Digital Speech

Coding for Low Bit Rate Communication Systems
2. Revised Auflage 2004
ISBN: 978-0-470-87007-5
Verlag: Wiley

Buch, Englisch, 458 Seiten, Format (B × H): 173 mm x 250 mm, Gewicht: 1012 g

Building on the success of the first edition Digital Speech offers extensive new, updated and revised material based upon the latest research. This Second Edition continues to provide the fundamental technical background required for low bit rate speech coding and the hottest developments in digital speech coding techniques that are applicable to evolving communication systems. - Features new chapters on Pitch Estimation and Voice-Unvoiced Classification of Speech, Harmonic Speech Coding and Multimode Speech Coding
- Presents a comprehensively revised chapter entitled Analysis by Synthesis LPC Coding including specific examples of popular speech coders such as CELP (Code-Excited Linear Predictive) Coding
- Contains an updated chapter on Efficient LPC Quantization Methods including MSVQ and anti-aliasing filtering
- Discusses Voice Activity Detection (VAD) methods
- Offers expanded coverage of speech enhancement techniques such as echo cancellation and noise suppression

Written by a well-known, highly respected academic, this authoritative volume will be invaluable to practising engineers, network designers, computer scientists and advanced students in communications, electrical and electronic engineering.

Autoren/Hrsg.

Kondoz, A M

Fachgebiete

• Technische Wissenschaften Sonstige Technologien | Angewandte Technik Signalverarbeitung, Bildverarbeitung, Scanning
• Mathematik | Informatik EDV | Informatik Informatik Künstliche Intelligenz Spracherkennung, Sprachverarbeitung
• Technische Wissenschaften Elektronik | Nachrichtentechnik Nachrichten- und Kommunikationstechnik
• Mathematik | Informatik EDV | Informatik Informatik Tonsignalverarbeitung

Inhaltsverzeichnis

Preface xiii
Acknowledgements xv

1 Introduction 1

2 Coding Strategies and Standards 5
2.1 Introduction 5
2.2 Speech Coding Techniques 6
2.3 Algorithm Objectives and Requirements 9
2.4 Standard Speech Coders 12
2.5 Summary 18

3 Sampling and Quantization 23
3.1 Introduction 23
3.2 Sampling 23
3.3 Scalar Quantization 26
3.4 Vector Quantization 39
3.5 Summary 54

4 Speech Signal Analysis and Modelling 57
4.1 Introduction 57
4.2 Short-Time Spectral Analysis 57
4.3 Linear Predictive Modelling of Speech Signals 65
4.4 Pitch Prediction 77
4.5 Summary 84

5 Efficient LPC Quantization Methods 87
5.1 Introduction 87
5.2 Alternative Representation of LPC 87
5.3 LPC to LSF Transformation 90
5.4 LSF to LPC Transformation 101
5.5 Properties of LSFs 103
5.6 LSF Quantization 105
5.7 Codebook Structures 110
5.8 MSVQ Performance Analysis 117
5.9 Inter-frame Correlation 121
5.10 Improved LSF Estimation Through Anti-Aliasing Filtering 130
5.11 Summary 146

6 Pitch Estimation and Voiced–Unvoiced Classification of Speech 149
6.1 Introduction 149
6.2 Pitch Estimation Methods 150
6.3 Voiced–Unvoiced Classification 178
6.4 Summary 196

7 Analysis by Synthesis LPC Coding 199
7.1 Introduction 199
7.2 Generalized AbS Coding 200
7.3 Code-Excited Linear Predictive Coding 219
7.4 Summary 258

8 Harmonic Speech Coding 261
8.1 Introduction 261
8.2 Sinusoidal Analysis and Synthesis 262
8.3 Parameter Estimation 263
8.4 Common Harmonic Coders 268
8.5 Summary 275

9 Multimode Speech Coding 277
9.1 Introduction 277
9.2 Design Challenges of a Hybrid Coder 280
9.3 Summary of Hybrid Coders 281
9.4 Synchronized Waveform-Matched Phase Model 285
9.5 Hybrid Encoder 298
9.6 Speech Classification 311
9.7 Hybrid Decoder 319
9.8 Performance Evaluation 320
9.9 Quantization Issues of Hybrid Coder Parameters 322
9.10 Variable Bit Rate Coding 331
9.11 Acoustic Noise and Channel Error Performance 336
9.12 Summary 350

10 Voice Activity Detection 357
10.1 Introduction 357
10.2 Standard VAD Methods 360
10.3 Likelihood-Ratio-Based VAD 368
10.4 Summary 375

11 Speech Enhancement 379
11.1 Introduction 379
11.2 Review of STSA-based Speech Enhancement 381
11.3 Noise Adaptation 402
11.4 Echo Cancellation 406
11.5 Summary 423

Bibliography 426
Index 429

Über Autor(innen)

Professor Kondoz joined the university of Surrey as a PhD. student in October 1984. From 1986 to 1988 he was employed as a research fellow in the communications group. After completing his PhD, in 1988 he was appointed as a lecturer. In 1995 he became a Reader and in 1997 Professor and Deputy Director in the Centre for Communication Systems Research (CCSR).
He has been involved in teaching of digital signal processing, telecommunications theory and source coding in both undergradute and postgraduate levels. In research he has been heading Multimedia Communication Research Group since 1990. To date, Professor Kondoz has supervised 20 successful PhD students in Speech, Video and Channel coding, Source data packetisation, Error resilient speech and video transmission and Mobile multimedia communications. His current research interests are, Low bit rate speech, image and video coding error resilient video transmission, mobile multimedia communications, robust wireless ATM, real-time terminal design and implementation for mobile communications.
Outside the University, Professor Kondoz has been a member of both the IEE and IEEE. He is a CEng and served on E5. He is on EPSRC College for signal processing and communications.