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Digital speech transmission : enhancement coding and error consealment
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Haboken, NJ : John Wiley & Sons, 2006
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30000010114545 TK7882.S65 V37 2006 Open Access Book Book
30000010114544 TK7882.S65 V37 2006 Open Access Book Book
30000004616003 TK7882.S65 V37 2006 Open Access Book Book
30000010128750 TK7882.S65 V37 2006 Open Access Book Book
30000010114546 TK7882.S65 V37 2006 Open Access Book Book

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The enormous advances in digital signal processing (DSP) technology have contributed to the wide dissemination and success of speech communication devices - be it GSM and UMTS mobile telephones, digital hearing aids, or human-machine interfaces. Digital speech transmission techniques play an important role in these applications, all the more because high quality speech transmission remains essential in all current and next generation communication networks.

Enhancement, coding and error concealment techniques improve the transmitted speech signal at all stages of the transmission chain, from the acoustic front-end to the sound reproduction at the receiver. Advanced speech processing algorithms help to mitigate a number of physical and technological limitations such as background noise, bandwidth restrictions, shortage of radio frequencies, and transmission errors.

Digital Speech Transmission provides a single-source, comprehensive guide to the fundamental issues, algorithms, standards, and trends in speech signal processing and speech communication technology. The authors give a solid, accessible overview of
* fundamentals of speech signal processing
* speech coding, including new speech coders for GSM and UMTS
* error concealment by soft decoding
* artificial bandwidth extension of speech signals
* single and multi-channel noise reduction
* acoustic echo cancellation

This text is an invaluable resource for engineers, researchers, academics, and graduate students in the areas of communications, electrical engineering, and information technology.

Author Notes

Peter Vary is the author of Digital Speech Transmission: Enhancement, Coding and Error Concealment, published by Wiley.

Rainer Martin is the author of Digital Speech Transmission: Enhancement, Coding and Error Concealment, published by Wiley.

Table of Contents

1 Introduction
2 Models of Speech Production and Hearing
2.1 Organs of Speech Production
2.2 Characteristics of Speech Signals
2.3 Model of Speech Production
2.4 Anatomy of Hearing
2.5 Performance of the Auditory Organs
3 Spectral Transformations
3.1 Fourier Transform of Continuous Signals
3.2 Fourier Transform of Discrete Signals
3.3 Linear Shift Invariant Systems
3.4 Thez-Transform
3.5 The Discrete Fourier Transform
3.6 Fast Convolution
3.7 Cepstral Analysis
4 Filter Banks for Spectral Analysis and Synthesis
4.1 Spectral Analysis Using Narrow-Band Filters
4.2 Polyphase Network Filter Banks
4.3 QuadratureMirror Filter Banks
5 Stochastic Signals and Estimation
5.1 Basic Concepts
5.2 Expectations andMoments
5.3 Bivariate Statistics
5.4 Probability and Information
5.5 Multivariate Statistics
5.6 Stochastic Processes
5.7 Estimation of Statistical Quantities by Time Averages
5.8 Power Spectral Densities
5.9 Estimation of the Power Spectral Density
5.10 Statistical Properties of Speech Signals
5.11 Statistical Properties of DFT Coe.cients
5.12 Optimal Estimation
6 Linear Prediction
6.1 Vocal TractModels and Short-TermPrediction
6.2 Optimal Prediction Coe.cients for Stationary Signals
6.3 Predictor Adaptation
6.4 Long-TermPrediction
7 Quantization
7.1 Analog Samples and Digital Presentation
7.2 Uniform Quantization
7.3 Non-uniformQuantization
7.4 OptimalQuantization
7.5 Adaptive Quantization
7.6 Vector Quantization
7.6.1 Principle
8 Speech Coding
8.1 Classi.cation of Speech Coding Algorithms
8.2 Model-Based Predictive Coding
8.3 Di.erentialWaveform Coding
8.4 Parametric Coding
8.5 Hybrid Coding
8.6 Adaptive Post.ltering
9 Error Concealment and Softbit Decoding
9.1 Hardbit Source Decoding
9.2 Conventional Error Concealment
9.3 Softbits and L-Values
9.4 Softbit Source Decoding (SD)
9.5 Application toModel Parameters
9.6 Further Improvements
10 Bandwidth Extension of Speech Signals (BWE)
10.1 Narrowband versusWideband Telephony
10.2 Speech Coding with Integrated BWE
10.3 BWE without Auxiliary Transmission
11 Single and Dual Channel Noise Reduction
11.1 Introduction
11.2 LinearMMSE Estimators
11.3 Speech Enhancement in the DFT Domain
11.4 Optimal Non-Linear Estimators
11.5 Joint Optimum Detection and Estimation of Speech
11.6 Computation of Likelihood Ratios
11.7 Estimation of theA Priory Probability of Speech Presence
11.8 VAD and Noise Estimation Techniques
11.9 Dual-Channel Noise Reduction
12 Multi-Channel Noise Reduction
12.1 Introduction
12.2 Spatial Sampling of Sound Fields
12.3 Beamforming
12.4 PerformanceMeasures and Spatial Aliasing
12.5 Design of Fixed Beamformers
12.6 Adaptive Beamformers
13 Acoustic Echo Control
13.1 The Echo Control Problem
13.2 Evaluation Criteria
13.3 TheWiener Solution
13.4 The LMS and NLMS Algorithm
13.5 Convergence Analysis and Control of the LMS Algorithm
13.6 Geometric Projection Interpretation of the NLMS Algorithm
13.7 The Projection Algorithm
13.8 Least-Squares and Recursive Least-Squares Algorithms
13.9 Block Processing and Frequency-Domain Adaptive Filters
13.9.1 Block LMS Algorithm
13.10 Additional Measures for Echo Control
13.11 Stereophonic Acoustic Echo Control
A Codec Standards
B Speech Quality Assessment
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