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Die Zukunft der Musikarchivierung und der Suchmaschinen liegt im Deep Learning und in Big Data. Algorithmen zum Abrufen von Musikinformationen analysieren automatisch musikalische Merkmale wie Klangfarbe, Melodie, Rhythmus oder musikalische Form, und künstliche Intelligenz sortiert und verknüpft diese Merkmale dann. Auf dem ersten International Symposium on Computational Ethnomusicological Archiving, das vom 9. bis 11. November 2017 am Institut für Systematische Musikwissenschaft in Hamburg stattfand, wurde ein neuer Standard für Computational Phonogram Archiving als interdisziplinärer Ansatz diskutiert. Ethnomusikologen, Musik- und Computerwissenschaftler, Systematische Musikwissenschaftler sowie Musikarchivare, Komponisten und Musiker stellten Werkzeuge, Methoden und Plattformen vor und tauschten Erfahrungen aus der Feldforschung und Archivierung in den Bereichen musikalische Akustik, Informatik, Musiktheorie sowie Musikspeicherung, -wiedergabe und Metadaten aus. Der Standard Computational Phonogram Archiving ist auch auf dem Musikmarkt als Suchmaschine für Musikkonsumenten sehr gefragt. Dieses Buch bietet einen umfassenden Überblick über das Gebiet, geschrieben von führenden Forschern aus aller Welt.

This textbook on Underwater Acoustics has a structure that is more organic than logical. It thereby unifies diverse areas of research, including topics of signal processing, the sonar equation, sources and receivers, scattering and reverberation, wave propagation, propagation models, and inverse problems. It also provides code fragments written in Python which complement the discussion. This is a book written for both beginners and specialists, as well as for biologists, oceanographers, computer engineers, physicists, and mathematicians, and for civilian and naval personnel who are looking for a introductory overview of the topic.

This book highlights the advantages of the vector-phase method in underwater acoustic measurements and presents results of theoretical and experimental studies of the deep open ocean and shallow sea based on vector-phase representations. Based on the physical phenomena discovered and compensation of counter streams of energy and vortices of the acoustic intensity vector, processes of transmitting acoustic energy of a tonal signal in the real ocean are described. The book also discusses the development of advanced detection tools based on vector-phase sonar. This book provides useful content for professionals and researchers working in various fields of applied underwater acoustics.

This book features an extensive index and all Mathcad worksheets. Vinyl is back, tubes/valves are back, on the high-end field SMD-free analog amplification surpasses digitalized chains, and top microphone manufacturers still set on good old op-amps or on fully discrete BJT, FET, and/or tube-driven amplifiers. There is only one problem that is not satisfyingly well solved by the manufacturers: It is the noise production of the active components and the useful reflection in simulation tools, in tables or graphs of the datasheets/data books. Nowadays, mostly surrounded by many digital helping tools, it makes sense using them—also by analog aficionados. It saves cost and time simulating first before spending money. Presented in this book the software tool LTSpice which is the free software solution from Linear Technology (today Analog Devices) that could also be used by full analog lovers to simulate the noise production of their amplifier design. All we need is the right creation approach to develop simulation models for the active components. Inter alia this is already done for tubes and BJTs in the 2nd editions of my “How to Gain Gain” and “Balanced Phono-Amps” books. For op-amps, the missing approaches are presented in the book on hand. It cannot be denied that mathematical software like Mathcad is extremely helpful to find the right equations for graphically presented noise curves which we can find in the literature. Nevertheless, it also works well with other types of math software to fulfill the parameter needs of the here presented modeling approaches for the input referred voltage and current noise of—not only—excellent sounding vintage op-amps, applicable in the audio range from 1 Hz to 100 kHz.

This book compiles and presents a synopsis on current global research efforts to push forward the state of the art in dialogue technologies, including advances to language and context understanding, and dialogue management, as well as human–robot interaction, conversational agents, question answering and lifelong learning for dialogue systems.

This book first introduced the theoretical foundation of nonlinear acoustics such as the basic equations of nonlinear acoustics followed by a statistical mechanics approach to nonlinear acoustics, then a curvilinear spacetime approach to nonlinear acoustics, then a gauge invariance approach to nonlinear acoustics, and application of chaos theory to nonlinear acoustics. Various formats of nonlinear acoustical imaging are given such as B/A nonlinear parameter acoustical imaging, fractal imaging, harmonics imaging, nonclassical nonlinear acoustical imaging, and modulation method in nonlinear acoustical imaging with their applications.

