Hänsler / Schmidt Speech and Audio Processing in Adverse Environments

1. Auflage 2008
ISBN: 978-3-540-70602-1
Verlag: Springer Berlin Heidelberg
Format: PDF
Kopierschutz: 1 - PDF Watermark

E-Book, Englisch, 736 Seiten

Reihe: Signals and Communication Technology

Users of signal processing systems are never satis?ed with the system they currently use. They are constantly asking for higher quality, faster perf- mance, more comfort and lower prices. Researchers and developers should be appreciative for this attitude. It justi?es their constant e?ort for improved systems. Better knowledge about biological and physical interrelations c- ing along with more powerful technologies are their engines on the endless road to perfect systems. This book is an impressive image of this process. After 'Acoustic Echo 1 and Noise Control' published in 2004 many new results lead to 'Topics in 2 Acoustic Echo and Noise Control' edited in 2006 . Today - in 2008 - even morenew?ndingsandsystemscouldbecollectedinthisbook.Comparingthe contributions in both edited volumes progress in knowledge and technology becomesclearlyvisible:Blindmethodsandmultiinputsystemsreplace'h- ble' low complexity systems. The functionality of new systems is less and less limited by the processing power available under economic constraints. The editors have to thank all the authors for their contributions. They cooperated readily in our e?ort to unify the layout of the chapters, the ter- nology, and the symbols used. It was a pleasure to work with all of them. Furthermore, it is the editors concern to thank Christoph Baumann and the Springer Publishing Company for the encouragement and help in publi- ing this book.

Autoren/Hrsg.

