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9780470870082

Digital Speech Coding for Low Bit Rate Communication Systems

by
  • ISBN13:

    9780470870082

  • ISBN10:

    0470870087

  • Edition: 2nd
  • Format: Paperback
  • Copyright: 2004-10-29
  • Publisher: WILEY
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Summary

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.

Author Biography

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.

Table of Contents

Preface xiii
Acknowledgements xv
Introduction
1(4)
Coding Strategies and Standards
5(18)
Introduction
5(1)
Speech Coding Techniques
6(3)
Parametric Coders
7(1)
Waveform-approximating Coders
8(1)
Hybrid Coding of Speech
8(1)
Algorithm Objectives and Requirements
9(3)
Quality and Capacity
9(1)
Coding Delay
10(1)
Channel and Background Noise Robustness
10(1)
Complexity and Cost
11(1)
Tandem Connection and Transcoding
11(1)
Voiceband Data Handling
11(1)
Standard Speech Coders
12(6)
ITU-T Speech Coding Standard
12(1)
European Digital Cellular Telephony Standards
13(1)
North American Digital Cellular Telephony Standards
14(1)
Secure Communication Telephony
14(1)
Satellite Telephony
15(1)
Selection of a Speech Coder
15(3)
Summary
18(5)
Bibliography
18(5)
Sampling and Quantization
23(34)
Introduction
23(1)
Sampling
23(3)
Scalar Quantization
26(13)
Quantization Error
27(1)
Uniform Quantizer
28(1)
Optimum Quantizer
29(3)
Logarithmic Quantizer
32(1)
Adaptive Quantizer
33(3)
Differential Quantizer
36(3)
Vector Quantization
39(15)
Distortion Measures
42(1)
Codebook Design
43(1)
Codebook Types
44(8)
Training, Testing and Codebook Robustness
52(2)
Summary
54(3)
Bibliography
54(3)
Speech Signal Analysis and Modelling
57(30)
Introduction
57(1)
Short-Time Spectral Analysis
57(8)
Role of Windows
58(7)
Linear Predictive Modelling of Speech Signals
65(12)
Source Filter Model of Speech Production
65(2)
Solutions to LPC Analysis
67(7)
Practical Implementation of the LPC Analysis
74(3)
Pitch Prediction
77(7)
Periodicity in Speech Signals
77(1)
Pitch Predictor (Filter) Formulation
78(6)
Summary
84(3)
Bibliography
84(3)
Efficient LPC Quantization Methods
87(62)
Introduction
87(1)
Alternative Representation of LPC
87(3)
LPC to LSF Transformation
90(11)
Complex Root Method
95(1)
Real Root Method
95(3)
Ratio Filter Method
98(2)
Chebyshev Series Method
100(1)
Adaptive Sequential LMS Method
100(1)
LSF to LPC Transformation
101(2)
Direct Expansion Method
101(1)
LPC Synthesis Filter Method
102(1)
Properties of LSFs
103(2)
LSF Quantization
105(5)
Distortion Measures
106(1)
Spectral Distortion
106(1)
Average Spectral Distortion and Outliers
107(1)
MSE Weighting Techniques
107(3)
Codebook Structures
110(7)
Split Vector Quantization
111(2)
Multi-Stage Vector Quantization
113(1)
Search strategies for MSVQ
114(2)
MSVQ Codebook Training
116(1)
MSVQ Performance Analysis
117(4)
Codebook Structures
117(1)
Search Techniques
117(2)
Perceptual Weighting Techniques
119(2)
Inter-frame Correlation
121(9)
LSF Prediction
122(2)
Prediction Order
124(1)
Prediction Factor Estimation
125(1)
Performance Evaluation of MA Prediction
126(2)
Joint Quantization of LSFs
128(1)
Use of MA Prediction in Joint Quantization
129(1)
Improved LSF Estimation Through Anti-Aliasing Filtering
130(16)
LSF Extraction
131(4)
Advantages of Low-pass Filtering in Moving Average Prediction
135(11)
Summary
146(3)
Bibliography
146(3)
Pitch Estimation and Voiced-Unvoiced Classification of Speech
149(50)
Introduction
149(1)
Pitch Estimation Methods
150(28)
Time-Domain PDAs
151(4)
Frequency-Domain PDAs
155(3)
Time-and Frequency-Domain PDAs
158(8)
Pre- and Post-processing Techniques
166(12)
Voiced-Unvoiced Classification
178(18)
Hard-Decision Voicing
178(11)
Soft-Decision Voicing
189(7)
Summary
196(3)