This book highlights time reversal acoustics, techniques based on the symmetry properties of acoustic fields. It has the unique feature that the first eleven chapters of the book are on the indepth studies of the theories of time reversal acoustics. The remaining chapters are on the four major applications of time reversal acoustics, together with their experimental setups and case studies: underwater communication, seismic exploration,nondestructive evaluation, and medical ultrasound imaging.. The gauge invariance approach to acoustic fields, proposed by the author in 2007, is confirmed by the successful fabrication of acoustical metamaterials and the applications of time reversal acoustics to superresolution. The book also presents groundbreaking applications of time reversal acoustics to underwater communication technology and the application of metamaterials to time reversal acoustics.

This book discusses the applications of signal and image processing in acoustical imaging. It first describes the basic tools involved – the 2D transform, fast Fourier transform (FFT) and applications, and deconvolution – before introducing readers to higher-order statistics, wavelets, and neural networks. It also addresses the important topic of digital signal processing, focusing on the example of homomorphic signal processing. The book then details the design of digital filters and array signal processing, and lastly examines applications in image processing: image enhancement and optimization, image restoration, and image compression.

In diesem Band der Reihe Fachwissen Technische Akustik wird das Verfahren der experimentellen Modalanalyse vorgestellt. Mit diesem Verfahren können die von der Ausbreitung von Luft- und Körperschall bestimmten dynamischen Eigenschaften von Systemen untersucht werden. Beispiele für solche Systeme sind Strukturen im Maschinen- und Fahrzeugbau oder auch kleinere Innenräume, deren akustischen Verhalten von Interesse ist. In einer Einführung wird zunächst auf den Zusammenhang des physikalischen Modells und des systemtheoretischen Modells eingegangen sowie der Nutzen des modalen Modells für die Beschreibung der Systemeigenschaften erläutert. Danach wird die dem modalen Modell zugrunde liegende Theorie sowie der Zusammenhang der modalen Parameter mit den im Systemmodell verwendeten Frequenzgängen dargestellt. Verschiedene Verfahren der experimentellen Modalanalyse werden diskutiert, darunter sowohl solche zur getrennten Bestimmung einzelner modaler Parameter als auch solche, bei denen eine Vielzahl modaler Parameter gleichzeitig aus den gemessenen Frequenzgängen ermittelt wird. Zusätzlich wird auf das praktische Vorgehen bei der Gewinnung der dazu notwendigen Messdaten und die Möglichkeiten zur Überprüfung der Ergebnisse eingegangen. Zur Demonstration der verschiedenen Möglichkeiten und Verfahren wird ein einfaches praktisches Beispiel ausführlich behandelt. Das umfasst die Vorgehensweise bei der Messung ebenso wie die Anwendung unterschiedlich aufwändiger Verfahren zur Extraktion der modalen Parameter. Dazu werden zahlreiche Ergebnisse gezeigt, so dass Möglichkeiten und Grenzen der experimentellen Modalanalyse deutlich werden.

This textbook presents the fundamentals of audio coding, used to compress audio and music signals, using Python programs both as examples to illustrate the principles and for experiments for the reader. Together, these programs then form complete audio coders. The author starts with basic knowledge of digital signal processing (sampling, filtering) to give a thorough introduction to filter banks as used in audio coding, and their design methods. He then continues with the next core component, which are psycho-acoustic models. The author finally shows how to design and implement them. Lastly, the author goes on to describe components for more specialized coders, like the Integer-to-Integer MDCT filter bank, and predictive coding for lossless and low delay coding. Included are Python program examples for each section, which illustrate the principles and provide the tools for experiments.Comprehensively explains the fundamentals of filter banks and audio coding;Provides Python examples for each principle so that completed audio coders are obtained in the language;Includes a suite of classroom materials including exercises, experiments, and examples.

This book presents the proceedings of the 46th National Symposium on Acoustics (NSA 2017). The main goal of this symposium is to discuss key opportunities and challenges in acoustics, especially as applied to engineering problems. The book covers topics ranging from hydro-acoustics, environmental acoustics, bio-acoustics to musical acoustics, electro-acoustics and sound perception. The contents of this volume will prove useful to researchers and practicing engineers working on acoustics problems.

