Now in its tenth edition, the Audio Production Worktext offers a comprehensive introduction to audio production in radio, television, and film. This hands-on, student-friendly text demonstrates how to navigate modern radio production studios and utilize the latest equipment and software. Key chapters address production planning, the use of microphones, audio consoles, and sound production for the visual media. The reader is shown the reality of audio production both within the studio and on location. New to this edition is material covering podcasting, including online storage and distribution. The new edition also includes an updated glossary and appendix on analog and original digital applications, as well as self-study questions and projects that students can use to further enhance their learning. The accompanying instructor website has been refreshed and includes an instructor's manual and PowerPoint images. This book remains an essential text for audio and media production students seeking a thorough introduction to the field.

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.

This agenda-setting book presents state of the art research in Music and Human-Computer Interaction (also known as 'Music Interaction'). Music Interaction research is at an exciting and formative stage. Topics discussed include interactive music systems, digital and virtual musical instruments, theories, methodologies and technologies for Music Interaction. Musical activities covered include composition, performance, improvisation, analysis, live coding, and collaborative music making. Innovative approaches to existing musical activities are explored, as well as tools that make new kinds of musical activity possible. Music and Human-Computer Interaction is stimulating reading for professionals and enthusiasts alike: researchers, musicians, interactive music system designers, music software developers, educators, and those seeking deeper involvement in music interaction. It presents the very latest research, discusses fundamental ideas, and identifies key issues and directions for future work.

In December 1974 the first realtime conversation on the ARPAnet took place between Culler-Harrison Incorporated in Goleta, California, and MIT Lincoln Laboratory in Lexington, Massachusetts. This was the first successful application of realtime digital speech communication over a packet network and an early milestone in the explosion of realtime signal processing of speech, audio, images, and video that we all take for granted today. It could be considered as the first voice over Internet Protocol (VoIP), except that the Internet Protocol (IP) had not yet been established. In fact, the interest in realtime signal processing had an indirect, but major, impact on the development of IP. This is the story of the development of linear predictive coded (LPC) speech and how it came to be used in the first successful packet speech experiments. Several related stories are recounted as well. The history is preceded by a tutorial on linear prediction methods which incorporates a variety of views to provide context for the stories. This part is a technical survey of the fundamental ideas of linear prediction that are important for speech processing, but the development departs from traditional treatments and takes advantage of several shortcuts, simplifications, and unifications that come with years of hindsight. In particular, some of the key results are proved using short and simple techniques that are not as well known as they should be, and it also addresses some of the common assumptions made when modeling random signals. Linear Predictive Coding and the Internet Protocol is an insightful and comprehensive review of an underpinning technology of the internet and other packet switched networks. It will be enjoyed by everyone with an interest in past and present real time signal processing on the internet.

Current speech recognition systems are based on speaker independent speech models and suffer from inter-speaker variations in speech signal characteristics. This work develops an integrated approach for speech and speaker recognition in order to gain space for self-learning opportunities of the system. This work introduces a reliable speaker identification which enables the speech recognizer to create robust speaker dependent models In addition, this book gives a new approach to solve the reverse problem, how to improve speech recognition if speakers can be recognized. The speaker identification enables the speaker adaptation to adapt to different speakers which results in an optimal long-term adaptation.

Spoken dialog systems have the potential to offer highly intuitive user interfaces, as they allow systems to be controlled using natural language. However, the complexity inherent in natural language dialogs means that careful testing of the system must be carried out from the very beginning of the design process. This book examines how user models can be used to support such early evaluations in two ways: by running simulations of dialogs, and by estimating the quality judgments of users. First, a design environment supporting the creation of dialog flows, the simulation of dialogs, and the analysis of the simulated data is proposed. How the quality of user simulations may be quantified with respect to their suitability for both formative and summative evaluation is then discussed. The remainder of the book is dedicated to the problem of predicting quality judgments of users based on interaction data. New modeling approaches are presented, which process the dialogs as sequences, and which allow knowledge about the judgment behavior of users to be incorporated into predictions. All proposed methods are validated with example evaluation studies.

