Audio Signal Processing for Next-Generation Multimedia Communication Systems presents cutting-edge digital signal processing theory and implementation techniques for problems including speech acquisition and enhancement using microphone arrays, new adaptive filtering algorithms, multichannel acoustic echo cancellation, sound source tracking and separation, audio coding, and realistic sound stage reproduction. This book's focus is almost exclusively on the processing, transmission, and presentation of audio and acoustic signals in multimedia communications for telecollaboration where immersive acoustics will play a great role in the near future.
Author(s): Huang Y., Benesty J.
Year: 2004
Language: English
Pages: 389
Team DDU......Page 1
Contents......Page 6
Preface......Page 12
Contributing Authors......Page 14
1. Multimedia Communications......Page 16
2. Challenges and Opportunities......Page 18
3. Organization of the Book......Page 19
Part I Speech Acquisition and Enhancement......Page 24
1. Introduction......Page 26
2. Differential Microphone Arrays......Page 27
3. Array Directional Gain......Page 37
4.1 Maximum Directional Gain......Page 39
4.2 Maximum Directivity Index for Differential Microphones......Page 43
4.3 Maximum Front-to-Back Ratio......Page 47
4.4 Minimum Peak Directional Response......Page 52
5. Design Examples......Page 54
5.1 First-Order Designs......Page 55
5.2 Second-Order Designs......Page 59
5.3 Third-Order Designs......Page 67
5.4 Higher-Order designs......Page 73
6. Sensitivity to Microphone Mismatch and Noise......Page 75
7. Conclusions......Page 79
1. Introduction......Page 82
2. Fundamental Concept......Page 84
3. The Eigenbeamformer......Page 86
3.2 The Eigenbeams......Page 88
3.3 The Modal Coefficients......Page 89
4.2 Steering Unit......Page 91
5. Robustness Measure......Page 92
6.2 Optimum Beampattern Design......Page 94
7. Measurements......Page 98
8. Summary......Page 101
9. Appendix A......Page 104
1. Introduction......Page 106
2. Wiener Filtering......Page 109
3.1 Short-Time Fourier Analysis and Synthesis......Page 110
3.2 Short-Time Wiener Filter......Page 111
3.3 Power Subtraction......Page 112
3.4 Magnitude Subtraction......Page 113
3.5 Parametric Wiener Filtering......Page 114
3.6 Review and Discussion......Page 115
4. Averaging Techniques for Envelope Estimation......Page 119
4.2 Single-Pole Recursion......Page 120
4.3 Two-Sided Single-Pole Recursion......Page 121
5. Example Implementation......Page 122
5.1 Subband Filter Bank Architecture......Page 123
5.2 A-Posteriori-SNR Voice Activity Detector......Page 124
6. Conclusion......Page 126
Part II Acoustic Echo Cancellation......Page 132
5 Adaptive Algorithms for MIMO Acoustic Echo Cancellation......Page 134
1. Introduction......Page 135
2.1 Normal Equations......Page 136
2.2 The Nonuniqueness Problem......Page 139
2.3 The Impulse Response Tail Effect......Page 140
2.4 Some Different Solutions for Decorrelation......Page 141
3. The Classical and Factorized Multichannel RLS......Page 143
4. The Multichannel Fast RLS......Page 145
5.1 Classical Derivation......Page 147
5.2 Improved Version......Page 148
6.1 The Straightforward Multichannel APA......Page 149
6.2 The Improved Two-Channel APA......Page 150
6.3 The Improved Multichannel APA......Page 151
7. The Multichannel Exponentiated Gradient Algorithm......Page 152
8. The Multichannel Frequency-domain Adaptive Algorithm......Page 157
9. Conclusions......Page 160
1. Introduction......Page 164
2.2 The Generic DTD......Page 167
2.3 A Suggestion to Performance Evaluation of DTDs......Page 168
3.2 The Cross-Correlation Method......Page 169
3.3 The Normalized Cross-Correlation Method......Page 170
3.4 The Coherence Method......Page 172
3.5 The Normalized Cross-correlation Matrix......Page 174
3.6 The Two-Path Model......Page 176
3.7 DTD Combinations with Robust Statistics......Page 178
4. Comparison of DTDs by Means of the ROC......Page 180
5. Discussion......Page 182
7 The WinEC: A Real-Time Hands-Free Stereo Communication System......Page 186