This extensively reworked 2nd edition of the book includes ten new chapters. It also features an updated discussion of simulation software tools, covering topics such as simulating complex and / or expensive amplifier structures with the free LTspice software by developing a broad range of additional simulation models, especially those for triodes and transformers.The book adopts the structure used in The Sound of Silence books, with the first part, Basics - Calculations and Simulations, providing deep simulation-triggered insights into the gain and noise mechanisms of differential amplifiers, BJTs, resistors, and triodes. The second part then discusses the RIAA Phono-Amp Engine II, describing all the necessary design, simulation, calculation, construction and measurement processes for this multi-functional MC amplifier.The third part, Knowledge Transfer, presents new ideas on draft designs of the linear low-noise MC input stages (also an extremely low-noise one) and a range of practical measurement tools. Additionally, it includes a chapter on MM amplifiers and their noise production, and offers some surprising solutions. The brand new and extensive chapter on all the simulation models developed and used in the book rounds-out the voyage through the jungle of compromises, allowing best-in-class balanced MC phono-amplifiers to be produced.Lastly, the book also features an extensive index, and free downloads of all Mathcad worksheets are available on Springer's Extra Materials website (extra.springer.com).

This textbook presents the fundamentals of engineering acoustics and examines in depth concepts within the domain that apply to reducing noise, measuring noise, and designing microphones and loudspeakers. The book particularly emphasizes the physical principles used in designing miniature microphones. These devices are used in billions of electronic products, most visibly, cell phones and hearing aids, and enable countless other applications. Distinct from earlier books on this topic that take the view of the electrical engineer analyzing mechanical systems using electric circuit analogies. This text uses Newtonian mechanics as a more appropriate paradigm for analyzing these mechanical systems and in so doing provides a more direct method of modeling. Written at a level appropriate for upper-division undergraduate courses, and enhanced with end-of-chapter problems and MatLab routines, the book is ideal as a core text for students interested in engineering acoustics in ME, EE, and physics programs, as well as a reference for engineers and technicians working in the huge global industry of miniature microphone design.

Luftschall überstreicht nicht nur den Bereich menschlicher Sprachkommunikation, sondern belegt als Infraschall und Ultraschall Bereiche, die auch intensiv technisch oder medizinisch genutzt werden. Diese Nutzung setzt die Kenntnis seiner Eigenschaften voraus, die mit dem zeitlichen und räumlichenVerlauf der Druckänderungen im Schallfeld beschrieben werden können. Für die Wandlung der Schallparameter in registrierbare Werte stehen verschiedene Möglichkeiten zur Verfügung. Die Umsetzung in elektrische Spannungen mit Mikrofonen steht dabei weit an der Spitze. Je nach erforderlicher Genauigkeit werden hierbei Konstruktionen auf unterschiedlicher Basis verwendet. Dieser Band befasst sich mit den Eigenschaften und Möglichkeiten der verschiedenen Lösungen, den damit verbundenen Grenzen der Schallgrößenerfassung sowie einem Ausblick auf zukünftige Entwicklungen.

Für akustische Planungen ist es unerlässlich zu wissen, wie viel Schall eine Quelle abstrahlt. Auch die Wahl einer geräuscharmen Maschine ist nur möglich, wenn die jeweiligen Herstellerangaben miteinander vergleichbar sind. Die Schallleistung ist in beiden Fällen die passende Größe, da sie die Schallabstrahlung einer Quelle unabhängig vom umgebenden Raum beschreibt. Gerade diese Ortsunabhängigkeit ist aber erforderlich, um mehrere Quellen zuverlässig vergleichen zu können oder um z.B. die an einem Arbeitsplatz in einer Werkhalle oder an einem Immissionsort in der Nachbarschaft zu erwartenden Geräuschimmissionen prognostizieren zu können.In dem Band'Schallleistung'werden die theoretischen Grundlagen erläutert und verschiedene, grundlegende Messmethoden vorgestellt. Damit wird die Leserin oder der Leser in die Lage versetzt, das für eine spezifische Messaufgabe passende Messverfahren auszuwählen und, falls in den einschlägigen Regelwerken keine Vorgaben für besondere Einzelfälle zu finden sind, angepasste Lösungen zu erarbeiten, die mit den jeweiligen Messprinzipien in Einklang stehen.

This textbook provides a detailed introduction to the use of software in combination with simple and economical hardware (a sound level meter with calibrated AC output and a digital recording system) to obtain sophisticated measurements usually requiring expensive equipment. It emphasizes the use of free, open source, and multiplatform software. Many commercial acoustical measurement systems use software algorithms as an integral component; however the methods are not disclosed. This book enables the reader to develop useful algorithms and provides insight into the use of digital audio editing tools to document features in the signal. Topics covered include acoustical measurement principles, in-depth critical study of uncertainty applied to acoustical measurements, digital signal processing from the basics, and metrologically-oriented spectral and statistical analysis of signals. The student will gain a deep understanding of the use of software for measurement purposes; the ability to implement software-based measurement systems; familiarity with the hardware necessary to acquire and store signals; an appreciation for the key issue of long-term preservation of signals; and a full grasp of the often neglected issue of uncertainty in acoustical measurements. Pedagogical features include in-text worked-out examples, end-of-chapter problems, a glossary of metrology terms, and extensive appendices covering statistics, proofs, additional examples, file formats, and underlying theory.

