During the last few years cavity-optomechanics has emerged as a new field of research. This highly interdisciplinary field studies the interaction between micro and nano mechanical systems and light. Possible applications range from novel high-bandwidth mechanical sensing devices through the generation of squeezed optical or mechanical states to even tests of quantum theory itself. This is one of the first books in this relatively young field. It is aimed at scientists, engineers and students who want to obtain a concise introduction to the state of the art in the field of cavity optomechanics. It is valuable to researchers in nano science, quantum optics, quantum information, gravitational wave detection and other cutting edge fields. Possible applications include biological sensing, frequency comb applications, silicon photonics etc. The technical content will be accessible to those who have familiarity with basic undergraduate physics.

This book provides developers, engineers, researchers and students with detailed knowledge about the High Efficiency Video Coding (HEVC) standard. HEVC is the successor to the widely successful H.264/AVC video compression standard, and it provides around twice as much compression as H.264/AVC for the same level of quality. The applications for HEVC will not only cover the space of the well-known current uses and capabilities of digital video – they will also include the deployment of new services and the delivery of enhanced video quality, such as ultra-high-definition television (UHDTV) and video with higher dynamic range, wider range of representable color, and greater representation precision than what is typically found today. HEVC is the next major generation of video coding design – a flexible, reliable and robust solution that will support the next decade of video applications and ease the burden of video on world-wide network traffic. This book provides a detailed explanation of the various parts of the standard, insight into how it was developed, and in-depth discussion of algorithms and architectures for its implementation.

This book provides a comprehensive overview of the recent advancement in the field of automatic speech recognition with a focus on deep learning models including deep neural networks and many of their variants. This is the first automatic speech recognition book dedicated to the deep learning approach. In addition to the rigorous mathematical treatment of the subject, the book also presents insights and theoretical foundation of a series of highly successful deep learning models.

E.F.F. Chladni's experiments and observations with sound and vibrations profoundly influenced the development of the field of Acoustics. The famous Chladni diagrams along with other observations are contained in Die Akustik, published in German in 1802 and Traite d'Acoustique, a greatly expanded version, published in French in 1809. This is the first comprehensive translation of the expanded French version of Traite d'Acoustique, using the 1802 German publication for reference and clarification. The translation was undertaken by Robert T. Beyer, PhD (1920-2008), noted acoustician, Professor of Physics at Brown University, and Gold Medal recipient of the Acoustical Society of America. Along with many other projects completed over the course of his career, Dr. Beyer translated Von Neumann's seminal work, Mathematical Foundations of Quantum Mechanics from the original German, spent 30 years translating Russian physics treatises and journals, served as editor of the English translation of the Soviet Journal of Experimental and Theoretical Physics, and also authored Sounds of our Times: Two Hundred Years of Acoustics.

The Wireless Identification and Sensing Platform (WISP) is the first of a new class of RF-powered sensing and computing systems. Rather than being powered by batteries, these sensor systems are powered by radio waves that are either deliberately broadcast or ambient. Enabled by ongoing exponential improvements in the energy efficiency of microelectronics, RF-powered sensing and computing is rapidly moving along a trajectory from impossible (in the recent past), to feasible (today), toward practical and commonplace (in the near future). This book is a collection of key papers on RF-powered sensing and computing systems including the WISP. Several of the papers grew out of the WISP Challenge, a program in which Intel Corporation donated WISPs to academic applicants who proposed compelling WISP-based projects. The book also includes papers presented at the first WISP Summit, a workshop held in Berkeley, CA in association with the ACM Sensys conference, as well as other relevant papers. The book provides a window into the fascinating new world of wirelessly powered sensing and computing.

This textbook provides both profound technological knowledge and a comprehensive treatment of essential topics in music processing and music information retrieval. Including numerous examples, figures, and exercises, this book is suited for students, lecturers, and researchers working in audio engineering, computer science, multimedia, and musicology. The book consists of eight chapters. The first two cover foundations of music representations and the Fourier transform-concepts that are then used throughout the book. In the subsequent chapters, concrete music processing tasks serve as a starting point. Each of these chapters is organized in a similar fashion and starts with a general description of the music processing scenario at hand before integrating it into a wider context. It then discusses-in a mathematically rigorous way-important techniques and algorithms that are generally applicable to a wide range of analysis, classification, and retrieval problems. At the same time, the techniques are directly applied to a specific music processing task. By mixing theory and practice, the book's goal is to offer detailed technological insights as well as a deep understanding of music processing applications. Each chapter ends with a section that includes links to the research literature, suggestions for further reading, a list of references, and exercises. The chapters are organized in a modular fashion, thus offering lecturers and readers many ways to choose, rearrange or supplement the material. Accordingly, selected chapters or individual sections can easily be integrated into courses on general multimedia, information science, signal processing, music informatics, or the digital humanities.

