Digital Speech Transmission and Enhancement 2nd Edition by Peter Vary, Rainer Martin 1119060982 9781119060987

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Product details:

ISBN-10 :  1119060982 

ISBN-13 :  9781119060987

Author : Peter Vary, Rainer Martin 

DIGITAL SPEECH TRANSMISSION AND ENHANCEMENT
Enables readers to understand the latest developments in speech enhancement/transmission due to advances in computational power and device miniaturization

The Second Edition of Digital Speech Transmission and Enhancement has been updated throughout to provide all the necessary details on the latest advances in the theory and practice in speech signal processing and its applications, including many new research results, standards, algorithms, and developments which have recently appeared and are on their way into state-of-the-art applications.

 

Digital Speech Transmission and Enhancement 2nd Table of contents:

1 Introduction
2 Models of Speech Production and Hearing
2.1 Sound Waves
2.2 Organs of Speech Production
2.3 Characteristics of Speech Signals
2.4 Model of Speech Production
2.5 Anatomy of Hearing
2.6 Psychoacoustic Properties of the Auditory System
References
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 The z‐transform
3.5 The Discrete Fourier Transform
3.6 Fast Convolution
3.7 Analysis–Modification–Synthesis Systems
3.8 Cepstral Analysis
References
Notes
4 Filter Banks for Spectral Analysis and Synthesis
4.1 Spectral Analysis Using Narrowband Filters
4.2 Polyphase Network Filter Banks
4.3 Quadrature Mirror Filter Banks
4.4 Filter Bank Equalizer
References
Notes
5 Stochastic Signals and Estimation
5.1 Basic Concepts
5.2 Expectations and Moments
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 Spectrum and its Estimation
5.9 Statistical Properties of Speech Signals
5.10 Statistical Properties of DFT Coefficients
5.11 Optimal Estimation
5.12 Non‐Linear Estimation with Deep Neural Networks
References
Notes
6 Linear Prediction
6.1 Vocal Tract Models and Short‐Term Prediction
6.2 Optimal Prediction Coefficients for Stationary Signals
6.3 Predictor Adaptation
6.4 Long‐Term Prediction
References
7 Quantization
7.1 Analog Samples and Digital Representation
7.2 Uniform Quantization
7.3 Non‐uniform Quantization
7.4 Optimal Quantization
7.5 Adaptive Quantization
7.6 Vector Quantization
7.7 Quantization of the Predictor Coefficients
References
Note
8 Speech Coding
8.1 Speech‐Coding Categories
8.2 Model‐Based Predictive Coding
8.3 Linear Predictive Waveform Coding
8.4 Parametric Coding
8.5 Hybrid Coding
8.6 Adaptive Postfiltering
8.7 Speech Codec Standards: Selected Examples
References
Note
9 Concealment of Erroneous or Lost Frames
9.1 Concepts for Error Concealment
9.2 Examples of Error Concealment Standards
9.3 Further Improvements
References
10 Bandwidth Extension of Speech Signals
10.1 BWE Concepts
10.2 BWE using the Model of Speech Production
10.3 Speech Codecs with Integrated BWE
References
Note
11 NELE: Near‐End Listening Enhancement
11.1 Frequency Domain NELE (FD)
11.2 Time Domain NELE (TD)
References
Notes
12 Single‐Channel Noise Reduction
12.1 Introduction
12.2 Linear MMSE Estimators
12.3 Speech Enhancement in the DFT Domain
12.4 Optimal Non‐linear Estimators
12.5 Joint Optimum Detection and Estimation of Speech
12.6 Computation of Likelihood Ratios
12.7 Estimation of the A Priori and A Posteriori Probabilities of Speech Presence
12.8 VAD and Noise Estimation Techniques
12.9 Noise Reduction with Deep Neural Networks
References
13 Dual‐Channel Noise and Reverberation Reduction
13.1 Dual‐Channel Wiener Filter
13.2 The Ideal Diffuse Sound Field and Its Coherence
13.3 Noise Cancellation
13.4 Noise Reduction
13.5 Dual‐Channel Dereverberation
13.6 Methods Based on Deep Learning
References
14 Acoustic Echo Control
14.1 The Echo Control Problem
14.2 Echo Cancellation and Postprocessing
14.3 Evaluation Criteria
14.4 The Wiener Solution
14.5 The LMS and NLMS Algorithms
14.6 Convergence Analysis and Control of the LMS Algorithm
14.7 Geometric Projection Interpretation of the NLMS Algorithm
14.8 The Affine Projection Algorithm
14.9 Least‐Squares and Recursive Least‐Squares Algorithms
14.10 Block Processing and Frequency Domain Adaptive Filters
14.11 Stereophonic Acoustic Echo Control
References
15 Microphone Arrays and Beamforming
15.1 Introduction
15.2 Spatial Sampling of Sound Fields
15.3 Beamforming
15.4 Performance Measures and Spatial Aliasing
15.5 Design of Fixed Beamformers
15.6 Multichannel Wiener Filter and Postfilter
15.7 Adaptive Beamformers
15.8 Non‐linear Multi‐channel Noise Reduction

 

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