In Max/MSP/Jitter for Music, expert author and music technologist V. J. Manzo provides a user-friendly introduction to a powerful programming language that can be used to write custom software for musical interaction. Through clear, step-by-step instructions illustrated with numerous examples of working systems, the book equips readers with everything they need to know in order to design and complete meaningful music projects. The book also discusses ways to interact with software beyond the mouse and keyboard through use of camera tracking, pitch tracking, video game controllers, sensors, mobile devices, and more. The book does not require any prerequisite programming skills, but rather walks readers through a series of small projects through which they will immediately begin to develop software applications for practical musical projects. As the book progresses, and as the individual's knowledge of the language grows, the projects become more sophisticated. This new and expanded second edition brings the book fully up-to-date including additional applications in integrating Max with Ableton Live. It also includes a variety of additional projects as part of the final three project chapters. The book is of special value both to software programmers working in Max/MSP/Jitter and to music educators looking to supplement their lessons with interactive instructional tools, develop adaptive instruments to aid in student composition and performance activities, and create measurement tools with which to conduct music education research.

Digital Speech Processing Using Matlab deals with digital speech pattern recognition, speech production model, speech feature extraction, and speech compression. The book is written in a manner that is suitable for beginners pursuing basic research in digital speech processing. Matlab illustrations are provided for most topics to enable better understanding of concepts. This book also deals with the basic pattern recognition techniques (illustrated with speech signals using Matlab) such as PCA, LDA, ICA, SVM, HMM, GMM, BPN, and KSOM.

Magneto-resistive recording heads are sensors that exploit magneto resistance effects to read digital magnetically recorded data. The industry of disk drives is growing because of the need for increased storage capacity.

In this work, the authors present a fully statistical approach to model non--native speakers' pronunciation. Second-language speakers pronounce words in multiple different ways compared to the native speakers. Those deviations, may it be phoneme substitutions, deletions or insertions, can be modelled automatically with the new method presented here.

The methods is based on a discrete hidden Markov model as a word pronunciation model, initialized on a standard pronunciation dictionary. The implementation and functionality of the methodology has been proven and verified with a test set of non-native English in the regarding accent.

The book is written for researchers with a professional interest in phonetics and automatic speech and speaker recognition.

Proactive Spoken Dialogue Interaction in Multi-Party Environments describes spoken dialogue systems that act as independent dialogue partners in the conversation with and between users. The resulting novel characteristics such as proactiveness and multi-party capabilities pose new challenges on the dialogue management component of such a system and require the use and administration of an extensive dialogue history. In order to assist the proactive spoken dialogue systems development, a comprehensive data collection seems mandatory and may be performed in a Wizard-of-Oz environment. Such an environment builds also the appropriate basis for an extensive usability and acceptance evaluation.

Proactive Spoken Dialogue Interaction in Multi-Party Environments is a useful reference for students and researchers in speech processing.

This book presents computational methods for extracting the useful information from audio signals, collecting the state of the art in the field of sound event and scene analysis. The authors cover the entire procedure for developing such methods, ranging from data acquisition and labeling, through the design of taxonomies used in the systems, to signal processing methods for feature extraction and machine learning methods for sound recognition. The book also covers advanced techniques for dealing with environmental variation and multiple overlapping sound sources, and taking advantage of multiple microphones or other modalities. The book gives examples of usage scenarios in large media databases, acoustic monitoring, bioacoustics, and context-aware devices. Graphical illustrations of sound signals and their spectrographic representations are presented, as well as block diagrams and pseudocode of algorithms.

This is an edited volume, written by well-recognized international researchers with extended chapter style versions of the best papers presented at the SITIS 2006 International Conference. This book presents the state-of-the-art and recent research results on the application of advanced signal processing techniques for improving the value of image and video data. It introduces new results on video coding on time-honored topic of securing image information. The book is designed for a professional audience composed of practitioners and researchers in industry. This book is also suitable for advanced-level students in computer science.

