Speech Enhancement Techniques for Digital Hearing Aids (eBook)

Komal R. Borisagar, Rohit M. Thanki, Bhavin S. Sedani (Autoren)

eBook Download: PDF

2018 | 1st ed. 2019
IX, 155 Seiten
Springer International Publishing (Verlag)
978-3-319-96821-6 (ISBN)

Infos

?This book provides various speech enhancement algorithms for digital hearing aids. It covers information on noise signals extracted from silences of speech signal. The description of the algorithm used for this purpose is also provided. Different types of adaptive filters such as Least Mean Squares (LMS), Normalized LMS (NLMS) and Recursive Lease Squares (RLS) are described for noise reduction in the speech signals. Different types of noises are taken to generate noisy speech signals, and therefore information on various noises signals is provided. The comparative performance of various adaptive filters for noise reduction in speech signals is also described. In addition, the book provides a speech enhancement technique using adaptive filtering and necessary frequency strength enhancement using wavelet transform as per the requirement of audiogram for digital hearing aids.

• Presents speech enhancement techniques for improving performance of digital hearing aids;
  
• Covers various types of adaptive filters and their advantages and limitations;
  
• Provides a hybrid speech enhancement technique using wavelet transform and adaptive filters.

Dr. Komal R. Borisagar has obtained her Ph.D. in Speech Enhancement Techniques for Digital Hearing Aids in 2012. She has teaching experience of over 15 years. She is working as an Associate Professor at Electronics & Communication Department, Atmiya Institute of Technology and Science, Rajkot. She has published 2 books, 3 book chapters and more than 50 research papers to her credit in referred & indexed journals, conferences at international and in IEEE digital library. She has achieved best paper award five times for her research articles and presentation. Her areas of interest are wireless communication, speech processing, signal & system and image processing.

 Dr. Rohit M. Thanki has obtained his Ph.D. in Multibiometric System Security using CS Theory and Watermarking from C. U. Shah University, Wadhwan city, Gujarat, India in 2017. His area of research is Digital Watermarking, Biometrics System, Security, Compressive Sensing, Pattern Recognition and Image Processing. He has published 7 books, 9 book chapters and more than 25 research papers to his credit in refereed & indexed journals, and conferences at international and national level. He has achieved best paper presentation awards 2 times for his research articles. His international recognition includes his professional memberships & services in refereed organizations, programme committees and reviewer for journals published by IEEE, Elsevier, Taylor & Francis, Springer, IGI-Global etc.

Dr. Bhavin S. Sedani is working as a professor in Electronics and Communication Department at L.D. College of Engineering, Ahmedabad. He has teaching experience of 16 years. He has presented   more than 46 research papers in various international and national conferences. His 21 research papers are published in various international journals and IEEE Xplore. He has achieved best paper presentation awards 7 times for his research articles. He is awarded with pedagogical award for continuing efforts towards teaching learning methodology, research and innovation in 2017 by Gujarat Technological University, Young Researcher Award in 2018 by Integrated Intelligent Research, ISTE Professional Center, Anna University Campus, Chennai - India. His area of research is in wireless communication and speech processing.

