This book focuses on a class of single-channel noise reduction methods that are performed in the frequency domain via the short-time Fourier transform (STFT). The simplicity and relative effectiveness of this class of approaches make them the dominant choice in practical systems. Even though many popular algorithms have been proposed through more than four decades of continuous research, there are a number of critical areas where our understanding and capabilities still remain quite rudimentary, especially with respect to the relationship between noise reduction and speech distortion. All existing frequency-domain algorithms, no matter how they are developed, have one feature in common: the solution is eventually expressed as a gain function applied to the STFT of the noisy signal only in the current frame. As a result, the narrowband signal-to-noise ratio (SNR) cannot be improved, and any gains achieved in noise reduction on the fullband basis come with a price to pay, which is speech distortion. In this book, we present a new perspective on the problem by exploiting the difference between speech and typical noise in circularity and interframe self-correlation, which were ignored in the past. By gathering the STFT of the microphone signal of the current frame, its complex conjugate, and the STFTs in the previous frames, we construct several new, multiple-observation signal models similar to a microphone array system: there are multiple noisy speech observations, and their speech components are correlated but not completely coherent while their noise components are presumably uncorrelated. Therefore, the multichannel Wiener filter and the minimum variance distortionless response (MVDR) filter that were usually associated with microphone arrays will be developed for single-channel noise reduction in this book. This might instigate a paradigm shift geared toward speech distortionless noise reduction techniques. Table of Contents: Introduction / Problem Formulation / Performance Measures / Linear and Widely Linear Models / Optimal Filters with Model 1 / Optimal Filters with Model 2 / Optimal Filters with Model 3 / Optimal Filters with Model 4 / Experimental Study

This classic reference on musical acoustics and performance practice begins with a brief introduction to the fundamentals of acoustics and the generation of musical sounds. It then discusses the particulars of the sounds made by all the standard instruments in a modern orchestra as well as the human voice, the way in which the sounds made by these instruments are dispersed and how the room into which they are projected affects the sounds.

Senior level/graduate level text/reference presenting state-of-the- art numerical techniques to solve the wave equation in heterogeneous fluid-solid media. Numerical models have become standard research tools in acoustic laboratories, and thus computational acoustics is becoming an increasingly important branch of ocean acoustic science. The first edition of this successful book, written by the recognized leaders of the field, was the first to present a comprehensive and modern introduction to computational ocean acoustics accessible to students. This revision, with 100 additional pages, completely updates the material in the first edition and includes new models based on current research. It includes problems and solutions in every chapter, making the book more useful in teaching (the first edition had a separate solutions manual). The book is intended for graduate and advanced undergraduate students of acoustics, geology and geophysics, applied mathematics, ocean engineering or as a reference in computational methods courses, as well as professionals in these fields, particularly those working in government (especially Navy) and industry labs engaged in the development or use of propagating models.

This book basically involves the study of flight parameters, wing beat frequency, moment of inertia, and wing movements for developing various aerodynamic forces which have been calculated. The book is intended for biologists, physicists, nanotechnologists, and aerospace engineers. Resilin, an elastic polymer (4 ) which is present at the base of insect, plays a major role in Neurogenic and Myogenic insect flyers and influences the physiology of flight muscles. Leading edge vortex (LEV) is a special feature of insect flight. Insect wings have stalling angle above 60 degrees as compared to a man-made aeroplane stalling angle which is 16 degree. Reynolds number, the knowledge of LEV, and detailed study of moment of inertia help in developing flapping flexible wings for micro-aerial-vehicles. This book serves as an interface between biologists and engineers interested to develop biomimicking micro-aerial-vehicles. The contents of this book is useful to researchers and professionals alike.

Musical Sound, Instruments, and Equipment offers a basic understanding of sound, musical instruments and music equipment, geared towards a general audience and non-science majors. The book begins with an introduction of the fundamental properties of sound waves, and the perception of the characteristics of sound. The relation between intensity and loudness, and the relation between frequency and pitch are discussed. The basics of propagation of sound waves, and the interaction of sound waves with objects and structures of various sizes are introduced. Standing waves, harmonics and resonance are explained in simple terms, using graphics that provide a visual understanding.

