Speech Enhancement Techniques For Digital Hearing Aids
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Speech Enhancement Techniques For Digital Hearing Aids
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Author : Komal R. Borisagar
language : en
Publisher: Springer
Release Date : 2018-11-15
Speech Enhancement Techniques For Digital Hearing Aids written by Komal R. Borisagar and has been published by Springer this book supported file pdf, txt, epub, kindle and other format this book has been release on 2018-11-15 with Technology & Engineering categories.
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.
Advanced Signal Enhancement Techniques With Application To Speech And Hearing
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Author : Navin Chatlani
language : en
Publisher:
Release Date : 2011
Advanced Signal Enhancement Techniques With Application To Speech And Hearing written by Navin Chatlani and has been published by this book supported file pdf, txt, epub, kindle and other format this book has been release on 2011 with categories.
Advanced signal enhancement techniques with application to speech and hearing are presented that are applied to areas including adaptive noise cancellation (ANC) in noisy speech signals, single channel noise reduction for speech enhancement, voice activity detection (VAD) and noise reduction in binaural hearing aids. The performance enhancement of the new techniques over competing approaches is presented. For the domains of ANC and single channel noise reduction, the use of Empirical Mode Decomposition (EMD) is the underpinning technique employed. A novel approach to dual-channel speech enhancement using Adaptive Empirical Mode Decomposition (SEAEMD) is also presented, when a noise reference is available. The new SEAEMD system incorporates the multi-resolution approach EMD with ANC for effective speech enhancement in stationary and non-stationary noise environments. Two novel Empirical Mode Decomposition based filtering (EMDF) algorithms are presented for single channel speech enhancement. The first system is designed to be particularly effective in low frequency noise environments. The second generalized EMDF system is designed to operate under other noisy conditions, with results presented for babble noise, military vehicle noise and car interior noise. It is shown that the proposed EMDF techniques enhance the speech more effectively than current speech enhancement approaches that use effective noise estimation routines. Speech systems such as hearing aids require fast and computationally inexpensive signal processing technologies. A new and computationally efficient 1-dimensional local binary pattern (1-D LBP) signal processing procedure is designed and applied to (i) signal segmentation and (ii) the VAD problem. Both applications use the underlying features extracted from the 1-D LBP. The simplicity and low computational complexity of 1-D LBP processing are demonstrated. A novel binaural noise reduction system is presented for steering the focus direction of a hearing aid (2 microphones per hearing aid) to additional directions as well as 0/180 degrees. The system places a spatial null in the direction of the target speaker to obtain a noise estimate. The noisy speech signal is then filtered to perform noise reduction, and thus focus on the target speaker located at the desired direction. The results demonstrate its performance at attenuating multiple directional interferers.
Digital Speech Transmission
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Author : Peter Vary
language : en
Publisher: John Wiley & Sons
Release Date : 2006-08-04
Digital Speech Transmission written by Peter Vary and has been published by John Wiley & Sons this book supported file pdf, txt, epub, kindle and other format this book has been release on 2006-08-04 with Science categories.
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.
Audio Source Separation And Speech Enhancement
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Author : Emmanuel Vincent
language : en
Publisher: John Wiley & Sons
Release Date : 2018-07-24
Audio Source Separation And Speech Enhancement written by Emmanuel Vincent and has been published by John Wiley & Sons this book supported file pdf, txt, epub, kindle and other format this book has been release on 2018-07-24 with Technology & Engineering categories.
Learn the technology behind hearing aids, Siri, and Echo Audio source separation and speech enhancement aim to extract one or more source signals of interest from an audio recording involving several sound sources. These technologies are among the most studied in audio signal processing today and bear a critical role in the success of hearing aids, hands-free phones, voice command and other noise-robust audio analysis systems, and music post-production software. Research on this topic has followed three convergent paths, starting with sensor array processing, computational auditory scene analysis, and machine learning based approaches such as independent component analysis, respectively. This book is the first one to provide a comprehensive overview by presenting the common foundations and the differences between these techniques in a unified setting. Key features: Consolidated perspective on audio source separation and speech enhancement. Both historical perspective and latest advances in the field, e.g. deep neural networks. Diverse disciplines: array processing, machine learning, and statistical signal processing. Covers the most important techniques for both single-channel and multichannel processing. This book provides both introductory and advanced material suitable for people with basic knowledge of signal processing and machine learning. Thanks to its comprehensiveness, it will help students select a promising research track, researchers leverage the acquired cross-domain knowledge to design improved techniques, and engineers and developers choose the right technology for their target application scenario. It will also be useful for practitioners from other fields (e.g., acoustics, multimedia, phonetics, and musicology) willing to exploit audio source separation or speech enhancement as pre-processing tools for their own needs.
