Rainer Martin, Ulrich Heute, Christiane Antweiler

Advances in Digital Speech Transmission

Ed. by Rainer Martin, Ulrich Heute et al.

• Gebundenes Buch

Produktbeschreibung

Building upon the recently published Vary/Martin Digital Speech Transmission (Wiley), Advances in Digital Speech Transmission includes many interesting developments and challenging applications that were omitted in the Vary/Martin volume due to page constraints. Featuring contributions from internationally recognized researchers, this invaluable reference for anybody working in the areas of speech processing and/or speech transmission technology provides a well-balanced cross section of topics of high contemporary interest, such as speech coding for Voice over IP, blind source separation, digital hearing aids, and speech processing for automatic speech recognition.

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Speech processing and speech transmission technology are expanding fields of active research. New challenges arise from the 'anywhere, anytime' paradigm of mobile communications, the ubiquitous use of voice communication systems in noisy environments and the convergence of communication networks toward Internet based transmission protocols (such as Voice over IP). As a consequence, new speech coding, new enhancement and error concealment, and new quality assessment methods are emerging.

Advances in Digital Speech Transmission provides an up-to-date overview of the field, including topics such as speech coding in heterogeneous communication networks, wideband coding, and the quality assessment of modern speech transmission systems.

Provides an insight into the latest developments in speech processing and speech transmission, making it an essential reference to those working in these fields

Offers a balanced overview of technology and applications

Presents a unique in-depth discussion of soft-decision source decoding, joint source-channel coding and Turbo DeCodulation techniques

Explains speech signal processing in hearing instruments and human-machine interfaces from applications point of view

Covers speech coding for Voice over IP, source localization, digital hearing aids and speech processing for automatic speech recognition

Advances in Digital Speech Transmission serves as an essential link between the basics and the type of technology and applications (prospective) engineers work on in industry labs and academia. The book will also be of interest to advanced students, researchers, and other professionals who need to brush up their knowledge in this field.

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Produktdetails

• Verlag: Wiley & Sons
• 1. Auflage
• Seitenzahl: 572
• Erscheinungstermin: 1. März 2008
• Englisch
• Abmessung: 249mm x 175mm x 36mm
• Gewicht: 1125g
• ISBN-13: 9780470517390
• ISBN-10: 0470517395
• Artikelnr.: 23158027

Autorenporträt

Dr. Ing. Rainer Martin is a Professor of Information Technology at Ruhr University and Head of the Institute of Communication Acoustics, Bochum, Germany. His research interests are signal processing for voice communication systems, acoustics, and human-machine interfaces. He has worked on algorithms for noise reduction, acoustic echo cancellation, microphone arrays, and speech recognition. He is coauthor of the book Vary/Martin "Digital Speech Transmission", John Wiley, 2006. Ulrich Heute is a Professor for circuit and system theory at Christian-Albrechts University, Kiel. His research interests include digital signal processing, filters and filter banks, and spectral analysis, with applications in medical, audio, and, especially, speech-signal processing (combined source and channel coding, enhancement, modeling, speaker characterization, and instrumental quality assessment). Christiane Antweiler is a senior scientist at the Institute of Communication Systems and Data Processing of the RWTH Aachen University. Her interests are the design and implementation of digital signal and speech processing algorithms for real-time applications, and her special focus lies on speech coding for cellular mobile radio and the enhancement of digital speech signals. Furthermore she is interested in algorithms for system identification, in adaptive filter theory and in digital signal processing algorithms for medical diagnostics.