This book provides a complete guide on tools and techniques for modeling of supercritical fluid extraction (SFE) as well as sub critical fluid extraction (SCFE) processes and phenomena. It provides details for both SFE and SCFE from managing the experiments to modeling and optimization. It includes the fundamentals of SFE as well as the necessary experimental techniques to validate the models. The optimization section includes the use of process simulators, conventional optimization techniques and state-of-the-art genetic algorithm methods. Numerous practical examples and case studies on the application of the modeling and optimization techniques on the SFE processes are also provided. Detailed thermo dynamical modeling with and without co-solvent and non equilibrium system modeling is another feature of the book. The book consists of seven chapters. Chapter one provides an overview of the field of SFE/SCFE and their importance and relation to food, cosmetic and pharmaceutical industries. Chapter two covers SFE/SCFE fundamentals and presents process descriptions. Chapter 3 discusses thermodynamic modeling of SFE including thermodynamic modeling in the presence of co solvent and also non-equilibrium state modeling. Chapter 4 provides details on the general modeling as well as optimization tools. Chapter 5 covers the general modeling techniques illustrated in chapter 4 and applies it for supercritical and sub critical modeling. Chapter 6 applies the optimization tools (traditional/genetic algorithm) for supercritical subcritical modeling. Finally, chapter 7 describes the procedure on how to design and mange experiments in SFE.

This book delivers a comprehensive and up-to-date treatment of practical applications of metamaterials, structured media, and conventional porous materials. With increasing levels of urbanization, a growing demand for motorized transport, and inefficient urban planning, environmental noise exposure is rapidly becoming a pressing societal and health concern. Phononic and sonic crystals, acoustic metamaterials, and metasurfaces can revolutionize noise and vibration control and, in many cases, replace traditional porous materials for these applications. In this collection of contributed chapters, a group of international researchers reviews the essentials of acoustic wave propagation in metamaterials and porous absorbers with viscothermal losses, as well as the most recent advances in the design of acoustic metamaterial absorbers. The book features a detailed theoretical introduction describing commonly used modelling techniques such as plane wave expansion, multiple scattering theory, and the transfer matrix method. The following chapters give a detailed consideration of acoustic wave propagation in viscothermal fluids and porous media, and the extension of this theory to non-local models for fluid saturated metamaterials, along with a description of the relevant numerical methods. Finally, the book reviews a range of practical industrial applications, making it especially attractive as a white book targeted at the building, automotive, and aeronautic industries.

Periodic signals can be decomposed into sets of sinusoids having frequencies that are integer multiples of a fundamental frequency. The problem of finding such fundamental frequencies from noisy observations is important in many speech and audio applications, where it is commonly referred to as pitch estimation. These applications include analysis, compression, separation, enhancement, automatic transcription and many more. In this book, an introduction to pitch estimation is given and a number of statistical methods for pitch estimation are presented. The basic signal models and associated estimation theoretical bounds are introduced, and the properties of speech and audio signals are discussed and illustrated. The presented methods include both single- and multi-pitch estimators based on statistical approaches, like maximum likelihood and maximum a posteriori methods, filtering methods based on both static and optimal adaptive designs, and subspace methods based on the principles of subspace orthogonality and shift-invariance. The application of these methods to analysis of speech and audio signals is demonstrated using both real and synthetic signals, and their performance is assessed under various conditions and their properties discussed. Finally, the estimators are compared in terms of computational and statistical efficiency, generalizability and robustness. Table of Contents: Fundamentals / Statistical Methods / Filtering Methods / Subspace Methods / Amplitude Estimation

In this book, we introduce the background and mainstream methods of probabilistic modeling and discriminative parameter optimization for speech recognition. The specific models treated in depth include the widely used exponential-family distributions and the hidden Markov model. A detailed study is presented on unifying the common objective functions for discriminative learning in speech recognition, namely maximum mutual information (MMI), minimum classification error, and minimum phone/word error. The unification is presented, with rigorous mathematical analysis, in a common rational-function form. This common form enables the use of the growth transformation (or extended Baum-Welch) optimization framework in discriminative learning of model parameters. In addition to all the necessary introduction of the background and tutorial material on the subject, we also included technical details on the derivation of the parameter optimization formulas for exponential-family distributions, discrete hidden Markov models (HMMs), and continuous-density HMMs in discriminative learning. Selected experimental results obtained by the authors in firsthand are presented to show that discriminative learning can lead to superior speech recognition performance over conventional parameter learning. Details on major algorithmic implementation issues with practical significance are provided to enable the practitioners to directly reproduce the theory in the earlier part of the book into engineering practice. Table of Contents: Introduction and Background / Statistical Speech Recognition: A Tutorial / Discriminative Learning: A Unified Objective Function / Discriminative Learning Algorithm for Exponential-Family Distributions / Discriminative Learning Algorithm for Hidden Markov Model / Practical Implementation of Discriminative Learning / Selected Experimental Results / Epilogue / Major Symbols Used in the Book and Their Descriptions / Mathematical Notation / Bibliography

