Digital Speech Transmission and Enhancement

Informationen zum Autor Peter Vary is former Head of the Institute of Communication Systems at RWTH Aachen University, Germany. Professor Vary is a Fellow of IEEE, EURASIP, and ITG, and has been a Distinguished Lecturer of the IEEE Signal Processing Society. Rainer Martin is Head of the Institute of Communication Acoustics at Ruhr-Universität Bochum, Germany. Professor Martin is a Fellow of the IEEE. Both authors have been actively involved in speech processing research and teaching over several decades. Klappentext Enables readers to understand the latest developments in speech enhancement/transmission due to advances in computational power and device miniaturizationThe 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.Besides mobile communications, which constituted the main application domain of the first edition, speech enhancement for hearing instruments and man-machine interfaces has gained significantly more prominence in the past decade, and as such receives greater focus in this updated and expanded 2nd edition.In the Second Edition of Digital Speech Transmission and Enhancement, readers can expect to find information and novel methods on:* Low-latency spectral analysis-synthesis, single-channel and dual-channel algorithms for noise reduction and dereverberation.* Multi-microphone processing methods, which are now widely used in applications such as mobile phones, hearing aids, and man-computer interfaces.* Algorithms for near-end listening enhancement, which provide a significantly increased speech intelligibility for users at the noisy receiving side of their mobile phone.* Fundamentals of speech signal processing, estimation and machine learning, speech coding, error concealment by soft decoding, and artificial bandwidth extension of speech signalsDigital Speech Transmission and Enhancement is a single-source, comprehensive guide to the fundamental issues, algorithms, standards, and trends in speech signal processing and speech communication technology, and as such is an invaluable resource for engineers, researchers, academics, and graduate students in the areas of communications, electrical engineering, and information technology. Zusammenfassung DIGITAL SPEECH TRANSMISSION AND ENHANCEMENTEnables readers to understand the latest developments in speech enhancement/transmission due to advances in computational power and device miniaturizationThe 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.Besides mobile communications, which constituted the main application domain of the first edition, speech enhancement for hearing instruments and man-machine interfaces has gained significantly more prominence in the past decade, and as such receives greater focus in this updated and expanded second edition.Readers can expect to find information and novel methods on:* Low-latency spectral analysis-synthesis, single-channel and dual-channel algorithms for noise reduction and dereverberation* Multi-microphone processing methods, which are now widely used in applications such as mobile phones, hearing aids, and man-computer interfaces* Algorithms for near-end listening enhancement, which provide a significantly increased speech intelligibility for users at the noisy receiving si...