Inhaltsangabe

List of Contributors xxi
Preface xxvii
1 Introduction 1
Rainer Martin, Ulrich Heute, Christiane Antweiler
I Speech Quality Assessment 7
2 Speech-Transmission Quality: Aspects and Assessment for Wideband vs.
Narrowband Signals 9
Ulrich Heute
2.1 Introduction 9
2.2 Speech Signals . 10
2.3 Telephone-Band Speech Signals 11
2.4 Wideband Speech Signals 14
2.5 Speech-Quality Assessment 25
2.6 Wideband Speech-Quality Assessment 30
2.7 Concluding Remarks 43
Bibliography 44
3 Parametric Quality Assessment of Narrowband Speech in Mobile
Communication Systems 51
Marc Werner
3.1 Introduction 51
3.2 Simulations of GSM and UMTS Speech Transmissions 58
3.3 Speech Quality Measures based on Transmission Parameters 65
3.4 Discussion and Conclusions 73
Bibliography 73
II Adaptive Algorithms in Acoustic Signal Processing 77
4 Kalman Filtering in Acoustic Echo Control: A Smooth Ride on a Rocky Road
79
Gerald Enzner
4.1 Introduction 79
4.2 A Comprehensive Theory of Acoustic Echo Control 85
4.3 The Kalman Filter for Conditional Mean and Covariance Estimation 90
4.4 AEC Performance of the Frequency-Domain Adaptive Kalman Filter 100
4.5 Discussion and Conclusions 102
Bibliography 103
5 Noise Reduction - Statistical Analysis and Control of Musical Noise 107
Colin Breithaupt, Rainer Martin
5.1 Introduction 107
5.2 Speech Enhancement in the DFT Domain 109
5.3 Measurement and Assessment of Unnatural Fluctuations 115
5.4 Avoidance of Processing Artifacts 120
5.5 Control of Spectral Fluctuations in the Cepstral Domain 123
5.6 Discussion and Conclusions 128
5.7 Acknowledgements 129
5.8 Appendix 129
Bibliography 131
6 Acoustic Source Localization with Microphone Arrays 135
Nilesh Madhu, Rainer Martin
6.1 Introduction 135
6.2 SignalModel 136
6.3 Localization Approach Taxonomy 141
6.4 Indirect Localization Approaches 141
6.5 Direct Localization Approaches 148
6.6 Evaluation of Localization Algorithms 156
6.7 Conclusions 166
Bibliography 166
7 Multi-Channel System Identification with Perfect Sequences - Theory and
Applications - 171
Christiane Antweiler
7.1 Introduction 171
7.2 System Identification with Perfect Sequences 174
7.3 Multi-Channel System Identification 185
7.4 Applications 191
7.5 Discussion and Conclusions 195
Bibliography 195
III Speech Coding for Heterogeneous Networks 199
8 Embedded Speech Coding: From G.711 to G.729.1 201
Bernd Geiser, Stéphane Ragot, Hervé Taddei
8.1 Introduction 201
8.2 Theory and Tools of Embedded Speech Coding 203
8.3 Embedded Speech Coding Methods 212
8.4 Standardized Embedded Speech Coders 219
8.5 Network Aspects of Embedded Speech Coding 232
8.6 Conclusions and Perspectives 237
Bibliography 238
9 Backwards Compatible Wideband Telephony 249
Peter Jax
9.1 Introduction 249
9.2 From Narrowband Telephony to Wideband Telephony 250
9.3 Stand-Alone Bandwidth Extension 254
9.4 Embedded Wideband Coding Using Bandwidth Extension Techniques 257
9.5 Combination of Bandwidth Extension with Watermarking 262
9.6 Advanced Transmission of Highband Parameters 267
9.7 Conclusions 274
Bibliography 274
IV Joint Source-Channel Coding 279
10 Parameter Models and Estimators in Soft Decision Source Decoding 281
Tim Fingscheidt
10.1 Introduction 281
10.2 Overview to Soft Decision Source Decoding 283
10.3 The Markovian Parameter Model 287
10.4 Basic Extrapolative Estimators 290
10.5 Joint Extrapolative Estimation of Two Different Parameters 298
10.6 Extrapolative Estimation with Repeated Parameter Transmission 301
10.7 Interpolative Estimation of a Parameter 304
10.8 Discussion and Conclusions 307
Bibliography 307
11 Optimal MMSE Estimation for Vector Sources with Spatially and Temporally
Correlated Elements 311
Stefan Heinen, Marc Adrat
11.1 Introduction 311
11.2 Source Model 312
11.3 Transmission Channel 316
11.4 Optimal MMSE Parameter Estimator 316
11.5 Near-Optimal MMSE Parameter Estimator 320
11.6 Illustrative Comparison 323
11.7 Simulation Results 325
11.8 Conclusions 327
Bibliography 327
12 Source Optimized Channel Codes & Source Controlled Channel Decoding 329
Stefan Heinen, Thomas Hindelang
12.1 Introduction 329
12.2 The Transmission System Used as Reference 330
12.3 Source Optimized Channel Coding (SOCC) 332
12.4 Source Controlled Channel Decoding (SCCD) 341
12.5 Comparison of SOCC versus SCCD 357
12.6 Conclusions 362
Bibliography 363
13 Iterative Source-Channel Decoding & Turbo DeCodulation 365
Marc Adrat, Thorsten Clevorn, Laurent Schmalen
13.1 Introduction 365
13.2 The Key of the Turbo Principle: Extrinsic Information 366
13.3 Iterative Source-Channel Decoding (ISCD) 379
13.4 Turbo DeCodulation (TDeC) 387
13.5 Conclusions 394
Bibliography 395
V Speech Processing in Hearing Instruments 399
14 Binaural Signal Processing in Hearing Aids 401
Volkmar Hamacher, Ulrich Kornagel, Thomas Lotter, Henning Puder
14.1 Introduction 401
14.2 Wireless System for Hearing Aids 405
14.3 Binaural Classification Systems 410
14.4 Binaural Beamformer 415
14.5 Blind Source Separation (BSS): An Application for a Binaural
Directional Microphone Array in Hearing Aids 422
14.6 Conclusions 427
Bibliography 428
15 Auditory-profile-based Physical Evaluation of Multi-microphone Noise
Reduction Techniques in Hearing Instruments 431
Koen Eneman, Arne Leijon, Simon Doclo, Ann Spriet, Marc Moonen, Jan Wouters
15.1 Introduction 431
15.2 Multi-microphone Noise Reduction in Hearing Instruments 434
15.3 Auditory-profile-based Physical Evaluation 441
15.4 Test Conditions 449
15.5 Simulation Results 450
15.6 Discussion 452
15.7 Conclusions 455
Bibliography 456
VI Speech Processing for Human-Machine Interfaces 459
16 Automatic Speech Recognition in Adverse Acoustic Conditions 461
Hans-Günter Hirsch
16.1 Introduction 461
16.2 Structure of Speech Recognition Systems 462
16.3 Acoustic Scenarios during Speech Input 468
16.4 Improving the Recognition Performance in Adverse Conditions 476
16.5 Conclusions 493
Bibliography 494
17 Speaker Classification for Next-Generation Voice-Dialog Systems 497
Felix Burkhardt, Florian Metze, Joachim Stegmann
17.1 Introduction 497
17.2 Speaker Classification 498
17.3 Detection of Age and Gender 505
17.4 Detection of Anger 510
17.5 Applications in IVR Systems 517
17.6 Discussion and Conclusion 523
Bibliography 525
Index 529
Permissions List 541

Rezensionen

"This book is recommended to anyone interested in the details of the latest developments in the area of digital-speech enhancement and processing." ( Computing Reviews , August 8, 2008)