1. Introduction......Page 187
2.1 The Audio Module......Page 188
2.2 The Network Module......Page 191
3. Algorithms of the Echo Canceler Module......Page 192
3.1 Adaptive Filter Algorithm......Page 193
4. Residual Echo and Noise Suppression......Page 196
4.1 Masking Threshold for Residual Echo in Noise......Page 198
4.2 Analysis of Echo Suppression Requirements......Page 199
5. Simulations......Page 201
6.2 Multi-Point Communication......Page 204
6.3 Transatlantic Teleconference in Stereo......Page 205
7. Discussion......Page 206
Part III Sound Source Tracking and Separation......Page 210
8 Time Delay Estimation......Page 212
1. Introduction......Page 213
2.1 Ideal Propagation Model......Page 215
2.2 Multipath Model......Page 216
3. Generalized Cross-Correlation Method......Page 217
4.1 Spatial Prediction Technique......Page 219
4.2 Time Delay Estimation Using Spatial Prediction......Page 222
4.3 Other Information from the Spatial Correlation Matrix......Page 223
5. Adaptive Eigenvalue Decomposition Algorithm......Page 226
6.1 Principle......Page 228
6.2 Time-Domain Multichannel LMS Approach......Page 229
6.3 Frequency-Domain Adaptive Algorithms......Page 230
7.1 Experimental Setup......Page 234
7.2 Performance Measure......Page 235
7.3 Experimental Results......Page 236
8. Conclusions......Page 238
9 Source Localization......Page 244
1. Introduction......Page 245
2. Source Localization Problem......Page 247
3. Measurement Model and Cramer-Rao Lower Bound for Source Localization......Page 249
4. Maximum Likelihood Estimator......Page 250
5. Least Squares Estimators......Page 251
5.1 The Least Squares Error Criteria......Page 252
5.3 Spherical Interpolation (SI) Estimator......Page 254
5.4 Linear-Correction Least Squares Estimator......Page 255
6. Example System Implementation......Page 261
7. Source Localization Examples......Page 262
8. Conclusions......Page 264
10 Blind Source Separation for Convolutive Mixtures: A Unified Treatment......Page 270
1. Introduction......Page 271
2.1 Matrix Notation for Convolutive Mixtures......Page 274
2.2 Cost Function and Algorithm Derivation......Page 276
2.3 Equivariance Property and Natural Gradient......Page 278
2.4 Special Cases and Links to Known Time-Domain Algorithms......Page 280
3.1 General Frequency-Domain Formulation......Page 286
3.2 Natural Gradient in the Frequency Domain......Page 291
3.3 Special Cases and Links to Known Frequency-Domain Algorithms......Page 292
4. Weighting Function......Page 299
4.2 On-line Implementation......Page 300
5. Experiments and Results......Page 301
6. Conclusions......Page 304
Part IV Audio Coding and Realistic Sound Stage Reproduction......Page 310
1. Introduction......Page 312
2. Psycho-Acoustics......Page 313
3. Filter Banks......Page 315
3.1 Polyphase Formulation......Page 316
3.2 Modulated Filter Banks......Page 317
3.3 Block Switching......Page 323
4. Current and Basic Coder Structures......Page 324
5. Stereo Coding......Page 326
6. Low Delay Audio Coding......Page 329
7. Conclusions......Page 336
12 Sound Field Synthesis......Page 338
1. Introduction......Page 339
2.1 Physical Foundation of Wave Field Synthesis......Page 340
2.2 Wave Field Synthesis Based Sound Reproduction......Page 342
3.1 Data-Based Rendering......Page 344
3.2 Model-Based Rendering......Page 345
4. Wave Field Analysis......Page 346
5. Loudspeaker and Listening Room Compensation......Page 348
5.1 Listening Room Compensation......Page 349
5.2 Loudspeaker Compensation......Page 352
6.1 Acquisition of Source Signals......Page 354
6.2 Sound Stage Reproduction Using Wave Field Synthesis......Page 356
7. Summary......Page 357
1. Introduction......Page 360
1.1 Scope......Page 362
2.1 Interaural Coordinate System......Page 363
2.2 Interaural Differences......Page 364
2.3 Spectral Cues......Page 366
2.4 Distance Cues......Page 367
3.1 The HRTF......Page 368
3.2 Room Acoustics......Page 372
4.1 HRTF Measurement......Page 373
4.2 HRTF Modelling......Page 375
4.3 Virtual Spatial Sound Rendering......Page 378
5. Conclusions......Page 381
Index......Page 386