This improved and updated second edition covers the theory, development, and design of electro-acoustic transducers for underwater applications. This highly regarded text discusses the basics of piezoelectric and magnetostrictive transducers that are currently being used as well as promising new designs. It presents the basic acoustics as well as the specific acoustics data needed in transducer design and evaluation. A broad range of designs of projectors and hydrophones are described in detail along with methods of modeling, evaluation, and measurement. Analysis of projector and hydrophone transducer arrays, including the effects of mutual radiation impedance and numerical models for elements and arrays, are also covered. The book includes new advances in transducer design and transducer materials and has been completely reorganized to be suitable for use as a textbook, as well as a reference or handbook. The new edition contains corrections to the first edition, end-of-chapter exercises, and solutions to selected exercises. Each chapter includes a short introduction, end-of-chapter summary, and an extensive reference list offering the reader more detailed information and historical context. A glossary of key terms is also included at the end.

This book provides comprehensive, graduate-level treatment of analog and digital signal analysis suitable for course use and self-guided learning. This expert text guides the reader from the basics of signal theory through a range of application tools for use in acoustic analysis, geophysics, and data compression. Each concept is introduced and explained step by step, and the necessary mathematical formulae are integrated in an accessible and intuitive way. The first part of the book explores how analog systems and signals form the basics of signal analysis. This section covers Fourier series and integral transforms of analog signals, Laplace and Hilbert transforms, the main analog filter classes, and signal modulations. Part II covers digital signals, demonstrating their key advantages. It presents z and Fourier transforms, digital filtering, inverse filters, deconvolution, and parametric modeling for deterministic signals. Wavelet decomposition and reconstruction of non-stationary signals are also discussed. The third part of the book is devoted to random signals, including spectral estimation, parametric modeling, and Tikhonov regularization. It covers statistics of one and two random variables and the principles and methods of spectral analysis. Estimation of signal properties is discussed in the context of ergodicity conditions and parameter estimations, including the use of Wiener and Kalman filters. Two appendices cover the basics of integration in the complex plane and linear algebra. A third appendix presents a basic Matlab toolkit for computer signal analysis. This expert text provides both a solid theoretical understanding and tools for real-world applications.

By adaptive signal processing, we mean, in general, adaptive?ltering.In- known environments where we need to model, identify, or track time-varying channels, adaptive?ltering has been proven to be an e?ective and powerful tool. As a result, this tool is now in use in many di?erent?elds. Since the invention, by Widrow and Ho? in 1959, of one of the?rst ad- tive?lters, the so-called least-mean-square, many applications appeared to have the potential to use this fundamental concept. While the number of - plications (using adaptive algorithms) has been (and keeps)?ourishing with time, thanks to several successes, the need for more sophisticated adaptive algorithms became obvious as real-world problems are more complex and more demanding. Even though the theory of adaptive?ltering is already a well-established topic in signal processing, new and improved concepts are discovered every year by researchers. Some of these recent approaches are discussed in this book. The goal of this book is to provide, for the?rst time, a reference to the hottest real-world applications where adaptive?ltering techniques play an important role. To do so, we invited top researchers in di?erent?elds to c- tribute chapters addressing their speci?c topic of study. Thousands of pages wouldprobablynotbe enoughto describeallthe practicalapplicationsutil- ing adaptive algorithms. Therefore, we limited the topics to some important applications in acoustics, speech, wireless, and networking, where research is still very active and open.

Despite their novelty, wavelets have a tremendous impact on a number of modern scientific disciplines, particularly on signal and image analysis. Because of their powerful underlying mathematical theory, they offer exciting opportunities for the design of new multi-resolution processing algorithms and effective pattern recognition systems. This book provides a much-needed overview of current trends in the practical application of wavelet theory. It combines cutting edge research in the rapidly developing wavelet theory with ideas from practical signal and image analysis fields. Subjects dealt with include balanced discussions on wavelet theory and its specific application in diverse fields, ranging from data compression to seismic equipment. In addition, the book offers insights into recent advances in emerging topics such as double density DWT, multiscale Bayesian estimation, symmetry and locality in image representation, and image fusion. Audience: This volume will be of interest to graduate students and researchers whose work involves acoustics, speech, signal and image processing, approximations and expansions, Fourier analysis, and medical imaging.

