Digital Speech: Coding for Low Bit Rate Communication Systems

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.

Table of Contents:
Preface. Acknowledgements. 1 Introduction. 2 Coding Strategies and Standards. 2.1 Introduction. 2.2 Speech Coding Techniques. 2.3 Algorithm Objectives and Requirements. 2.4 Standard Speech Coders. 2.5 Summary. Bibliography. 3 Sampling and Quantization. 3.1 Introduction. 3.2 Sampling. 3.3 Scalar Quantization. 3.4 Vector Quantization. 3.5 Summary. Bibliography. 4 Speech Signal Analysis and Modelling. 4.1 Introduction. 4.2 Short-Time Spectral Analysis. 4.3 Linear Predictive Modelling of Speech Signals. 4.4 Pitch Prediction. 4.5 Summary. Bibliography. 5 Efficient LPC QuantizationMethods. 5.1 Introduction. 5.2 Alternative Representation of LPC. 5.3 LPC to LSF Transformation. 5.4 LSF to LPC Transformation. 5.5 Properties of LSFs. 5.6 LSF Quantization. 5.7 Codebook Structures. 5.8 MSVQ Performance Analysis. 5.9 Inter-frame Correlation. 5.10 Improved LSF Estimation Through Anti-Aliasing Filtering. 5.11 Summary. Bibliography. 6 Pitch Estimation and Voiced–Unvoiced Classification of Speech. 6.1 Introduction. 6.2 Pitch Estimation Methods. 6.3 Voiced–Unvoiced Classification. 6.4 Summary. Bibliography. 7 Analysis by Synthesis LPC Coding. 7.1 Introduction. 7.2 Generalized AbS Coding. 7.3 Code-Excited Linear Predictive Coding. 7.4 Summary. Bibliography. 8 Harmonic Speech Coding. 8.1 Introduction. 8.2 Sinusoidal Analysis and Synthesis. 8.3 Parameter Estimation. 8.4 Common Harmonic Coders. 8.5 Summary. Bibliography. 9 Multimode Speech Coding. 9.1 Introduction. 9.2 Design Challenges of a Hybrid Coder. 9.3 Summary of Hybrid Coders. 9.4 Synchronized Waveform-Matched Phase Model. 9.5 Hybrid Encoder. 9.6 Speech Classification. 9.7 Hybrid Decoder. 9.8 Performance Evaluation. 9.9 Quantization Issues of Hybrid Coder Parameters. 9.10 Variable Bit Rate Coding. 9.11 Acoustic Noise and Channel Error Performance. 9.12 Summary. Bibliography. 10 Voice Activity Detection. 10.1 Introduction. 10.2 Standard VAD Methods. 10.3 Likelihood-Ratio-Based VAD. 10.4 Summary. Bibliography. 11 Speech Enhancement. 11.1 Introduction. 11.2 Review of STSA-based Speech Enhancement. 11.3 Noise Adaptation. 11.4 Echo Cancellation. 11.5 Summary. Bibliography. Index.

About the Author :
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.