Muutke küpsiste eelistusi

Acoustic Signal Processing for Telecommunication 2001 ed. [Kõva köide]

Edited by , Edited by
  • Kõva köide
  • Hind: 141,35 €*
  • * hind on lõplik, st. muud allahindlused enam ei rakendu
  • Tavahind: 166,29 €
  • Säästad 15%
  • Raamatu kohalejõudmiseks kirjastusest kulub orienteeruvalt 2-4 nädalat
  • Kogus:
  • Lisa ostukorvi
  • Tasuta tarne
  • Tellimisaeg 2-4 nädalat
  • Lisa soovinimekirja
The current revolution in electronic switching and transport technologies promises a dramatic increase in the intimacy and satisfaction that users will experience over imminent telecommunications networks. However, unless there is a corresponding improvement in the technologies of the acoustics of telecommunications, this promise will soon prove empty. The sense of presence that people feel when together in a room is largely due to the psycho-acoustic cues they sense from the human binaural hearing system evolved over millennia. Modern acoustic signal processing is just now beginning to be able to deliver that same experience to users remotely located from each other. This includes the ability to communicate in full duplex with wider bandwidths and multiple audio streams (e.g. stereo, 3D audio). It also involves locating and separating audio sources, suppressing noise, and using sound to automatically steer video cameras.
Acoustic Signal Processing for Telecommunication presents digital signal processing techniques for telecommunications acoustics that are both cutting-edge and practical. Each chapter presents material that has not appeared in book form before and yet is easily realizable in today's technology. To this end, both new theory and new implementation techniques are presented. Topics include new adaptive filtering algorithms, multi-channel acoustic echo cancellation, noise control, virtual sound, sound source localization for camera tracking, source separation, and microphone arrays.