Advances In Digital Speech Transmission
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Author : Prof Rainer Martin
language : en
Publisher: John Wiley & Sons
Release Date : 2008-02-28
Advances In Digital Speech Transmission written by Prof Rainer Martin and has been published by John Wiley & Sons this book supported file pdf, txt, epub, kindle and other format this book has been release on 2008-02-28 with Technology & Engineering categories.
Speech processing and speech transmission technology are expanding fields of active research. New challenges arise from the 'anywhere, anytime' paradigm of mobile communications, the ubiquitous use of voice communication systems in noisy environments and the convergence of communication networks toward Internet based transmission protocols, such as Voice over IP. As a consequence, new speech coding, new enhancement and error concealment, and new quality assessment methods are emerging. Advances in Digital Speech Transmission provides an up-to-date overview of the field, including topics such as speech coding in heterogeneous communication networks, wideband coding, and the quality assessment of wideband speech. Provides an insight into the latest developments in speech processing and speech transmission, making it an essential reference to those working in these fields Offers a balanced overview of technology and applications Discusses topics such as speech coding in heterogeneous communications networks, wideband coding, and the quality assessment of the wideband speech Explains speech signal processing in hearing instruments and man-machine interfaces from applications point of view Covers speech coding for Voice over IP, blind source separation, digital hearing aids and speech processing for automatic speech recognition Advances in Digital Speech Transmission serves as an essential link between the basics and the type of technology and applications (prospective) engineers work on in industry labs and academia. The book will also be of interest to advanced students, researchers, and other professionals who need to brush up their knowledge in this field.
Speech Enhancement
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Author : Philipos C. Loizou
language : en
Publisher: CRC Press
Release Date : 2013-02-25
Speech Enhancement written by Philipos C. Loizou and has been published by CRC Press this book supported file pdf, txt, epub, kindle and other format this book has been release on 2013-02-25 with Technology & Engineering categories.
With the proliferation of mobile devices and hearing devices, including hearing aids and cochlear implants, there is a growing and pressing need to design algorithms that can improve speech intelligibility without sacrificing quality. Responding to this need, Speech Enhancement: Theory and Practice, Second Edition introduces readers to the basic problems of speech enhancement and the various algorithms proposed to solve these problems. Updated and expanded, this second edition of the bestselling textbook broadens its scope to include evaluation measures and enhancement algorithms aimed at improving speech intelligibility. Fundamentals, Algorithms, Evaluation, and Future Steps Organized into four parts, the book begins with a review of the fundamentals needed to understand and design better speech enhancement algorithms. The second part describes all the major enhancement algorithms and, because these require an estimate of the noise spectrum, also covers noise estimation algorithms. The third part of the book looks at the measures used to assess the performance, in terms of speech quality and intelligibility, of speech enhancement methods. It also evaluates and compares several of the algorithms. The fourth part presents binary mask algorithms for improving speech intelligibility under ideal conditions. In addition, it suggests steps that can be taken to realize the full potential of these algorithms under realistic conditions. What’s New in This Edition Updates in every chapter A new chapter on objective speech intelligibility measures A new chapter on algorithms for improving speech intelligibility Real-world noise recordings (on accompanying CD) MATLAB® code for the implementation of intelligibility measures (on accompanying CD) MATLAB and C/C++ code for the implementation of algorithms to improve speech intelligibility (on accompanying CD) Valuable Insights from a Pioneer in Speech Enhancement Clear and concise, this book explores how human listeners compensate for acoustic noise in noisy environments. Written by a pioneer in speech enhancement and noise reduction in cochlear implants, it is an essential resource for anyone who wants to implement or incorporate the latest speech enhancement algorithms to improve the quality and intelligibility of speech degraded by noise. Includes a CD with Code and Recordings The accompanying CD provides MATLAB implementations of representative speech enhancement algorithms as well as speech and noise databases for the evaluation of enhancement algorithms.
Digital Speech Transmission And Enhancement
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Author : Peter Vary
language : en
Publisher: John Wiley & Sons
Release Date : 2024-01-23
Digital Speech Transmission And Enhancement written by Peter Vary and has been published by John Wiley & Sons this book supported file pdf, txt, epub, kindle and other format this book has been release on 2024-01-23 with Technology & Engineering categories.