We are witnessing an ever-increasing thrust toward the era of multimedia information networks, largely spurred by the U.S. Government's proposal for the National Information Infrastructure in the fall of 1993. While more people are subscribing to the services of narrowband ISDN, the implementation of broadband ISDN by means of Asynchronous Transfer Mode (ATM) has accelerated since the formation of the ATM Forum in 1993. In the meantime, frame relay may prevail for inter-LAN connections. In the'upper layer'of the network, commercial use of Internet is rapidly emerging. To ensure the successful development of technology, it is vital to use a judicious approach in assessing the architecture and performance of the systems that implement the technology. It is this spirit that underlies the present conference, which is intended to provide an international forum for the presentation of recent research results in the area of local and metropolitan communication systems. This conference has two sets of predecessors. It is the third in a series of international conferences on Local and Metropolitan Communication Systems -LAN & MAN; the first was held in Toulouse in 1986 and the second in Palma de Mallorca in 1991. It is also the fourth in a triennial series organized by Kyoto University and others on the performance of communication-related systems; the previous ones were held in Tokyo (1985) and Kyoto (1988, 1991).

Open Distributed Processing contains the selected proceedings of the Third International Conference on Open Distributed Systems, organized by the International Federation for Information Processing and held in Brisbane, Australia, in February 1995. The book deals with the interconnectivity problems that advanced computer networking raises, providing those working in the area with the most recent research, including security and management issues.

The second edition of Loudspeaker Handbook follows the same general outlines as the highly successful first edition and has been augmented and updated in many areas of technology. Most notable are the developments in large-scale, programmable line arrays, distributed mode loudspeakers, and ultrasonic-based audio transduction. Additionally, the core chapters on low frequency systems, system concepts, and horn systems have been expanded to include both more analytical material and a richer array of examples. Much of the success of the first edition has been due to its accessibility both to loudspeaker engineers and to lay technicians working in the field - a point of view the author maintains in the present work. A full understanding of the underlying technology requires a fairly rigorous engineering background through the second year of professional study. At the same time, the generous use of graphs, with their intuitive thrust, will be useful to all readers. Loudspeaker Handbook, Second Edition continues to be appropriate for use in courses at the undergraduate senior level, for graduate students, and for professionals in audio and acoustical engineering.

Radar Signal Processing and Its Applications brings together in one place important contributions and up-to-date research results in this fast-moving area. In twelve selected chapters, it describes the latest advances in architectures, design methods, and applications of radar signal processing. The contributors to this work were selected from the leading researchers and practitioners in the field. This work, originally published as Volume 14, Numbers 1-3 of the journal, Multidimensional Systems and Signal Processing, will be valuable to anyone working or researching in the field of radar signal processing. It serves as an excellent reference, providing insight into some of the most challenging issues being examined today.

The study and implementation of microphone arrays originated over 20 years ago. Thanks to the research and experimental developments pursued to the present day, the field has matured to the point that array-based technology now has immediate applicability to a number of current systems and a vast potential for the improvement of existing products and the creation of future devices. In putting this book together, our goal was to provide, for the first time, a single complete reference on microphone arrays. We invited the top re­ searchers in the field to contribute articles addressing their specific topic(s) of study. The reception we received from our colleagues was quite enthusi­ astic and very encouraging. There was the general consensus that a work of this kind was well overdue. The results provided in this collection cover the current state of the art in microphone array research, development, and technological application. This text is organized into four sections which roughly follow the major areas of microphone array research today. Parts I and II are primarily the­ oretical in nature and emphasize the use of microphone arrays for speech enhancement and source localization, respectively. Part III presents a num­ ber of specific applications of array-based technology. Part IV addresses some open questions and explores the future of the field.

For many decades, hybrid devices have been used to connect 2-wire local circuits and 4-wire long distance circuits in telephone lines. This leads to a weIl known problem, whereby echoes are generated. The delay introduced by telecommunication satellites exacerbated this problem and the need for new methods of echo control soon became obvious. The best solution to date for solving this problem was invented in the 1960s at Bell Labs by Kelly, Logan, and Sondhi, and consists of identifying the echo path generated by the hybrid by means of an adaptive filter, a technique that became known as an echo canceler. The echo canceler allowed full-duplex communication which was not possible with older echo suppression techniques. Later, with the development of hands-free teleconferencing systems, an­ other echo problem appeared; but this time the echo was due to the coupling between the loudspeaker and microphone. It is not surprising that the same solution was proposed to solve this problem, and most of today's telecon­ ferencing systems have an acoustic echo canceler. More recently, attention has been given to the very interesting problem of multichannel acoustic echo cancellation, which leads to more exciting applications that take advantage of our binaural auditory system.

