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E-raamat: Theory and Applications of Spherical Microphone Array Processing

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This book presents the signal processing algorithms that have been developed to process the signals acquired by a spherical microphone array. Spherical microphone arrays can be used to capture the sound field in three dimensions and have received significant interest from researchers and audio engineers. Algorithms for spherical array processing are different to corresponding algorithms already known in the literature of linear and planar arrays because the spherical geometry can be exploited to great beneficial effect.The authors aim to advance the field of spherical array processing by helping those new to the field to study it efficiently and from a single source, as well as by offering a way for more experienced researchers and engineers to consolidate their understanding, adding either or both of breadth and depth. The level of the presentation corresponds to graduate studies at MSc and PhD level.This book begins with a presentation of some of the essential mathematical a

nd physical theory relevant to spherical microphone arrays, and of an acoustic impulse response simulation method, which can be used to comprehensively evaluate spherical array processing algorithms in reverberant environments.The chapter on acoustic parameter estimation describes the way in which useful descriptions of acoustic scenes can be parameterized, and the signal processing algorithms that can be used to estimate the parameter values using spherical microphone arrays. Subsequent chapters exploit these parameters including in particular measures of direction-of-arrival and of diffuseness of a sound field.The array processing algorithms are then classified into two main classes, each described in a separate chapter. These are signal-dependent and signal-independent beamforming algorithms. Although signal-dependent beamforming algorithms are in theory able to provide better performance compared to the signal-independent algorithms, they are currently rarely used in pract

ice. The main reason for this is that the statistical information required by these algorithms is difficult to estimate. In a subsequent chapter it is shown how the estimated acoustic parameters can be used in the design of signal-dependent beamforming algorithms. This final step closes, at least in part, the gap between theory and practice.

Introduction.- Theoretical Preliminaries of Acoustics.- Spatial Sampling and Signal Transformation.- Spherical Array Acoustic Impulse Response Simulation.- Acoustic Parameter Estimation.- Signal-Independent Array Processing.
1 Introduction
1(10)
1.1 Background and Context
1(1)
1.2 Microphone Array Signal Processing
2(3)
1.3 Organization of the Book
5(6)
References
8(3)
2 Theoretical Preliminaries of Acoustics
11(12)
2.1 Fundamentals of Acoustics
11(3)
2.2 Sound Field Representation Using Spherical Harmonic Expansion
14(5)
2.3 Sign Convention
19(1)
2.4 Sound Intensity
20(1)
2.5
Chapter Summary
21(2)
References
22(1)
3 Spatial Sampling and Signal Transformation
23(16)
3.1 Time-Frequency Domain Processing
23(2)
3.2 Complex Spherical Harmonic Domain Processing
25(2)
3.3 Real Spherical Harmonic Domain Processing
27(2)
3.4 Spatial Sampling
29(7)
3.4.1 Sampling Schemes
32(1)
3.4.2 Array Configurations
33(3)
3.5
Chapter Summary
36(3)
References
36(3)
4 Spherical Array Acoustic Impulse Response Simulation
39(26)
4.1 Allen and Berkley's Image Method
40(2)
4.1.1 Green's Function
40(1)
4.1.2 Image Method
41(1)
4.2 SMIR Method in the Spherical Harmonic Domain
42(6)
4.2.1 Green's Function
43(1)
4.2.2 Neumann Green's Function
44(1)
4.2.3 Scattering Model
44(3)
4.2.4 SMIR Method
47(1)
4.3 Implementation
48(3)
4.3.1 Truncation Error
48(2)
4.3.2 Computational Complexity
50(1)
4.3.3 Algorithm Summary
51(1)
4.4 Examples and Applications
51(10)
4.4.1 Diffuse Sound Field Energy
51(3)
4.4.2 Binaural Interaural Time and Level Differences
54(5)
4.4.3 Mouth Simulator
59(2)
4.5
Chapter Summary and Conclusions
61(4)
Appendix: Spatial Correlation in a Diffuse Sound Field
61(2)
References
63(2)
5 Acoustic Parameter Estimation
65(28)
5.1 Direction of Arrival Estimation
65(15)
5.1.1 Problem Formulation
66(2)
5.1.2 Steered Response Power
68(1)
5.1.3 Intensity-Based Method
69(3)
5.1.4 Subspace Methods
72(5)
5.1.5 Results
77(3)
5.2 Signal-to-Diffuse Ratio Estimation
80(8)
5.2.1 Problem Formulation
80(2)
5.2.2 Coefficient-of-Variation Method
82(1)
5.2.3 Coherence-Based Method
83(2)
5.2.4 Results
85(3)
5.3
Chapter Summary and Conclusions
88(5)
Appendix: Relationship Between the Zero-Order Eigenbeam and the Omnidirectional Reference Microphone Signal
88(2)
References
90(3)
6 Signal-Independent Array Processing
93(20)
6.1 Signal Model
93(4)
6.2 Design Criteria
97(5)
6.2.1 Directivity
97(1)
6.2.2 Front-to-Back Ratio
98(1)
6.2.3 White Noise Gain
98(2)
6.2.4 Spatial Response
100(2)
6.3 Signal-Independent Beamformers
102(9)
6.3.1 Farfield Beamformers
102(8)
6.3.2 Nearfield Beamformers
110(1)
6.4
Chapter Summary
111(2)
References
111(2)
7 Signal-Dependent Array Processing
113(28)
7.1 Signal Model
113(4)
7.2 Performance Measures
117(3)
7.2.1 Speech Distortion Index
117(1)
7.2.2 Noise Reduction Factor
118(1)
7.2.3 Array Gain
119(1)
7.2.4 Mean Square Error
119(1)
7.3 Signal-Dependent Beamformers
120(13)
7.3.1 Maximum SNR Filter
120(1)
7.3.2 Wiener Filter
121(1)
7.3.3 Minimum Variance Distortionless Response Filter
122(4)
7.3.4 Parametric Wiener Filter
126(1)
7.3.5 Linearly Constrained Minimum Variance Filter
127(2)
7.3.6 Generalized Sidelobe Canceller Structure
129(4)
7.4 Relative Transfer Function Estimation
133(4)
7.4.1 Covariance Subtraction Method
134(1)
7.4.2 Generalized Eigenvector Method
134(1)
7.4.3 Temporal Averaging Method
135(2)
7.5
Chapter Summary
137(4)
References
137(4)
8 Parametric Array Processing
141(10)
8.1 Signal Model
142(2)
8.2 Parameter Estimation
144(1)
8.3 Sound Pressure Estimation
145(1)
8.4 Applications
146(3)
8.4.1 Directional Filtering
147(1)
8.4.2 Dereverberation
148(1)
8.5
Chapter Summary
149(2)
References
149(2)
9 Informed Array Processing
151(34)
9.1 Noise Reduction Using Narrowband DOA Estimates
152(17)
9.1.1 Signal Models
153(2)
9.1.2 Tradeoff Beamformer
155(1)
9.1.3 Signal Statistics Estimation
156(3)
9.1.4 Desired Speech Presence Probability Estimation
159(3)
9.1.5 Algorithm Summary
162(1)
9.1.6 Results
163(6)
9.2 Dereverberation Using Signal-to-Diffuse Ratio Estimates
169(13)
9.2.1 Problem Formulation
170(3)
9.2.2 Informed Filter for Dereverberation
173(3)
9.2.3 Relation to Robust MVDR Filter
176(1)
9.2.4 Performance Evaluation
176(6)
9.3
Chapter Summary and Conclusions
182(3)
References
182(3)
Index 185