Muu info

Springer Book Archives
List of Figures xi List of Tables xviii Preface xix Contributing Authors xxi An Introduction to Acoustic Echo and Noise Control 1(22) Steven L. Gay Jacob Benesty Human Perception of Echoes 1(2) The Network Echo Problem 3(3) The Acoustic Echo Problem 6(2) Adaptive Filters for Echo Cancellation 8(9) The LMS and NLMS Algorithms 9(3) Least Squares and Recursive Least Squares Algorithms 12(5) Noise Reduction 17(1) Conclusions 18(5) Part I Mono-Channel Acoustic Echo Cancellation The Fast Affine Projection Algorithm 23(24) Steven L. Gay Introduction 23(1) The Affine Projection Algorithm 24(7) Projections Onto an Affine Subspace 26(3) Convergence and Regularization 29(1) The Connection Between APA and Recursive Least Squares 29(2) Fast Affine Projections 31(8) Fast Residual Echo Vector Calculation 31(2) Fast Adaptive Coefficient Vector Calculation 33(3) Fast Normalized Residual Echo Vector Calculation 36(1) The FAP Algorithm 37(2) Simulations 39(1) Numerical Considerations 40(1) Conclusions 40(7) Appendix: Sliding Windowed Fast Recursive Least Squares 42(5) Subband Acoustic Echo Cancellation Using the FAP-RLS Algorithm: Fixed-Point Implementation Issues 47(20) Mohamed Ghanassi Benoit Champagne Introduction 47(2) Overview of FAP-Based Subband AEC System 49(5) FAP-RLS Algorithm 50(3) Uniform DFT Filter Banks 53(1) Scope of Fixed-Point Study 54(1) Fixed-Point Implementation of FAP-RLS 55(5) Update of Inverse Data Covariance Matrix 56(2) Update of Correlation Vector 58(1) Filtering and Adaptation 58(2) Algorithm Precision 60(1) Fixed-Point WOA Implementation 60(3) DFT or FFT? 60(2) Analysis Bank 62(1) Synthesis Bank 62(1) Evaluation of Complete Algorithm 63(1) Conclusion 63(4) Real-Time Implementation of the Exact Block NLMS Algorithm for Acoustic Echo Control in Hands-Free Telephone Systems 67(14) Bernhard H. Nitsch Introduction 67(1) Block Processing 68(1) The Exact Block NLMS Algorithm 69(2) Reduction of the Signal Delay 71(2) The PEFBNLMS Algorithm 73(1) Performance 74(3) Real-Time Implementation 77(3) Conclusions 80(1) Double-Talk Detection Schemes for Acoustic Echo Cancellation 81(20) Tomas Gansler Jacob Benesty Steven L. Gay Introduction 81(3) Basics of AEC and DTD 84(2) AEC Notations 84(1) The Generic DTD 84(1) A Suggestion to Performance Evaluation of DTDs 85(1) Double-Talk Detection Algorithms 86(9) Geigel Algorithm 86(1) Cross-Correlation Method 86(1) Normalized Cross-Correlation Method 87(1) Coherence Method 88(2) Normalized Cross-correlation Matrix 90(2) Two-Path Model 92(1) DTD Combinations with Robust Statistics 93(2) Discussion 95(6) Part II Multi-Channel Acoustic Echo Cancellation Multi-Channel Sound, Acoustic Echo Cancellation, and Multi-Channel Time-Domain Adaptive Filtering 101(20) Jacob Benesty Tomas Gansler Peter Eneroth Introduction 101(3) Multi-Channel Identification and the Nonuniqueness Problem 104(2) Some Different Solutions for Decorrelation 106(2) The Hybrid Mono/Stereo Acoustic Echo Canceler 108(2) Multi-Channel Time-Domain Adaptive Filters 110(8) The Classical and Factorized Multi-Channel RLS 110(2) The Multi-Channel Fast RLS 112(1) The Multi-Channel LMS Algorithm 113(3) The Multi-Channel APA 116(2) Discussion 118(3) Multi-Channel Frequency-Domain Adaptive Filtering 121(14) Jacob Benesty Dennis R. Morgan Introduction 121(1) Mono-Channel Frequency-Domain Adaptive Filtering Revisited 122(5) Generalization to the Multi-Channel Case 127(2) Application to Acoustic Echo Cancellation and Simulations 129(2) Conclusions 131(4) A Real-time Stereophonic Acoustic Subband Echo Canceler 135(20) Peter Eneroth Steven L. Gay Tomas Gansler Jacob Benesty Introduction 136(1) Acoustic Echo Canceler Components 137(11) Adaptive Algorithm 137(2) Filterbank Design 139(6) Residual Echo Suppression 145(1) Computational Complexity 146(1) Implementation Aspects 147(1) Simulations 148(7) Part III Noise Reduction Techniques with a Single Microphone Subband Noise Reduction Methods for Speech Enhancement 155(26) Eric J. Diethorn Introduction 155(3) Wiener Filtering 158(1) Speech Enhancement by Short-Time Spectral Modification 159(10) Short-Time Fourier Analysis and Synthesis 159(1) Short-Time Wiener Filter 160(1) Power Subtraction 161(1) Magnitude Subtraction 162(1) Parametric Wiener Filtering 163(1) Review and Discussion 164(5) Averaging Techniques for Envelope Estimation 169(3) Moving Average 169(1) Single-Pole Recursion 170(1) Two-Sided Single-Pole Recursion 170(1) Nonlinear Data Processing 171(1) Example Implementation 172(3) Subband Filter Bank Architecture 172(1) A-Posteriori-SNR Voice Activity Detector 173(2) Example 175(1) Conclusion 175(6) Part IV Microphone Arrays Superdirectional Microphone Arrays 181(58) Gary W. Elko Introduction 181(1) Differential Microphone Arrays 182(10) Array Directional Gain 192(1) Optimal Arrays for Spherically Isotropic Fields 193(8) Maximum Gain for Omnidirectional Microphones 193(2) Maximum Directivity Index for Differential Microphones 195(2) Maximimum Front-to-Back Ratio 197(3) Minimum Peak Directional Response 200(1) Beamwidth 201(1) Design Examples 201(21) First-Order Designs 202(5) Second-Order Designs 207(9) Third-Order Designs 216(5) Higher-Order designs 221(1) Optimal Arrays for Cylindrically Isotropic Fields 222(8) Maximum Gain for Omnidirectional Microphones 222(2) Optimal Weights for Maximum Directional Gain 224(1) Solution for Optimal Weights for Maximum Front-to-Back Ratio for Cylindrical Noise 225(5) Sensitivity to Microphone Mismatch and Noise 230(3) Conclusions 233(6) Appendix: Directivity Factor and Room Acoustics 236(3) Microphone Arrays for Video Camera Steering 239(22) Yiteng (Arden) Huang Jacob Benesty Gary W. Elko Introduction 239(2) Time Delay Estimation 241(6) Acoustic Models for the TDE Problem 242(1) The GCC Method 243(1) Adaptive Eigenvalue Decomposition Algorithm 244(3) Source Localization 247(8) Source Localization Problem 247(1) Ideal Maximum Likelihood Locator 248(2) Triangulation Locator 250(1) The Spherical Equations 250(1) CLS and Spherical Intersection (SX) Methods 251(1) Spherical Interpolation (SI) Locator 252(1) One Step Least Squares (OSLS) Locator 253(2) System Implementation 255(2) Summary 257(4) Nonlinear, Model-Based Microphone Array Speech Enhancement 261(22) Michael S. Brandstein Scott M. Griebel Introduction 261(2) Speech Enhancement Methods 263(1) Nonlinear, Model-Based Processing 264(1) A Multi-Channel Speech Enhancement Algorithm 265(10) Algorithm Details 266(8) Simulations 274(1) Conclusion 275(8) Part V Virtual Sound 3D Audio and Virtual Acoustical Environment Synthesis 283(20) Jiashu Chen Introduction 283(1) Sound Localization Cues and Synthetic 3D Audio 284(6) Interaural Cues for Sound Localization 285(1) Head-Related Transfer Function (HRTF) 286(1) Synthetic 3D Audio 287(1) Modeling the Measured HRTFs 288(2) Spatial Feature Extraction and Regularization (SFER) Model for HRTFs 290(5) SFER Model for Head-Related Impulse Response 290(2) TDSFER Model for Multiple 3D Sound Source Positioning 292(3) Computing Architectures Using TDSFER Model 295(4) Multiple Sources with Multiple Reflections 295(3) Single Source with Multiple Reflections 298(1) Specific Issues for VAES Implementation 299(1) Conclusions 299(4) Virtual Sound Using Loudspeakers: Robust Acoustic Crosstalk Cancellation 303(18) Darren B. Ward Gary W. Elko Introduction 303(2) Acoustic Crosstalk Cancellation 305(2) Problem Statement 305(1) Selection of the Design Matrix 306(1) Robustness Analysis 307(6) Robustness Measure 307(1) Analysis of the Design Matrix 307(1) Example of Ear Responses 308(2) Spatial Responses 310(3) Effect of Loudspeaker Position 313(3) A Robust CCS 315(1) Discussion and Conclusions 316(5) Part VI Blind Source Separation An Introduction to Blind Source Separation of Speech Signals 321(10) Jacob Benesty Introduction 321(1) The Information Maximization Principle 322(2) Different Stochastic Gradient Ascent Rules Based on ME 324(2) The Informax Stochastic Gradient Ascent Learning Rule 324(1) The Natural Gradient Algorithm 325(1) A Normalized Natural Gradient Algorithm 325(1) Simulations 326(2) Conclusions 328(3) Index 331