Mathematical Modeling And Signal Processing In Speech And Hearing Sciences
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Mathematical Modeling and Signal Processing in Speech and Hearing Sciences by Jack Xin,Yingyong Qi Pdf
The aim of the book is to give an accessible introduction of mathematical models and signal processing methods in speech and hearing sciences for senior undergraduate and beginning graduate students with basic knowledge of linear algebra, differential equations, numerical analysis, and probability. Speech and hearing sciences are fundamental to numerous technological advances of the digital world in the past decade, from music compression in MP3 to digital hearing aids, from network based voice enabled services to speech interaction with mobile phones. Mathematics and computation are intimately related to these leaps and bounds. On the other hand, speech and hearing are strongly interdisciplinary areas where dissimilar scientific and engineering publications and approaches often coexist and make it difficult for newcomers to enter.
Multiscale Modeling of Pedestrian Dynamics by Emiliano Cristiani,Benedetto Piccoli,Andrea Tosin Pdf
This book presents mathematical models and numerical simulations of crowd dynamics. The core topic is the development of a new multiscale paradigm, which bridges the microscopic and macroscopic scales taking the most from each of them for capturing the relevant clues of complexity of crowds. The background idea is indeed that most of the complex trends exhibited by crowds are due to an intrinsic interplay between individual and collective behaviors. The modeling approach promoted in this book pursues actively this intuition and profits from it for designing general mathematical structures susceptible of application also in fields different from the inspiring original one. The book considers also the two most traditional points of view: the microscopic one, in which pedestrians are tracked individually and the macroscopic one, in which pedestrians are assimilated to a continuum. Selected existing models are critically analyzed. The work is addressed to researchers and graduate students.
The Mimetic Finite Difference Method for Elliptic Problems by Lourenco Beirao da Veiga,Konstantin Lipnikov,Gianmarco Manzini Pdf
This book describes the theoretical and computational aspects of the mimetic finite difference method for a wide class of multidimensional elliptic problems, which includes diffusion, advection-diffusion, Stokes, elasticity, magnetostatics and plate bending problems. The modern mimetic discretization technology developed in part by the Authors allows one to solve these equations on unstructured polygonal, polyhedral and generalized polyhedral meshes. The book provides a practical guide for those scientists and engineers that are interested in the computational properties of the mimetic finite difference method such as the accuracy, stability, robustness, and efficiency. Many examples are provided to help the reader to understand and implement this method. This monograph also provides the essential background material and describes basic mathematical tools required to develop further the mimetic discretization technology and to extend it to various applications.
Signals and Systems for Speech and Hearing by Stuart Rosen,Peter Howell Pdf
This book introduces speech and hearing sciences students to the principles of "signal" and "system" analysis. Beginning with an examination of what signals and systems are, the book develops a thorough background from which many of the most important issues in speech and hearing can be tackled. Numerous illustrations.
Speech dynamics refer to the temporal characteristics in all stages of the human speech communication process. This speech “chain” starts with the formation of a linguistic message in a speaker's brain and ends with the arrival of the message in a listener's brain. Given the intricacy of the dynamic speech process and its fundamental importance in human communication, this monograph is intended to provide a comprehensive material on mathematical models of speech dynamics and to address the following issues: How do we make sense of the complex speech process in terms of its functional role of speech communication? How do we quantify the special role of speech timing? How do the dynamics relate to the variability of speech that has often been said to seriously hamper automatic speech recognition? How do we put the dynamic process of speech into a quantitative form to enable detailed analyses? And finally, how can we incorporate the knowledge of speech dynamics into computerized speech analysis and recognition algorithms? The answers to all these questions require building and applying computational models for the dynamic speech process. What are the compelling reasons for carrying out dynamic speech modeling? We provide the answer in two related aspects. First, scientific inquiry into the human speech code has been relentlessly pursued for several decades. As an essential carrier of human intelligence and knowledge, speech is the most natural form of human communication. Embedded in the speech code are linguistic (as well as para-linguistic) messages, which are conveyed through four levels of the speech chain. Underlying the robust encoding and transmission of the linguistic messages are the speech dynamics at all the four levels. Mathematical modeling of speech dynamics provides an effective tool in the scientific methods of studying the speech chain. Such scientific studies help understand why humans speak as they do and how humans exploit redundancy and variability by way of multitiered dynamic processes to enhance the efficiency and effectiveness of human speech communication. Second, advancement of human language technology, especially that in automatic recognition of natural-style human speech is also expected to benefit from comprehensive computational modeling of speech dynamics. The limitations of current speech recognition technology are serious and are well known. A commonly acknowledged and frequently discussed weakness of the statistical model underlying current speech recognition technology is the lack of adequate dynamic modeling schemes to provide correlation structure across the temporal speech observation sequence. Unfortunately, due to a variety of reasons, the majority of current research activities in this area favor only incremental modifications and improvements to the existing HMM-based state-of-the-art. For example, while the dynamic and correlation modeling is known to be an important topic, most of the systems nevertheless employ only an ultra-weak form of speech dynamics; e.g., differential or delta parameters. Strong-form dynamic speech modeling, which is the focus of this monograph, may serve as an ultimate solution to this problem. After the introduction chapter, the main body of this monograph consists of four chapters. They cover various aspects of theory, algorithms, and applications of dynamic speech models, and provide a comprehensive survey of the research work in this area spanning over past 20~years. This monograph is intended as advanced materials of speech and signal processing for graudate-level teaching, for professionals and engineering practioners, as well as for seasoned researchers and engineers specialized in speech processing