List of contents

Preface xv
 
1 Introduction 1
 
2 Models of Speech Production and Hearing 5
 
2.1 Sound Waves 5
 
2.2 Organs of Speech Production 7
 
2.3 Characteristics of Speech Signals 9
 
2.4 Model of Speech Production 10
 
2.4.1 Acoustic Tube Model of the Vocal Tract 12
 
2.4.2 Discrete Time All-Pole Model of the Vocal Tract 19
 
2.5 Anatomy of Hearing 25
 
2.6 Psychoacoustic Properties of the Auditory System 27
 
2.6.1 Hearing and Loudness 27
 
2.6.2 Spectral Resolution 29
 
2.6.3 Masking 31
 
2.6.4 Spatial Hearing 32
 
2.6.4.1 Head-Related Impulse Responses and Transfer Functions 33
 
2.6.4.2 Law of The First Wavefront 34
 
References 35
 
3 Spectral Transformations 37
 
3.1 Fourier Transform of Continuous Signals 37
 
3.2 Fourier Transform of Discrete Signals 38
 
3.3 Linear Shift Invariant Systems 41
 
3.3.1 Frequency Response of LSI Systems 42
 
3.4 The z-transform 42
 
3.4.1 Relation to Fourier Transform 43
 
3.4.2 Properties of the ROC 44
 
3.4.3 Inverse z-Transform 44
 
3.4.4 z-Transform Analysis of LSI Systems 46
 
3.5 The Discrete Fourier Transform 47
 
3.5.1 Linear and Cyclic Convolution 48
 
3.5.2 The DFT of Windowed Sequences 51
 
3.5.3 Spectral Resolution and Zero Padding 54
 
3.5.4 The Spectrogram 55
 
3.5.5 Fast Computation of the DFT: The FFT 56
 
3.5.6 Radix-2 Decimation-in-Time FFT 57
 
3.6 Fast Convolution 60
 
3.6.1 Fast Convolution of Long Sequences 60
 
3.6.2 Fast Convolution by Overlap-Add 61
 
3.6.3 Fast Convolution by Overlap-Save 61
 
3.7 Analysis-Modification-Synthesis Systems 64
 
3.8 Cepstral Analysis 66
 
3.8.1 Complex Cepstrum 67
 
3.8.2 Real Cepstrum 69
 
3.8.3 Applications of the Cepstrum 70
 
3.8.3.1 Construction of Minimum-Phase Sequences 70
 
3.8.3.2 Deconvolution by Cepstral Mean Subtraction 71
 
3.8.3.3 Computation of the Spectral Distortion Measure 72
 
3.8.3.4 Fundamental Frequency Estimation 73
 
References 75
 
4 Filter Banks for Spectral Analysis and Synthesis 79
 
4.1 Spectral Analysis Using Narrowband Filters 79
 
4.1.1 Short-Term Spectral Analyzer 83
 
4.1.2 Prototype Filter Design for the Analysis Filter Bank 86
 
4.1.3 Short-Term Spectral Synthesizer 87
 
4.1.4 Short-Term Spectral Analysis and Synthesis 88
 
4.1.5 Prototype Filter Design for the Analysis-Synthesis filter bank 90
 
4.1.6 Filter Bank Interpretation of the DFT 92
 
4.2 Polyphase Network Filter Banks 94
 
4.2.1 PPN Analysis Filter Bank 95
 
4.2.2 PPN Synthesis Filter Bank 101
 
4.3 Quadrature Mirror Filter Banks 104
 
4.3.1 Analysis-Synthesis Filter Bank 104
 
4.3.2 Compensation of Aliasing and Signal Reconstruction 106
 
4.3.3 Efficient Implementation 109
 
4.4 Filter Bank Equalizer 112
 
4.4.1 The Reference Filter Bank 112
 
4.4.2 Uniform Frequency Resolution 113
 
4.4.3 Adaptive Filter Bank Equalizer: Gain Computation 117
 
4.4.3.1 Conventional Spectral Subtraction 117
 
4.4.3.2 Filter Bank Equalizer 118
 
4.4.4 Non-uniform Frequency Resolution 120
 
4.4.5 Design Aspects & Implementation 122
 
References 123
 
5 Stochastic Signals and Estimation 127
 
5.1 Basic Concepts 127
 
5.1.1 Random Events and Probability 127
 
5.1.2 Conditional Probabilities 128
 
5.1.3 Random Variables 129
 

Product details

• Authors: Rainer Martin, Martin Rainer, Peter Vary, Peter (Rwth Aachen University Vary, Vary Peter
• Publisher: Wiley, John and Sons Ltd
• Languages: English
• Product format: Hardback
• Released: 18.01.2024
• EAN: 9781119060963
• ISBN: 978-1-119-06096-3
• No. of pages: 592
• Series: Wiley - IEEE; IEEE Press
• Subjects: Natural sciences, medicine, IT, technology > Technology > Electronics, electrical engineering, communications engineering; Sprachverarbeitung, Signalverarbeitung, drahtlose kommunikation, Signal Processing, Electrical & Electronics Engineering, Elektrotechnik u. Elektronik, Audio-, Sprachverarbeitung u. Übertragung, Audio & Speech Processing & Broadcasting