Digital Signal Processing for Communication Systems examines the plans for the future and the progress that has already been made, in the field of DSP and its applications to communication systems. The book pursues the progression from communication and information theory through to the implementation, evaluation and performance enhancing of practical communication systems using DSP technology. Digital Signal Processing for Communication Systems looks at various types of coding and modulation techniques, describing different applications of Turbo-Codes, BCH codes and general block codes, pulse modulations, and combined modulation and coding in order to improve the overall system performance. The book examines DSP applications in measurements performed for channel characterisation, pursues the use of DSP for design of effective channel simulators, and discusses equalization and detection of various signal formats for different channels. A number of system design issues are presented where digital signal processing is involved, reporting on the successful implementation of the system components using DSP technology, and including the problems involved with implementation of some DSP algorithms. Digital Signal Processing for Communication Systems serves as an excellent resource for professionals and researchers who deal with digital signal processing for communication systems, and may serve as a text for advanced courses on the subject.

Computer Networks, Architecture and Applications covers many aspects of research in modern communications networks for computing purposes.

The usage of data communications and computer networks are ever in­ creasing. It is one of the few technological areas which brings benefits to most of the countries and the peoples of the world. Without it many industries could not exist. It is the objective of this book to discuss data communications in a readable form that students and professionals all over the world can understand. As much as possible the text uses dia­ grams to illustrate key points. Most currently available data communications books take their view­ point from either a computer scientists top-down approach or from an electronic engineers bottom-up approach. This book takes a practical ap­ proach and supports it with a theoretical background to create a textbook which can be used by electronic engineers, computer engineers, computer scientists and industry professionals. It discusses most of the current and future key data communications technologies, including: • Data Communications Standards and Models; • Local Area Networks (Ethernet, Token Ring and FDDI); • Transmission Control ProtocollInternet Protocol (TCPIIP); • High-level Data Link Control (HDLC); • X.25 Packet-switching; • Asynchronous Communications (RS-232) and Modems; • Pulse Coded Modulation (PCM); • Integrated Digital Services Network (ISDN); • Asynchronous Transfer Mode (ATM); • Error Control; • X-Windows. The chapters are ordered in a possible structure for the presentation of the material and have not been sectioned into data communications areas.

Mechanical engineering, an engineering discipline borne of the needs of the industrial revolution, is once again asked to do its substantial share in the call for industrial renewal. The general call is urgent as we face profound issues of productivity and competitiveness that require engineering solutions, among oth­ ers. The Mechanical Engineering Series features graduate texts and research monographs intended to address the need for information in contemporary areas of mechanical engineering. The series is conceived as a comprehensive one that covers a broad range of concentrations important to mechanical engineering graduate education and research. We are fortunate to have a distinguished roster of consulting editors on the advisory board, each an expert in one of the areas of concentration. The names of the consulting editors are listed on the facing page of this volume. The areas of concentration are: applied mechanics; biomechanics; computational mechanics; dynamic systems and control; energetics; mechanics of materials; processing; thermal science; and tribology. I am pleased to present this volume in the Series: Electromechanical Sensors and Actuators, by Ilene Busch-Vishniac. The selection of this volume under­ scores again the interest of the Mechanical Engineering series to provide our readers with topical monographs as well as graduate texts in a wide variety of fields.

Data-Driven Techniques in Speech Synthesis gives a first review of this new field. All areas of speech synthesis from text are covered, including text analysis, letter-to-sound conversion, prosodic marking and extraction of parameters to drive synthesis hardware. Fuelled by cheap computer processing and memory, the fields of machine learning in particular and artificial intelligence in general are increasingly exploiting approaches in which large databases act as implicit knowledge sources, rather than explicit rules manually written by experts. Speech synthesis is one application area where the new approach is proving powerfully effective, the reliance upon fragile specialist knowledge having hindered its development in the past. This book provides the first review of the new topic, with contributions from leading international experts. Data-Driven Techniques in Speech Synthesis is at the leading edge of current research, written by well respected experts in the field. The text is concise and accessible, and guides the reader through the new technology. The book will primarily appeal to research engineers and scientists working in the area of speech synthesis. However, it will also be of interest to speech scientists and phoneticians as well as managers and project leaders in the telecommunications industry who need an appreciation of the capabilities and potential of modern speech synthesis technology.