Modeling Sensorineural Hearing Loss by Walt Jesteadt Pdf
A recent study indicates that 20 million people in the United States have significant sensorineural hearing loss. Approximately 95% of those people have partial losses, with varying degrees of residual hearing. These percentages are similar in other developed countries. What changes in the function of the cochlea or inner ear cause such losses? What does the world sound like to the 19 million people with residual hearing? How should we transform sounds to correct for the hearing loss and maximize restoration of normal hearing? Answers to such questions require detailed models of the way that sounds are processed by the nervous system, both for listeners with normal hearing and for those with sensorineural hearing loss. This book contains chapters describing the work of 25 different research groups. A great deal of research in recent years has been aimed at obtaining a better physiological description of the altered processes that cause sensorineural hearing loss and a better understanding of transformations that occur in the perception of those sounds that are sufficiently intense that they can still be heard. Efforts to understand these changes in function have lead to a better understanding of normal function as well. This research has been based on rigorous mathematical models, computer simulations of mechanical and physiological processes, and signal processing simulations of the altered perceptual experience of listeners with sensorineural hearing loss. This book provides examples of all these approaches to modeling sensorineural hearing loss and a summary of the latest research in the field.
Mathematical Foundations of Speech and Language Processing by Mark Johnson,Sanjeev P. Khudanpur,Mari Ostendorf,Roni Rosenfeld Pdf
Speech and language technologies continue to grow in importance as they are used to create natural and efficient interfaces between people and machines, and to automatically transcribe, extract, analyze, and route information from high-volume streams of spoken and written information. The workshops on Mathematical Foundations of Speech Processing and Natural Language Modeling were held in the Fall of 2000 at the University of Minnesota's NSF-sponsored Institute for Mathematics and Its Applications, as part of a "Mathematics in Multimedia" year-long program. Each workshop brought together researchers in the respective technologies on the one hand, and mathematicians and statisticians on the other hand, for an intensive week of cross-fertilization. There is a long history of benefit from introducing mathematical techniques and ideas to speech and language technologies. Examples include the source-channel paradigm, hidden Markov models, decision trees, exponential models and formal languages theory. It is likely that new mathematical techniques, or novel applications of existing techniques, will once again prove pivotal for moving the field forward. This volume consists of original contributions presented by participants during the two workshops. Topics include language modeling, prosody, acoustic-phonetic modeling, and statistical methodology.
This book intends to provide highlights of the current research in signal processing area and to offer a snapshot of the recent advances in this field. This work is mainly destined to researchers in the signal processing related areas but it is also accessible to anyone with a scientific background desiring to have an up-to-date overview of this domain. The twenty-five chapters present methodological advances and recent applications of signal processing algorithms in various domains as telecommunications, array processing, biology, cryptography, image and speech processing. The methodologies illustrated in this book, such as sparse signal recovery, are hot topics in the signal processing community at this moment. The editor would like to thank all the authors for their excellent contributions in different areas of signal processing and hopes that this book will be of valuable help to the readers.
Mathematical Models for Speech Technology by Stephen Levinson Pdf
Mathematical Models of Spoken Language presents the motivations for, intuitions behind, and basic mathematical models of natural spoken language communication. A comprehensive overview is given of all aspects of the problem from the physics of speech production through the hierarchy of linguistic structure and ending with some observations on language and mind. The author comprehensively explores the argument that these modern technologies are actually the most extensive compilations of linguistic knowledge available.Throughout the book, the emphasis is on placing all the material in a mathematically coherent and computationally tractable framework that captures linguistic structure. It presents material that appears nowhere else and gives a unification of formalisms and perspectives used by linguists and engineers. Its unique features include a coherent nomenclature that emphasizes the deep connections amongst the diverse mathematical models and explores the methods by means of which they capture linguistic structure. This contrasts with some of the superficial similarities described in the existing literature; the historical background and origins of the theories and models; the connections to related disciplines, e.g. artificial intelligence, automata theory and information theory; an elucidation of the current debates and their intellectual origins; many important little-known results and some original proofs of fundamental results, e.g. a geometric interpretation of parameter estimation techniques for stochastic models and finally the author's own unique perspectives on the future of this discipline. There is a vast literature on Speech Recognition and Synthesis however, this book is unlike any other in the field. Although it appears to be a rapidly advancing field, the fundamentals have not changed in decades. Most of the results are presented in journals from which it is difficult to integrate and evaluate all of these recent ideas. Some of the fundamentals have been collected into textbooks, which give detailed descriptions of the techniques but no motivation or perspective. The linguistic texts are mostly descriptive and pictorial, lacking the mathematical and computational aspects. This book strikes a useful balance by covering a wide range of ideas in a common framework. It provides all the basic algorithms and computational techniques and an analysis and perspective, which allows one to intelligently read the latest literature and understand state-of-the-art techniques as they evolve.
