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...

Auteur

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.

Texte du rabat

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. 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 signals Digital 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.

Contenu

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 Convolu…