Immediately following the Second World War, between 1947 and 1955, several classic papers quantified the fundamentals of human speech information processing and recognition. In 1947 French and Steinberg published their classic study on the articulation index. In 1948 Claude Shannon published his famous work on the theory of information. In 1950 Fletcher and Galt published their theory of the articulation index, a theory that Fletcher had worked on for 30 years, which integrated his classic works on loudness and speech perception with models of speech intelligibility. In 1951 George Miller then wrote the first book Language and Communication, analyzing human speech communication with Claude Shannon's just published theory of information. Finally in 1955 George Miller published the first extensive analysis of phone decoding, in the form of confusion matrices, as a function of the speech-to-noise ratio. This work extended the Bell Labs' speech articulation studies with ideas from Shannon's Information theory. Both Miller and Fletcher showed that speech, as a code, is incredibly robust to mangling distortions of filtering and noise. Regrettably much of this early work was forgotten. While the key science of information theory blossomed, other than the work of George Miller, it was rarely applied to aural speech research. The robustness of speech, which is the most amazing thing about the speech code, has rarely been studied. It is my belief (i.e., assumption) that we can analyze speech intelligibility with the scientific method. The quantitative analysis of speech intelligibility requires both science and art. The scientific component requires an error analysis of spoken communication, which depends critically on the use of statistics, information theory, and psychophysical methods. The artistic component depends on knowing how to restrict the problem in such a way that progress may be made. It is critical to tease out the relevant from the irrelevant and dig for the key issues. This will focus us on the decoding of nonsense phonemes with no visual component, which have been mangled by filtering and noise. This monograph is a summary and theory of human speech recognition. It builds on and integrates the work of Fletcher, Miller, and Shannon. The long-term goal is to develop a quantitative theory for predicting the recognition of speech sounds. In Chapter 2 the theory is developed for maximum entropy (MaxEnt) speech sounds, also called nonsense speech. In Chapter 3, context is factored in. The book is largely reflective, and quantitative, with a secondary goal of providing an historical context, along with the many deep insights found in these early works.

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

Latent semantic mapping (LSM) is a generalization of latent semantic analysis (LSA), a paradigm originally developed to capture hidden word patterns in a text document corpus. In information retrieval, LSA enables retrieval on the basis of conceptual content, instead of merely matching words between queries and documents. It operates under the assumption that there is some latent semantic structure in the data, which is partially obscured by the randomness of word choice with respect to retrieval. Algebraic and/or statistical techniques are brought to bear to estimate this structure and get rid of the obscuring ""noise."" This results in a parsimonious continuous parameter description of words and documents, which then replaces the original parameterization in indexing and retrieval. This approach exhibits three main characteristics: -Discrete entities (words and documents) are mapped onto a continuous vector space; -This mapping is determined by global correlation patterns; and -Dimensionality reduction is an integral part of the process. Such fairly generic properties are advantageous in a variety of different contexts, which motivates a broader interpretation of the underlying paradigm. The outcome (LSM) is a data-driven framework for modeling meaningful global relationships implicit in large volumes of (not necessarily textual) data. This monograph gives a general overview of the framework, and underscores the multifaceted benefits it can bring to a number of problems in natural language understanding and spoken language processing. It concludes with a discussion of the inherent tradeoffs associated with the approach, and some perspectives on its general applicability to data-driven information extraction. Contents: I. Principles / Introduction / Latent Semantic Mapping / LSM Feature Space / Computational Effort / Probabilistic Extensions / II. Applications / Junk E-mail Filtering / Semantic Classification / Language Modeling / Pronunciation Modeling / Speaker Verification / TTS Unit Selection / III. Perspectives / Discussion / Conclusion / Bibliography

This fully updated, self-contained textbook covering modern optical microscopy equips students with a solid understanding of the theory underlying a range of advanced techniques. Two new chapters cover pump-probe techniques, and imaging in scattering media, and additional material throughout covers light-sheet microscopy, image scanning microscopy, and much more. An array of practical techniques are discussed, from classical phase contrast and confocal microscopy, to holographic, structured illumination, multi-photon, and coherent Raman microscopy, and optical coherence tomography. Fundamental topics are also covered, including Fourier optics, partial coherence, 3D imaging theory, statistical optics, and the physics of scattering and fluorescence. With a wealth of end-of-chapter problems, and a solutions manual for instructors available online, this is an invaluable book for electrical engineering, biomedical engineering, and physics students taking graduate courses on optical microscopy, as well as advanced undergraduates, professionals, and researchers looking for an accessible introduction to the field.

Acoustic Justice engages issues of recognition and misrecognition by mobilizing an acoustic framework. From the vibrational intensities of common life to the rhythm of bodies in movement, and drawing from his ongoing work on sound and agency, Brandon LaBelle positions acoustics, and the broader experience of listening, as a dynamic means for fostering responsiveness, understanding, dispute, and the work of reorientation. As such, acoustic justice emerges as a compelling platform for engaging struggles over the right to speak and to be heard that extends toward a broader materialist and planetary view. This entails critically addressing questions of space, borders, community, and the acoustic norms defining capacities of listening, leading to what LaBelle terms “poetic ecologies of resonance.” Acoustic Justice works at issues of recognition and resistance, place and displacement, by moving across a range of pertinent references and topics, from social practices and sound art to the performativity of skin and the poetics of Deaf voice. Through such transversality, LaBelle captures acoustics as the basis for strategies of refusal and repair.