Applications of Digital Signal Processing to Audio and Acoustics by Mark Kahrs,Karlheinz Brandenburg Pdf
Karlheinz Brandenburg and Mark Kahrs With the advent of multimedia, digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing. Today, the main appli cations of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including percep tual measurement techniques and analysis/synthesis methods. Smaller but nonetheless very important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application oriented research. The topics covered here coincide with the topics covered in the biannual work shop on “Applications of Signal Processing to Audio and Acoustics”. This event is sponsored by the IEEE Signal Processing Society (Technical Committee on Audio and Electroacoustics) and takes place at Mohonk Mountain House in New Paltz, New York. A short overview of each chapter will illustrate the wide variety of technical material presented in the chapters of this book. John Beerends: Perceptual Measurement Techniques. The advent of perceptual measurement techniques is a byproduct of the advent of digital coding for both speech and high quality audio signals. Traditional measurement schemes are bad estimates for the subjective quality after digital coding/decoding. Listening tests are subject to sta tistical uncertainties and the basic question of repeatability in a different environment.
Speech Enhancement by Jacob Benesty,Jesper Rindom Jensen,Mads Graesboll Christensen,Jingdong Chen Pdf
Speech enhancement is a classical problem in signal processing, yet still largely unsolved. Two of the conventional approaches for solving this problem are linear filtering, like the classical Wiener filter, and subspace methods. These approaches have traditionally been treated as different classes of methods and have been introduced in somewhat different contexts. Linear filtering methods originate in stochastic processes, while subspace methods have largely been based on developments in numerical linear algebra and matrix approximation theory. This book bridges the gap between these two classes of methods by showing how the ideas behind subspace methods can be incorporated into traditional linear filtering. In the context of subspace methods, the enhancement problem can then be seen as a classical linear filter design problem. This means that various solutions can more easily be compared and their performance bounded and assessed in terms of noise reduction and speech distortion. The book shows how various filter designs can be obtained in this framework, including the maximum SNR, Wiener, LCMV, and MVDR filters, and how these can be applied in various contexts, like in single-channel and multichannel speech enhancement, and in both the time and frequency domains. First short book treating subspace approaches in a unified way for time and frequency domains, single-channel, multichannel, as well as binaural, speech enhancement Bridges the gap between optimal filtering methods and subspace approaches Includes original presentation of subspace methods from different perspectives
Acoustics, Information, and Communication by Ning Xiang,Gerhard M. Sessler Pdf
This book explores the life and scientific legacy of Manfred Schroeder through personal reflections, scientific essays and Schroeder’s own memoirs. Reflecting the wide range of Schroeder’s activities, the first part of the book contains thirteen articles written by his colleagues and former students. Topics discussed include his early, pioneering contributions to the understanding of statistical room acoustics and to the measurement of reverberation time; his introduction of digital signal processing methods into acoustics; his use of ray tracing methods to study sound decay in rooms and his achievements in echo and feedback suppression and in noise reduction. Other chapters cover his seminal research in speech processing including the use of predictive coding to reduce audio bandwidth which led to various code-excited linear prediction schemes, today used extensively for speech coding. Several chapters discuss Schroeder’s work in low-peak factor signals, number theory, and maximum-length sequences with key applications in hearing research, diffraction gratings, artificial reverberators and de-correlation techniques for enhancing subjective envelopment in surround sound. In style, the articles range from truly scientific to conversationally personal. In all contributions, the relationship between the current research presented and Manfred Schroeder’s own fields of interest is, in general, evident. The second part of the book consists of Schroeder’s own memoirs, written over the final decade of his life. These recollections shed light on many aspects not only of Schroeder’s life but also on that of many of his colleagues, friends and contemporaries. They portray political, social and scientific events over a period that extends from pre-war to the present. These memoirs, written in an inimitable and witty style, are full of information, entertaining and fun to read, providing key insight into the life and work of one of the greatest acousticians of the 20th century.
An Interactive Multimedia Introduction to Signal Processing by Ulrich Karrenberg Pdf
This introduction to elementary signal processing connects theory and application, and bridges instruction between a book and a CD-ROM packed with video, software and more. The result is a unique, non-mathematical learning system using concepts drawn from modern brain research. Readers use the popular DasyLab metrology and control engineering program to develop applications. Processing of real signals is enabled via the sound card and the parallel port. Two hundred pre-programmed signal engineering systems and design transparencies are provided on the CD-ROM. There are numerous videos, more than 250 photos, and - most important – all "living" experiments and their results are visualized.