What is'digital telephony'? To the authors, the term digital telephony denotes the technology used to provide a completely digital telecommunication system from end-to-end. This implies the use of digital technology from one end instru­ ment through transmission facilities and switching centers to another end instru­ ment. Digital telephony has become possible only because of the recent and on­ going surge of semiconductor developments, allowing microminiaturization and high reliability along with reduced costs. This book deals with both the future and the present. Thus, the first chapter is entitled,'A Network in Transition.'As baselines, Chapters 2 and 11 provide the reader with the present status of teler-hone technology in terms of voice digiti­ zation as well as switching principles. The book is an outgrowth of the authors'consulting and teaching experience in the field since the early 1980s. The book has been written to provide both the engineering student and the practicing engineer a working knowledge of the prin­ ciples of present and future telecommunication systems based upon the use of the public switched network. Problems or discussion questions have been included at the ends of the chapters to facilitate the book's use as a senior-level or first­ year graduate-level course text. Numerous clients and associates of the authors as well as hundreds of others have provided useful information and examples for the text, and the authors wish to thank all those who have so contributed either directly or indirectly.

Blind Signal Separation (BSS) deals with recovering (filtered versions of) source signals from an observed mixture thereof. The term `blind'relates to the fact that there are no reference signals for the source signals and also that the mixing system is unknown. This book presents a new method for blind signal separation, which is developed to work on microphone signals. Acoustic Echo Cancellation (AEC) is a well-known technique to suppress the echo that a microphone picks up from a loudspeaker in the same room. Such acoustic feedback occurs for example in hands-free telephony and can lead to a perceived loud tone. For an application such as a voice-controlled television, a stereo AEC is required to suppress the contribution of the stereo loudspeaker setup. A generalized AEC is presented that is suited for multi-channel operation. New algorithms for Blind Signal Separation and multi-channel Acoustic Echo Cancellation are presented. A background is given in array signal processing methods, adaptive filter theory, and fast filtering in the frequency domain. The included CD-ROM can be played using any compact disc player to play the simulation results that are described in the text. When inserted into a computer, it furthermore gives Matlab implementations of the new algorithms along with audio data with which to experiment. This makes the book suited to researchers, engineers, and university students, who want to get acquainted with these emerging fields.

There is a wide field of tasks left that can only be satisfyingly attacked with the help of old-fashioned analogue technology, and one of the most important are amplifiers for analogue signals. The strongly expanded content of the second edition of'the sound of silence'leads to affordable amplifier design approaches which will end up in lowest-noise solutions not far away from the edge of physical boundaries set by room temperature and given cartridges - thus, fully compatible with very expensive so called'high-end'or'state-of-the-art'offers on today markets - and, from a noise point of view in most cases outperforming them! With easy to follow mathematical treatment it is demonstrated as well that theory is not far away from reality. Measured SNs will be found within 1dB off the calculated ones and deviations from the exact amplifier transfer won't cross the ± 0.1dB tolerance lines. Additionally, the book presents measurement set-ups and results. Consequently, comparisons with measurement results of test magazine will soon become easier to perform. This new edition includes a new chapters about reference levels, Noise in Amp Input sections, Humming Problems, and much more.

An emerging technology, Speaker Recognition is becoming well-known for providing voice authentication over the telephone for helpdesks, call centres and other enterprise businesses for business process automation.'Fundamentals of Speaker Recognition'introduces Speaker Identification, Speaker Verification, Speaker (Audio Event) Classification, Speaker Detection, Speaker Tracking and more. The technical problems are rigorously defined, and a complete picture is made of the relevance of the discussed algorithms and their usage in building a comprehensive Speaker Recognition System. Designed as a textbook with examples and exercises at the end of each chapter,'Fundamentals of Speaker Recognition'is suitable for advanced-level students in computer science and engineering, concentrating on biometrics, speech recognition, pattern recognition, signal processing and, specifically, speaker recognition. It is also a valuable reference for developers of commercial technology and for speech scientists. Please click on the link under'Additional Information'to view supplemental information including the Table of Contents and Index.

This is an up-to-date reference and textbook on modern acoustics from a signal-theoretic point of view, as well as a wave-theoretic approach for students, engineers, and researchers. It provides readers the fundamental basis of acoustics and vibration science and proceeds up to recent hot topics related to acoustic transfer functions and signal analysis including a perceptual point of view.In the first part, the work uniquely introduces into the fundamentals without using heavy mathematics The following, advanced chapters deal with new and deep insights into acoustic signal analysis and investigation of room transfer functions based on the poles and zeros.