Hänsler, Eberhard

Schmidt, Gerhard

Inhaltsverzeichnis

1;Preface;6
2;Contents;7
3;List of Contributors;18
4;Abbreviations and Acronyms;20
5;1 Introduction;26
5.1;1.1 Overview about the Book;27
6;Part I Speech Enhancement;30
6.1;2 Low Delay Filter-Banks for Speech and Audio Processing;31
6.1.1;2.1 Introduction;31
6.1.2;2.2 Analysis-Synthesis Filter-Banks;33
6.1.3;2.3 The Filter-Bank Equalizer;47
6.1.4;2.4 Further Measures for Signal Delay Reduction;62
6.1.5;2.5 Application to Noise Reduction;67
6.1.6;2.6 Conclusions;73
6.1.7;References;74
6.2;3 A Pre-Filter for Hands-Free Car Phone Noise Reduction: Suppression of Harmonic Engine Noise Components;80
6.2.1;3.1 Introduction;80
6.2.2;3.2 Analysis of the Different Car Noise Components;81
6.2.3;3.3 Engine Noise Removal Based on Notch Filters;85
6.2.4;3.4 Compensation of Engine Harmonics with Adaptive Filters;90
6.2.5;3.5 Evaluation and Comparison of the Results Obtained by the Notch Filter and the Compensation Approach;101
6.2.6;3.6 Conclusions and Summary;102
6.2.7;References;104
6.3;4 Model-Based Speech Enhancement;105
6.3.1;4.1 Introduction;105
6.3.2;4.2 Conventional Speech Enhancement Schemes;107
6.3.3;4.3 Speech Enhancement Schemes Based on Nonlinearities;109
6.3.4;4.4 Speech Enhancement Schemes Based on Speech Reconstruction;113
6.3.5;4.5 Combining the Reconstructed and the Noise Suppressed Signal;140
6.3.6;4.6 Summary and Outlook;149
6.3.7;References;149
6.4;5 Bandwidth Extension of Telephony Speech;151
6.4.1;5.1 Introduction;151
6.4.2;5.2 Organization of the Chapter;153
6.4.3;5.3 Basics;154
6.4.4;5.4 Non-Model-Based Algorithms for Bandwidth Extension;165
6.4.5;5.5 Model-Based Algorithms for Bandwidth Extension;169
6.4.6;5.6 Evaluation of Bandwidth Extension Algorithms;192
6.4.7;5.7 Conclusions;197
6.4.8;References;198
6.5;6 Dereverberation and Residual Echo Suppression in Noisy Environments;201
6.5.1;6.1 Introduction;202
6.5.2;6.2 Problem Formulation;204
6.5.3;6.3 OM-LSA Estimator for Multiple Interferences;207
6.5.4;6.4 Dereverberation of Noisy Speech Signals;211
6.5.5;6.5 Residual Echo Suppression;219
6.5.6;6.6 Joint Suppression of Reverberation, Residual Echo, and Noise;226
6.5.7;6.7 Experimental Results;228
6.5.8;6.8 Summary and Outlook;239
6.5.9;References;240
6.6;7 Low Distortion Noise Cancellers – Revival of a Classical Technique;244
6.6.1;7.1 Introduction;244
6.6.2;7.2 Distortions in Widrow’s Adaptive Noise Canceller;245
6.6.3;7.3 Paired Filter (PF) Structure;248
6.6.4;7.4 Crosstalk Resistant ANC and Cross-Coupled Structure;254
6.6.5;7.5 Cross-Coupled Paired Filter (CCPF) Structure;257
6.6.6;7.6 Generalized Cross-Coupled Paired Filter (GCCPF) Structure;262
6.6.7;7.7 Demonstration in a Personal Robot;276
6.6.8;7.8 Conclusions;276
6.6.9;References;278
7;Part II Echo Cancellation;280
7.1;8 Nonlinear Echo Cancellation Based on Spectral Shaping;281
7.1.1;8.1 Introduction;281
7.1.2;8.2 Frequency-Domain Model of Highly Nonlinear Residual Echo;282
7.1.3;8.3 Echo Canceller Based on the New Residual Echo Model;288
7.1.4;8.4 Evaluations;291
7.1.5;8.5 DSP Implementation and Real-Time Evaluation;294
7.1.6;8.6 Conclusions;294
7.1.7;References;295
8;Part III Signal and System Quality Evaluation;298
8.1;9 Telephone-Speech Quality;299
8.1.1;9.1 Telephone-Speech Signals;299
8.1.2;9.2 Speech-Signal Quality;301
8.1.3;9.3 Speech-Quality Assessment;304
8.1.4;9.4 Compound-System Quality Prediction;305
8.1.5;9.5 Auditory Total-Quality Assessment;306
8.1.6;9.6 Auditory Quality-Attribute Analysis;310
8.1.7;9.7 Instrumental Total-Quality Measurement;318
8.1.8;9.8 Instrumental Attribute-Based Quality Measurements;332
8.1.9;9.9 Conclusions, Outlook, and Final Remarks;343
8.1.10;References;344
8.2;10 Evaluation of Hands-free Terminals;350
8.2.1;10.1 Introduction;350
8.2.2;10.2 Quality Assessment of Hands-free Terminals;351
8.2.3;10.3 Subjective Methods for Determining the Communicational Quality;353
8.2.4;10.4 Test Environment;361
8.2.5;10.5 Test Signals and Analysis Methods;365
8.2.6;10.6 Result Representation;376
8.2.7;10.7 Related Aspects;379
8.2.8;References;386
9;Part IV Multi-Channel Processing;389
9.1;11 Correlation-Based TDOA-Estimation for Multiple Sources in Reverberant Environments;390
9.1.1;11.1 Introduction;390
9.1.2;11.2 Analysis of TDOA Ambiguities;392
9.1.3;11.3 Estimation of Direct Path TDOAs;399
9.1.4;11.4 Consistent TDOA Graphs;406
9.1.5;11.5 Experimental Results;415
9.1.6;11.6 Summary;423
9.1.7;References;424
9.2;12 Microphone Calibration for Multi-Channel Signal Processing;426
9.2.1;12.1 Introduction;426
9.2.2;12.2 Beamforming with Ideal Microphones;427
9.2.3;12.3 Microphone Mismatch and its E.ect on Beamforming;436
9.2.4;12.4 Calibration Techniques and their Limits for Real- World Applications;441
9.2.5;12.5 Self-Calibration Techniques;458
9.2.6;12.6 Summary;468
9.2.7;12.A Experimental Determination of the Directivity Index;469
9.2.8;References;474
9.3;13 Convolutive Blind Source Separation for Noisy Mixtures;477
9.3.1;13.1 Introduction;477
9.3.2;13.2 Blind Source Separation for Acoustic Mixtures Based on the TRINICON Framework;481
9.3.3;13.3 Extensions for Blind Source Separation in Noisy Environments;498
9.3.4;13.4 Conclusions;526
9.3.5;References;527
9.4;14 Binaural Speech Segregation;533
9.4.1;14.1 Introduction;533
9.4.2;14.2 T–F Masks for CASA;536
9.4.3;14.3 Anechoic Binaural Segregation;537
9.4.4;14.4 Reverberant Binaural Segregation;541
9.4.5;14.5 Evaluation;544
9.4.6;14.6 Concluding Remarks;551
9.4.7;14.7 Acknowledgments;554
9.4.8;References;554
9.5;15 Spatio-Temporal Adaptive Inverse Filtering in the Wave Domain;558
9.5.1;15.1 Introduction;558
9.5.2;15.2 Problem Description;560
9.5.3;15.3 Computation of Compensation Filters;570
9.5.4;15.4 Eigenspace Adaptive Filtering;575
9.5.5;15.5 Wave-Domain Adaptive Filtering;577
9.5.6;15.6 Application of WDAF to Adaptive Inverse Filtering Problems;583
9.5.7;15.7 Conclusions;585
9.5.8;References;587
10;Part V Selected Applications;591
10.1;16 Virtual Hearing;592
10.1.1;16.1 Previous Work;593
10.1.2;16.2 VirtualHearing;597
10.1.3;16.3 Room Acoustic Model;598
10.1.4;16.4 HRTF Simulation;602
10.1.5;16.5 Neural Model;605
10.1.6;16.6 The Software and Interface;610
10.1.7;16.7 Software Testing;614
10.1.8;16.8 Future Work and Conclusions;615
10.1.9;References;618
10.2;17 Dynamic Sound Control Algorithms in Automobiles;620
10.2.1;17.1 Introduction;620
10.2.2;17.2 Previous Systems – Description and Analysis;624
10.2.3;17.3 Spectrum-Based Dynamic Equalization Control;650
10.2.4;17.4 Conclusion and Outlook;675
10.2.5;17.5 Acknowledgement;678
10.2.6;References;679
10.3;18 Towards Robust Distant-Talking Automatic Speech Recognition in Reverberant Environments;684
10.3.1;18.1 Introduction;684
10.3.2;18.2 The Distant-Talking ASR Scenario;685
10.3.3;18.3 How to Deal with Reverberation in ASR Systems?;688
10.3.4;18.4 Effect of Reverberation in the Feature Domain;696
10.3.5;18.5 Signal Dereverberation and Beamforming;700
10.3.6;18.6 Robust Features;704
10.3.7;18.7 Model Training and Adaptation;705
10.3.8;18.8 Reverberation Modeling for Speech Recognition;707
10.3.9;18.9 Summary and Conclusions;727
10.3.10;References;728
11;Index;734