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.
Noise Reduction In Speech Processing
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Author : Jacob Benesty
language : en
Publisher: Springer Science & Business Media
Release Date : 2009-04-28
Noise Reduction In Speech Processing written by Jacob Benesty and has been published by Springer Science & Business Media this book supported file pdf, txt, epub, kindle and other format this book has been release on 2009-04-28 with Technology & Engineering categories.
Noise is everywhere and in most applications that are related to audio and speech, such as human-machine interfaces, hands-free communications, voice over IP (VoIP), hearing aids, teleconferencing/telepresence/telecollaboration systems, and so many others, the signal of interest (usually speech) that is picked up by a microphone is generally contaminated by noise. As a result, the microphone signal has to be cleaned up with digital signal processing tools before it is stored, analyzed, transmitted, or played out. This cleaning process is often called noise reduction and this topic has attracted a considerable amount of research and engineering attention for several decades. One of the objectives of this book is to present in a common framework an overview of the state of the art of noise reduction algorithms in the single-channel (one microphone) case. The focus is on the most useful approaches, i.e., filtering techniques (in different domains) and spectral enhancement methods. The other objective of Noise Reduction in Speech Processing is to derive all these well-known techniques in a rigorous way and prove many fundamental and intuitive results often taken for granted. This book is especially written for graduate students and research engineers who work on noise reduction for speech and audio applications and want to understand the subtle mechanisms behind each approach. Many new and interesting concepts are presented in this text that we hope the readers will find useful and inspiring.
Single Channel Phase Aware Signal Processing In Speech Communication
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Author : Pejman Mowlaee
language : en
Publisher: John Wiley & Sons
Release Date : 2016-12-27
Single Channel Phase Aware Signal Processing In Speech Communication written by Pejman Mowlaee and has been published by John Wiley & Sons this book supported file pdf, txt, epub, kindle and other format this book has been release on 2016-12-27 with Technology & Engineering categories.
An overview on the challenging new topic of phase-aware signal processing Speech communication technology is a key factor in human-machine interaction, digital hearing aids, mobile telephony, and automatic speech/speaker recognition. With the proliferation of these applications, there is a growing requirement for advanced methodologies that can push the limits of the conventional solutions relying on processing the signal magnitude spectrum. Single-Channel Phase-Aware Signal Processing in Speech Communication provides a comprehensive guide to phase signal processing and reviews the history of phase importance in the literature, basic problems in phase processing, fundamentals of phase estimation together with several applications to demonstrate the usefulness of phase processing. Key features: Analysis of recent advances demonstrating the positive impact of phase-based processing in pushing the limits of conventional methods. Offers unique coverage of the historical context, fundamentals of phase processing and provides several examples in speech communication. Provides a detailed review of many references and discusses the existing signal processing techniques required to deal with phase information in different applications involved with speech. The book supplies various examples and MATLAB® implementations delivered within the PhaseLab toolbox. Single-Channel Phase-Aware Signal Processing in Speech Communication is a valuable single-source for students, non-expert DSP engineers, academics and graduate students.
Digital Hearing Aids
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Author : Arthur Schaub
language : en
Publisher: Thieme
Release Date : 2011-01-01
Digital Hearing Aids written by Arthur Schaub and has been published by Thieme this book supported file pdf, txt, epub, kindle and other format this book has been release on 2011-01-01 with Medical categories.
An indispensable guide to digital hearing aid technology Digital Hearing Aids is an essential reference for information about the latest innovations in digital hearing aid technology. Concise descriptions and easy-to-reference tables and diagrams enable the reader to rapidly gain a solid understanding of digital signal processing, including such important topics as adaptive acoustic directionality, adaptive noise reduction, adaptive feedback cancellation, and sound classification. The book is divided into three main sections, with the first section providing an overview of foundational concepts, the second section presenting detailed analysis of state-of-the-art processing techniques, and the third section describing specific technical aspects of digital processing. Highlights: Each chapter opens with a brief overview of topics and questions, rapidly orienting the reader with the scope of the material presented Mathematical examples in the third section of the book allow the reader to work through practical calculations, comprehend the nuts and bolts of the processing schemes, and understand the benefits and limitations of each More than 170 illustrations and diagrams aid the comprehension of key concepts This handbook is ideal for audiologists, otolaryngologists, speech-language pathologists, and for other professionals involved in the applications of digital signal processing.