Inhalt

Preface 5
Contents 7
Chapter 1: Introduction 10
1.1 Sound 10
1.2 Ear Structure and Its Workings 10
1.2.1 External Ear 11
1.2.2 Middle Ear 11
1.2.3 Inner Ear 11
1.2.4 Cochlea 12
1.3 Hearing Impaired 12
1.4 Audiogram 13
1.5 Digital Hearing Aids 13
1.6 Issues in Digital Hearing Aids 16
1.7 Motivation for This Book 16
1.7.1 Important Areas of Speech Signal Covered in This Book 18
1.7.1.1 Voice Activity Detection in the Noisy Speech Signal 18
1.7.1.2 Adaptive Filtering for Speech Signal Processing 19
1.8 Book Organization 20
References 20
Chapter 2: Generation of Speech Signal and Its Characteristics 21
2.1 Speech Signal 21
2.1.1 Articulatory Phonetics and Speech Generation 21
2.1.2 Anatomy and Physiology of Speech Generation 22
2.1.3 Vocal Tract 22
2.1.4 Larynx and Vocal Folds or Cords 24
2.2 Major Features of Speech Articulation 26
2.3 Properties and Characteristics of Speech Signal 28
2.3.1 Time and Frequency Domain Characteristics of Speech 29
2.3.2 Waveforms 29
2.3.3 Fundamental Frequency 29
2.3.4 Overall Power 30
2.3.5 Overall Frequency Spectrum 30
2.3.6 Short-Time Energy 31
2.3.7 Spectrogram 31
2.3.8 Short-Time Average Zero Crossing Rate 32
References 35
Chapter 3: Introduction of Adaptive Filters and Noises for Speech 36
3.1 Adaptive Filter 36
3.2 LMS Adaptive Filter 37
3.2.1 Least Mean Square Adaptation Algorithm 40
3.2.2 Statistical LMS Theory 42
3.2.3 Direct Averaging Method 43
3.2.4 Small Step Size Statistical Theory 44
3.2.5 Natural Modes of the LMS Filter 45
3.2.6 Learning Curves for Adaptive Algorithms 46
3.2.7 Comparison of the LMS Algorithm with the Steepest Descent Algorithm 47
3.3 Normalized Least Mean Square (NLMS) Adaptive Filter 48
3.3.1 Structure and Operation of NLMS 49
3.3.2 Stability of the Normalized LMS Filter 52
3.3.3 Special Environment of Real Valued Data 53
3.4 Recursive Least Squares (RLS) Adaptive Filter 55
3.4.1 Regularization 56
3.4.2 Reformulation of the Normal Equations 57
3.4.3 Recursive Computations of (n) and z(n) 57
3.4.4 The Matrix Inversion Lemma 58
3.4.5 Selection of the Regularization Parameter 60
3.4.6 Convergence Analysis of RLS Algorithm 61
3.4.7 Convergence of the RLS Algorithm in the Mean Value 62
3.4.8 Mean Square Deviation of the RLS Algorithm 63
3.4.9 Ensemble Average Learning Curve of the RLS Algorithm 64
3.5 Noise 64
3.5.1 Sources of Noise 66
3.5.1.1 Different Road Traffic Noises 66
3.5.1.2 Babble Noise 66
References 68
Chapter 4: Fourier Transform, Short-Time Fourier Transform, and Wavelet Transform 70
4.1 Fourier Transform (FT) 70
4.2 Short-Time FT 70
4.3 Wavelet Transform (WT) 71
4.4 Comparison of the Wavelet Transform (WT) with FT and STFT 74
4.5 Multiresolution Algorithm 78
References 81
Chapter 5: Speech Signal Enhancement Using Adaptive Filters 82
5.1 Introduction 82
5.2 Steps for Speech Enhancement Process 83
5.3 Implementation Flow of VAD Algorithm 83
5.4 Speech Enhancement Process based on LMS Algorithm 84
5.4.1 Results for White Noise Signal 88
5.4.2 Results for Babble Noise Signal 93
5.4.3 Results for Traffic Jam Noise Signal 98
5.5 Speech Enhancement Process Based on the NLMS Algorithm 102
5.5.1 Results for White Noise Signal 105
5.5.2 Results for Babble Noise Signal 109
5.5.3 Results for Traffic Jam Noise Signal 113
5.6 Speech Enhancement Process Based on the RLS Algorithm 116
5.6.1 Results for White Noise Signal 117
5.6.2 Results for Babble Noise Signal 122
5.6.3 Results for Traffic Jam Noise Signal 126
5.7 Comparative Analysis of Simulation Results 130
References 131
Chapter 6: Speech Signal Enhancement Based on Wavelet Transform 132
6.1 Procedure for Speech Signal Enhancement Using Wavelet Transform 132
6.2 Implementation and Results of Speech Signal Enhancement Using Wavelet Transform 137
6.2.1 First Band Enhancement 138
6.2.2 Second Band Enhancement 140
6.2.3 Third Band Enhancement 141
6.2.4 Fourth Band Enhancement 144
6.2.5 Fifth Band Enhancement 145
6.2.6 Sixth Band Enhancement 147
6.2.7 Seventh Band Enhancement 149
6.2.8 Eighth Band Enhancement 150
6.2.9 Ninth Band Enhancement 152
References 155
Chapter 7: Summary of This Book and Future Research Directions 156
7.1 Important Points Covered in the Book 156
7.2 Future Research Direction 157
Index 158

Produkt-Details

• Erscheint lt. Verlag: 15.11.2018
• Zusatzinfo: IX, 155 p. 139 illus., 103 illus. in color.
• Verlagsort: Cham
• Sprache: englisch
• Themenwelt: Mathematik / Informatik ► Informatik ► Grafik / Design
• Themenwelt: Informatik ► Netzwerke ► Sicherheit / Firewall
• Themenwelt: Technik ► Bauwesen
• Themenwelt: Technik ► Elektrotechnik / Energietechnik
• Schlagworte: Adaptive Filters and Noises • Digital Hearing Aids • Speech Enhancement for Digital Hearing Aids • Speech Signal Enhancement using Adaptive Filters • Speech Signal Enhancement Wavelet Transforms • speech signals • wavelet transforms
• ISBN-10: 3-319-96821-1 / 3319968211
• ISBN-13: 978-3-319-96821-6 / 9783319968216