Rodney A. Kennedy, Robert C. Williamson ........ 3 1.1 Introduction................................................ 3 1.2 Problem Formulation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 7 1.3 Theoretical Solution. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1.3.1 Continuous sensor. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 1.3.2 Beam-shaping function. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8 1.4 Practical Implementation .................................... 9 1.4.1 Dimension-reducing parameterization. . . . . . . . . . . . . . . .. . . . 9 1.4.2 Reference beam-shaping filter. . . . . . . . . . . . . . . . . . . . . . . . . .. 11 1.4.3 Sensor placement. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 12 1.4.4 Summary of implementation . . . . . . . . . . . . . . . . . . . . . . . . . . .. 12 1.5 Examples;................................................. 13 1.6 Conclusions .. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 16 References ..................................................... 16 2 Superdirective Microphone Arrays Joerg Bitzer, K. Uwe Simmer .................................... 19 2.1 Introduction................................................ 19 2.2 Evaluation of Beamformers... .. . . . . .. .. . . . . . . . . . . . .. . . . . .. ... 20 2.2.1 Array-Gain........................................... 21 2.2.2 Beampattern.......................................... 22 2.2.3 Directivity............................................ 23 2.2.4 Front-to-Back Ratio ................................... 24 2.2.5 White Noise Gain ..................................... 24 2.3 Design of Superdirective Beamformers . . . . . . . . . . . . . . . . . . . . . . . .. 24 2.3.1 Delay-and-Sum Beamformer ............................ 26 2.3.2 Design for spherical isotropic noise. . . . . . . . . . . . . . . . . . . . . .. 26 2.3.3 Design for Cylindrical Isotropic Noise . . . . . . . . . . . . . . . . . . .. 30 2.3.4 Design for an Optimal Front-to-Back Ratio.. .. . . . . . . . . ... 30 2.3.5 Design for Measured Noise Fields. . .. . . . . . .. . . . . .. . . . . ... 32 2.4 Extensions and Details ...................................... 33 2.4.1 Alternative Form. . .. . . .. . . . . . . . . . . . .. . . . . . . . . . . . . . . . .. 33
X Contents 2.4.2 Comparison with Gradient Microphones. . . . . . . . . . . . . . . . .. 35 2.5 Conclusion................................................. 36 References ..................................................... 37 3 Post-Filtering Techniques K. Uwe Simmer, Joerg Bitzer, Claude Marro. .. . . . . . . . .. . . . . . . . . . .. 39 3.1 Introduction................................................ 39 3.2 Multi-channel Wiener Filtering in Subbands . . . . . . . . . . . . . . . . . . .. 41 3.2.1 Derivation of the Optimum Solution. . . . . . .. . . . . .. . . . . . .. 41 3.2.2 Factorization of the Wiener Solution . . . . . . . . . . . . . . . . . . . .. 42 3.2.3 Interpretation......................................... 45 3.3 Algorithms for Post-Filter Estimation ......................... 46 3.3.1 Analysis of Post-Filter Algorithms. . . . . . . . . . . . . . . . . . . . . .. 47 3.3.2 Properties of Post-Filter Algorithms ..................... 49 3.3.3 A New Post-Filter Algorithm ........................... 50 3.4 Performance Evaluation ..................................... 51 3.4.1 Simulation System. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 52 3.4.2 Objective Measures. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 52 3.4.3 Simulation Results. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 54 3.5 Conclusion................................................. 57 4 Spatial Coherence Functions for Differential Microphones in Isotropic Noise Fields Gary W. Elko .... . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 61 4.1 Introduction................................................ 61 4.2 Adaptive Noise Cancellation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 61 4.3 Spherically Isotropic Coherence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 65 4.4 Cylindrically Isotropic Fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 73 4.5 Conclusions................................................ 77 References ..................................................... 84 5 Robust Adaptive Beamforming Osamu Hoshuyama, Akihiko Sugiyama. . . . . . . . . . . . . . . . . . . . . . . . . . . .. 87 5.1 Introduction................................................ 87 5.2 Adaptive Beamformers . .... . . . . . . . . . . .. . . . . . . . . . .. . . .. . . . . . .. 88 5.3 Robustness Problem in the GJBF . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 90 5.4 Robust Adaptive Microphone Arrays - Solutions to Steering- Vector Errors .............................................. 92 5.4.1 LAF-LAF Structure ................................... 92 5.4.2 CCAF-LAF Structure. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. 94 5.4.3 CCAF-NCAF Structure.......... ... ...... ............. 95 5.4.4 CCAF-NCAF Structure with an AMC ................... 97 5.5 Software Evaluation of a Robust Adap~ive Microphone Array. . . .. 99 5.5.1 Simulated Anechoic Environment.. .. . . . . . .. . . . . .. . . . . . .. 99 5.5.2 Reverberant Environment .............................. 101
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