Dieser Bericht behandelt die akustischen Aspekte der sprachlichen Kommunikation, insbesondere wenn sie durch Geräusche, Nachhall oder Schwerhörigkeit gestört ist. Produktion und Wahrnehmung von Sprache, also Sprechen und Hören, dienen wesentlich der Integration eines Menschen in das soziale und kulturelle Leben. Beides ist die Voraussetzung für ein kompetentes Verhalten in der Familie, in der Öffentlichkeit, am Arbeitsplatz und in der Ausbildung. Der Fortschritt der Technik verlangt notwendig nach mehr Kommunikation: u.a. aufgrund veränderter Betriebsabläufe (Teamarbeit zur Verbindung von Entwicklung, Konstruktion, Produktion, Verteilung der Arbeit an unterschiedlichen Standorten), an mobilen Arbeitsplätzen, bei der Benutzung akustischer Medien oder mobiler Telefone und beim weltweiten Kommunizieren in einer Zweitsprache. Die Auswirkungen reduzierter Kommunikation aufgrund von Lärm oder Schwerhörigkeit sind deshalb beträchtlich. Durch akustisch schlecht gestaltete Räume, Arbeitsplätze und Kommunikationsgeräte und durch zu hohe Sprechpegel, z.B. aus Lautsprechern, wird die Sprachkommunikation zusätzlich eingeschränkt. Damit ist nicht nur Wahrnehmen und Verstehen von Sprache reduziert, sondern zwingend auch der Informationsaustausch: die private, soziale und betriebliche Interaktion. Zu beachten sind ebenso auch langfristige Folgen, wie z.B. ein verzögerter Spracherwerb in Kindergarten und Schule, die Abnahme der Qualität der Arbeit, die Zunahme von Konflikten und von Unfällen am Arbeitsplatz und eine Verminderung der physischen und psychischen Gesundheit. Ausgangspunkt der Analyse ist das Sprecher-Hörer-Modell im Kontext der Psychoakustik. Einleitend werden wichtige Grundlagen des Sprechens, der Sprache, der Wahrnehmung und des Verstehens von Sprache dargestellt: der Sprecher produziert Sprache (Sprechweise, Artikulation), die zum Hörer im Raum oder über elektroakustische Anlagen übertragen wird und gestört werden kann (Bandbegrenzung,Geräusche, Nachhall, Übersteuerung etc.). Der Hörer muss die Sprache wahrnehmen und verstehen, auch wenn er schwerhörig ist oder Gehörschutz trägt. Die Messung der Sprachverständlichkeit wird anhand der Sprachindices erläutert. Die Methoden zur Bestimmung der Qualität der Sprachkommunikation werden aufbauend auf der Verständlichkeit durch Parameter wie Konzentration und Belästigung ergänzt und neu formuliert. Die so entwickelten Qualitätsstufen werden ausführlich diskutiert, dazu werden Kriterien für die Raumgestaltung (u.a. Schulen) abgeleitet. Die Analyse soll es ermöglichen, den Umgebungsbereich für sprachliche Kommunikation'barrierefrei'zu gestalten, so dass die Kommunikationspartner ungeachtet ihres Alters, individueller Handicaps (begrenzte Sprachkompetenz, Schwerhörigkeit) in öffentlichen Bereichen und in der Arbeitswelt als gleichberechtigte Partner am Gespräch teilnehmen können. Ziel dieses Buches ist es auch über die Akustik hinaus einen großen Leserkreis unterschiedlicher Interessen und Berufsgruppen wie Ergonomen, Psychologen, Mediziner, Architekten, Medienexperten in die Diskussion über die Sprachkommunikation aus wissenschaftlicher und angewandter Sicht einzuführen und Hinweise für eine optimale Gestaltung zu geben.

We live in a noisy world! In all applications (telecommunications, hands-free communications, recording, human-machine interfaces, etc.) that require at least one microphone, the signal of interest is usually contaminated by noise and reverberation. As a result, the microphone signal has to be'cleaned'with digital signal processing tools before it is played out, transmitted, or stored. This book is about speech enhancement. Different well-known and state-of-the-art methods for noise reduction, with one or multiple microphones, are discussed. By speech enhancement, we mean not only noise reduction but also dereverberation and separation of independent signals. These topics are also covered in this book. However, the general emphasis is on noise reduction because of the large number of applications that can benefit from this technology. The goal of this book is to provide a strong reference for researchers, engineers, and graduate students who are interested in the problem of signal and speech enhancement. To do so, we invited well-known experts to contribute chapters covering the state of the art in this focused field.

Foundations of Voice and Speech Quality Perception starts out with the fundamental question of:'How do listeners perceive voice and speech quality and how can these processes be modeled?'Any quantitative answers require measurements. This is natural for physical quantities but harder to imagine for perceptual measurands. This book approaches the problem by actually identifying major perceptual dimensions of voice and speech quality perception, defining units wherever possible and offering paradigms to position these dimensions into a structural skeleton of perceptual speech and voice quality. The emphasis is placed on voice and speech quality assessment of systems in artificial scenarios. Many scientific fields are involved. This book bridges the gap between two quite diverse fields, engineering and humanities, and establishes the new research area of Voice and Speech Quality Perception.