This book provides the reader with the knowledge necessary for comprehension of the field of Intelligent Audio Analysis. It firstly introduces standard methods and discusses the typical Intelligent Audio Analysis chain going from audio data to audio features to audio recognition. Further, an introduction to audio source separation, and enhancement and robustness are given. After the introductory parts, the book shows several applications for the three types of audio: speech, music, and general sound. Each task is shortly introduced, followed by a description of the specific data and methods applied, experiments and results, and a conclusion for this specific task. The books provides benchmark results and standardized test-beds for a broader range of audio analysis tasks. The main focus thereby lies on the parallel advancement of realism in audio analysis, as too often today's results are overly optimistic owing to idealized testing conditions, and it serves to stimulate synergies arising from transfer of methods and leads to a holistic audio analysis.

This book offers an overview of models, measurements, calculations and examples connecting musical acoustics and music psychology. Indeed, many mathematical formulations that explain musical acoustics can also be used to help predict human auditory perception.

Der Band beschreibt die Entstehung, Ausbreitung, Abstrahlung und Messung von Korperschall - wichtige Themen fur die Larmminderung bei Maschinen oder Gebauden, aber auch bei der Messung mechanischer Materialdaten. In der 3. Auflage wurde der Band erneuert mit dem Ziel, den Geist des ursprunglichen Werks (Lothar Cremer/Manfred Heckl) zu bewahren und es zugleich an den aktuellen Wissensstand anzupassen. So fuhrt das erste Kapitel jetzt in den Korperschall und die physikalischen Prinzipien ein, der Messtechnik ist ein eigener Abschnitt gewidmet."

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.

Digital measurement of the analog acoustical parameters of a music performance hall is difficult. The aim of such work is to create a digital acoustical derivation that is an accurate numerical representation of the complex analog characteristics of the hall. The present study describes the exponential sine sweep (ESS) measurement process in the derivation of an acoustical impulse response function (AIRF) of three music performance halls in Canada. It examines specific difficulties of the process, such as preventing the external effects of the measurement transducers from corrupting the derivation, and provides solutions, such as the use of filtering techniques in order to remove such unwanted effects. In addition, the book presents a novel method of numerical verification through mean-squared error (MSE) analysis in order to determine how accurately the derived AIRF represents the acoustical behavior of the actual hall.

As speech processing devices like mobile phones, voice controlled devices, and hearing aids have increased in popularity, people expect them to work anywhere and at any time without user intervention. However, the presence of acoustical disturbances limits the use of these applications, degrades their performance, or causes the user difficulties in understanding the conversation or appreciating the device. A common way to reduce the effects of such disturbances is through the use of single-microphone noise reduction algorithms for speech enhancement. The field of single-microphone noise reduction for speech enhancement comprises a history of more than 30 years of research. In this survey, we wish to demonstrate the significant advances that have been made during the last decade in the field of discrete Fourier transform domain-based single-channel noise reduction for speech enhancement.Furthermore, our goal is to provide a concise description of a state-of-the-art speech enhancement system, and demonstrate the relative importance of the various building blocks of such a system. This allows the non-expert DSP practitioner to judge the relevance of each building block and to implement a close-to-optimal enhancement system for the particular application at hand. Table of Contents: Introduction / Single Channel Speech Enhancement: General Principles / DFT-Based Speech Enhancement Methods: Signal Model and Notation / Speech DFT Estimators / Speech Presence Probability Estimation / Noise PSD Estimation / Speech PSD Estimation / Performance Evaluation Methods / Simulation Experiments with Single-Channel Enhancement Systems / Future Directions

Noise is a widely recognized problem and health concern in the modern world. Given the importance of managing noise levels and developing suitable 'soundscapes' in contexts such as industry, schools, or public spaces, this is an area of active research for acousticians. But noise, in the sense of dissonance, can also be used positively; composers have employed it from Baroque music to Rock feedback; medicine harnesses it to shatter kidney stones and treat cancer; and even the military uses it in (real and rumoured) weapons. Mike Goldsmith looks back at the long history of the battle between people and noise - a battle that has changed our lives and moulded our societies. He investigates how increasing noise levels relate to human progress, from the clatter of wheels on cobbles to the sound of heavy machinery; he explains how our scientific understanding of sound and hearing has developed; and he looks at noise in nature, including the remarkable ways in which some animals, such as shrimps, use noise as a weapon or to catch prey. He concludes by turning to the future, discussing the noise sources which are likely to dominate it and the ways in which new science and new ideas may change the way our future will sound.