Bibliography
197(2)
Analysis by Synthesis LPC Coding
199(62)
Introduction
199(1)
Generalized AbS Coding
200(19)
Time-Varying Filters
202(1)
Perceptually-based Minimization Procedure
203(3)
Excitation Signal
206(2)
Determination of Optimum Excitation Sequence
208(4)
Characteristics of AbS-LPC Schemes
212(7)
Code-Excited Linear Predictive Coding
219(39)
LPC Prediction
221(1)
Pitch Prediction
222(8)
Multi-Pulse Excitation
230(8)
Codebook Excitation
238(14)
Joint LTP and Codebook Excitation Computation
252(3)
CELP with Post-Filtering
255(3)
Summary
258(3)
Bibliography
258(3)
Harmonic Speech Coding
261(16)
Introduction
261(1)
Sinusoidal Analysis and Synthesis
262(1)
Parameter Estimation
263(5)
Voicing Determination
264(2)
Harmonic Amplitude Estimation
266(2)
Common Harmonic Coders
268(7)
Sinusoidal Transform Coding
268(2)
Improved Multi-Band Excitation, INMARSAT-M Version
270(1)
Split-Band Linear Predictive Coding
271(4)
Summary
275(2)
Bibliography
275(2)
Multimode Speech Coding
277(80)
Introduction
277(3)
Design Challenges of a Hybrid Coder
280(1)
Reliable Speech Classification
281(1)
Phase Synchronization
281(1)
Summary of Hybrid Coders
281(4)
Prototype Waveform Interpolation Coder
282(1)
Combined Harmonic and Waveform Coding at Low Bit-Rates
282(1)
A 4 kb/s Hybrid MELP/CELP Coder
283(1)
Limitations of Existing Hybrid Coders
284(1)
Synchronized Waveform-Matched Phase Model
285(13)
Extraction of the Pitch Pulse Location
286(6)
Estimation of the Pitch Pulse Shape
292(5)
Synthesis using Generalized Cubic Phase Interpolation
297(1)
Hybrid Encoder
298(13)
Synchronized Harmonic Excitation
299(2)
Advantages and Disadvantages of SWPM
301(3)
Offset Target Modification
304(4)
Onset Harmonic Memory Initialization
308(1)
White Noise Excitation
309(2)
Speech Classification
311(8)
Open-Loop Initial Classification
312(3)
Closed-Loop Transition Detection
315(3)
Plosive Detection
318(1)
Hybrid Decoder
319(1)
Performance Evaluation
320(2)
Quantization Issues of Hybrid Coder Parameters
322(9)
Introduction
322(1)
Unvoiced Excitation Quantization
323(1)
Harmonic Excitation Quantization
323(8)
Quantization of ACELP Excitation at Transitions
331(1)
Variable Bit Rate Coding
331(5)
Transition Quantization with 4 kb/s ACELP
332(1)
Transition Quantization with 6 kb/s ACELP
332(1)
Transition Quantization with 8 kb/s ACELP
333(1)
Comparison
334(2)
Acoustic Noise and Channel Error Performance
336(14)
Performance under Acoustic Noise
337(8)
Performance under Channel Errors
345(4)
Performance Improvement under Channel Errors
349(1)
Summary
350(7)
Bibliography
351(6)
Voice Activity Detection
357(22)
Introduction
357(3)
Standard VAD Methods
360(8)
ITU-T G.729B/G.723.1A VAD
361(1)
ETSI GSM-FR/HR/EFR VAD
361(1)
ETSI AMR VAD
362(1)
TIA/EIA IS-127/733 VAD
363(1)
Performance Comparison of VADs
364(4)
Likelihood-Ratio-Based VAD
368(7)
Analysis and Improvement of the Likelihood Ratio Method
370(3)
Noise Estimation Based on SLR
373(1)
Comparison
373(2)
Summary
375(4)
Bibliography
375(4)
Speech Enhancement
379(50)
Introduction
379(2)
Review of STSA-based Speech Enhancement
381(21)
Spectral Subtraction
382(2)
Maximum-likelihood Spectral Amplitude Estimation
384(1)
Wiener Filtering
385(1)
MMSE Spectral Amplitude Estimation
386(1)
Spectral Estimation Based on the Uncertainty of Speech Presence
387(2)
Comparisons
389(3)
Discussion
392(10)
Noise Adaptation
402(4)
Hard Decision-based Noise Adaptation
402(1)
Soft Decision-based Noise Adaptation
403(1)
Mixed Decision-based Noise Adaptation
403(1)
Comparisons
404(2)
Echo Cancellation
406(17)
Digital Echo Canceller Set-up
411(2)
Echo Cancellation Formulation
413(2)
Improved Performance Echo Cancellation
415(8)
Summary
423(6)
Bibliography
426(3)
Index 429

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