The acoustics of a space can have a real impact on the sounds you create and capture. Acoustics and Psychoacoustics, Fifth Edition provides supportive tools and exercises to help you understand how music sounds and behaves in different spaces, whether during a performance or a recording, when planning a control room or listening space, and how it is perceived by performers, listeners, and recording engineers.

With their clear and simple style, Howard and Angus cover both theory and practice by addressing the science of sound engineering and music production, the acoustics of musical instruments, the ways in which we hear musical sounds, the underlying principles of sound processing, and the application of these concepts to music spaces to create professional sound. This new edition is fully revised to reflect new psychoacoustic information related to timbre and temporal perception, including an updated discussion of vocal fold vibration principles, samples of recent acoustic treatments, and a description of variable acoustics in spaces, as well as coverage of the environment’s effect on production listening, sonification, and other topics.

Devoted to the teaching of musical understanding, an accompanying website (www.routledge.com/cw/howard) features various audio clips, tutorial sheets, questions and answers, and trainings that will take your perception of sound to the next level.

This book will help you:

Gain a basic grounding in acoustics and psychoacoustics with respect to music audio technology systems

Incorporate knowledge of psychoacoustics in future music technology system designs as appropriate

Understand how we hear pitch, loudness, and timbre

Learn to influence the acoustics of an enclosed space through designed physical modifications

Table of Contents

Preface

Chapter 1: Introduction to Sound

Chapter 2: Introduction to Hearing

Chapter 3: Notes and Harmony

Chapter 4: Acoustic Model for Musical Instruments

Chapter 5: Hearing Timbre and Deceiving the Ear

Chapter 6: Hearing Music in Different Environments

Chapter 7: Applications: Acoustics and Psychoacoustics Combined

Appendix 1: The Fourier Transform

Appendix 2: Solving the ERB Equation

Appendix 3: Converting between Frequency Ratios and Cents

Appendix 4: Deriving the Reverberation Time Equation

Appendix 5: Deriving the Reverberation Time Equation for Different Frequencies and Surfaces

Appendix 6: The Effect of Speaker Sixe on its Polar Pattern

Appendix 7: Track Listing for the Audio Compact Disc

Index

Auralization is the technique of creation and reproduction of sound on the basis of computer data. With this tool it is possible to predict the character of sound signals which are generated at the source and modified by reinforcement, propagation and transmission in systems such as rooms, buildings, vehicles or other technical devices. This book is organized as a comprehensive collection of the basics of sound and vibration, acoustic modelling, simulation, signal processing and audio reproduction. With some mathematical prerequisites, the readers will be able to follow the main strategy of auralization easily and work out their own implementations of auralization in various fields of application in architectural acoustics, acoustic engineering, sound design and virtual reality. For readers interested in basic research, the technique of auralization may be useful to create sound stimuli for specific investigations in linguistic, medical, neurological and psychological research, and in the field of human-machine interaction.

This undergraduate textbook aids readers in studying music and color, which involve nearly the entire gamut of the fundamental laws of classical as well as atomic physics. The objective bases for these two subjects are, respectively, sound and light. Their corresponding underlying physical principles overlap greatly: Both music and color are manifestations of wave phenomena. As a result, commonalities exist as to the production, transmission, and detection of sound and light. Whereas traditional introductory physics textbooks are styled so that the basic principles are introduced first and are then applied, this book is based on a motivational approach: It introduces a subject with a set of related phenomena, challenging readers by calling for a physical basis for what is observed. A novel topic in the first edition and this second edition is a non-mathematical study of electric and magnetic fields and how they provide the basis for the propagation of electromagnetic waves, of light in particular. The book provides details for the calculation of color coordinates and luminosity from the spectral intensity of a beam of light as well as the relationship between these coordinates and the color coordinates of a color monitor. The second edition contains corrections to the first edition, the addition of more than ten new topics, new color figures, as well as more than forty new sample problems and end-of-chapter problems. The most notable additional topics are: the identification of two distinct spectral intensities and how they are related, beats in the sound from a Tibetan bell, AM and FM radio, the spectrogram, the short-time Fourier transform and its relation to the perception of a changing pitch, a detailed analysis of the transmittance of polarized light by a Polaroid sheet, brightness and luminosity, and the mysterious behavior of the photon. The Physics of Music and Color is written at a level suitable for college students without any scientific background, requiring only simple algebra and a passing familiarity with trigonometry. The numerous problems at the end of each chapter help the reader to fully grasp the subject.