This book addresses the problem of articulatory speech synthesis based on computed vocal tract geometries and the basic physics of sound production in it. Unlike conventional methods based on analysis/synthesis using the well-known source filter model, which assumes the independence of the excitation and filter, we treat the entire vocal apparatus as one mechanical system that produces sound by means of fluid dynamics. The vocal apparatus is represented as a three-dimensional time-varying mechanism and the sound propagation inside it is due to the non-planar propagation of acoustic waves through a viscous, compressible fluid described by the Navier-Stokes equations. We propose a combined minimum energy and minimum jerk criterion to compute the dynamics of the vocal tract during articulation. Theoretical error bounds and experimental results show that this method obtains a close match to the phonetic target positions while avoiding abrupt changes in the articulatory trajectory. The vocal folds are set into aerodynamic oscillation by the flow of air from the lungs. The modulated air stream then excites the moving vocal tract. This method shows strong evidence for source-filter interaction. Based on our results, we propose that the articulatory speech production model has the potential to synthesize speech and provide a compact parameterization of the speech signal that can be useful in a wide variety of speech signal processing problems. Table of Contents: Introduction / Literature Review / Estimation of Dynamic Articulatory Parameters / Construction of Articulatory Model Based on MRI Data / Vocal Fold Excitation Models / Experimental Results of Articulatory Synthesis / Conclusion

If you ve ever handled live sound, you know the recipe for creating quality live sound requires many steps. Your list of ingredients, shall we say, requires an understanding of sound and how it behaves, the know-how to effectively use a sound system), and the knowledge to choose and use your gear well. Add a dash of miking ability, stir in a pinch of thinking on your feet for when your system starts to hum or the vocals start to feed back, and mix.

In practice, there really is no "recipe" for creating a quality performance. Instead, musicians and engineers who effectively use sound systems have a wealth of knowledge that informs their every move before and during a live performance. You can slowly gather that knowledge over years of live performance, or you can speed up the process with "The SOS Guide to Live Sound."

With these pages, you get practical advice that will allow you to accomplish your live-sound goals in every performance. Learn how to choose, set up, and use a live-performance sound system. Get the basics of live-sound mixing, save money by treating your gear well with a crash course in maintenance, and fix issues as they happen with a section on problem-solving, full of real-world situations. You ll also get information on stage-monitoring, both conventional and in-ear, along with the fundamentals of radio microphones and wireless mixing solutions. Finally, a comprehensive glossary of terminology rounds out this must-have reference."

This book is about music. the instruments and players who produce it. and the technologies that support it. Although much modern music is produced by electronic means. its underlying basis is still traditional acoustical sound production. and that broad topic provides the basis for this book. There are many fine books available that treat musical acoustics largely from the physical point of view. The approach taken here is to present only the fundamentals of musical physics. while giving special emphasis to the relation between instrument and player and stressing the characteristics of instruments that are of special concern to engineers and technicians in volved in the fields of recording. sound reinforcement. and broadcasting. In order to understand musical instruments in their normal performance environments. the student must have a basic working knowledge of physical and architectural acoustics. The book begins with a review of the elements of acoustics. stressing the nature of sound fields and phenomena that are wavelength-dependent. The book then moves on to a discussion of those aspects of psychological acoustics that are of special concern to music technicians. most notably concepts of stereophonic imaging. loudness-related phenomena. and critical band theory."

This book introduces the theory, algorithms, and implementation techniques for efficient decoding in speech recognition mainly focusing on the Weighted Finite-State Transducer (WFST) approach. The decoding process for speech recognition is viewed as a search problem whose goal is to find a sequence of words that best matches an input speech signal. Since this process becomes computationally more expensive as the system vocabulary size increases, research has long been devoted to reducing the computational cost. Recently, the WFST approach has become an important state-of-the-art speech recognition technology, because it offers improved decoding speed with fewer recognition errors compared with conventional methods. However, it is not easy to understand all the algorithms used in this framework, and they are still in a black box for many people. In this book, we review the WFST approach and aim to provide comprehensive interpretations of WFST operations and decoding algorithms to help anyone who wants to understand, develop, and study WFST-based speech recognizers. We also mention recent advances in this framework and its applications to spoken language processing. Table of Contents: Introduction / Brief Overview of Speech Recognition / Introduction to Weighted Finite-State Transducers / Speech Recognition by Weighted Finite-State Transducers / Dynamic Decoders with On-the-fly WFST Operations / Summary and Perspective