The accurate determination of the speech spectrum, particularly for short frames, is commonly pursued in diverse areas including speech processing, recognition, and acoustic phonetics. With this book the author makes the subject of spectrum analysis understandable to a wide audience, including those with a solid background in general signal processing and those without such background. In keeping with these goals, this is not a book that replaces or attempts to cover the material found in a general signal processing textbook. Some essential signal processing concepts are presented in the first chapter, but even there the concepts are presented in a generally understandable fashion as far as is possible. Throughout the book, the focus is on applications to speech analysis; mathematical theory is provided for completeness, but these developments are set off in boxes for the benefit of those readers with sufficient background. Other readers may proceed through the main text, where the key results and applications will be presented in general heuristic terms, and illustrated with software routines and practical "show-and-tell" discussions of the results. At some points, the book refers to and uses the implementations in the Praat speech analysis software package, which has the advantages that it is used by many scientists around the world, and it is free and open source software. At other points, special software routines have been developed and made available to complement the book, and these are provided in the Matlab programming language. If the reader has the basic Matlab package, he/she will be able to immediately implement the programs in that platform---no extra "toolboxes" are required.

This book addresses the issue of music consumption in the digital era of technologies. It explores how individuals use music in the context of their everyday lives and how, in return, music acquires certain roles within everyday contexts and more broadly in their life narratives.

Speech Processing has rapidly emerged as one of the most widespread and well-understood application areas in the broader discipline of Digital Signal Processing. Besides the telecommunications applications that have hitherto been the largest users of speech processing algorithms, several non-traditional embedded processor applications are enhancing their functionality and user interfaces by utilizing various aspects of speech processing.

"Speech Processing in Embedded Systems" describes several areas of speech processing, and the various algorithms and industry standards that address each of these areas. The topics covered include different types of Speech Compression, Echo Cancellation, Noise Suppression, Speech Recognition and Speech Synthesis. In addition this book explores various issues and considerations related to efficient implementation of these algorithms on real-time embedded systems, including the role played by processor CPU and peripheral functionality.

This book describes the basic principles underlying the generation, coding and transmission of speech and audio signals and reveals the latest advances in this area. Waveform coding and parametric coding of speech are described and the fundamental principles behind these methods are delineated. Examples of speech coding standards in use today and their practical implementation are discussed. The principles underlying speech enhancement and speech recognition are also presented, along with the latest recent advances in these areas.

"Adaptive Digital Filters" presents an important discipline applied to the domain of speech processing. The book first makes the reader acquainted with the basic terms of filtering and adaptive filtering, before introducing the field of advanced modern algorithms, some of which are contributed by the authors themselves. Working in the field of adaptive signal processing requires the use of complex mathematical tools. The book offers a detailed presentation of the mathematical models that is clear and consistent, an approach that allows everyone with a college level of mathematics knowledge to successfully follow the mathematical derivations and descriptions of algorithms.
The algorithms are presented in flow charts, which facilitates their practical implementation. The book presents many experimental results and treats the aspects of practical application of adaptive filtering in real systems, making it a valuable resource for both undergraduate and graduate students, and for all others interested in mastering this important field.

The second edition of Human Factors and Voice Interactive Systems, in addition to updating chapters from the first edition, adds in-depth information on current topics of major interest to speech application developers. These topics include use of speech technologies in automobiles, speech in mobile phones, natural language dialogue issues in speech application design, and the human factors design, testing, and evaluation of interactive voice response (IVR) applications.

Fully updated, revised, and expanded, this second edition of Modern Cable Television Technology addresses the significant changes undergone by cable since 1999--including, most notably, its continued transformation from a system for delivery of television to a scalable-bandwidth platform for a broad range of communication services. It provides in-depth coverage of high speed data transmission, home networking, IP-based voice, optical dense wavelength division multiplexing, new video compression techniques, integrated voice/video/data transport, and much more.
Intended as a day-to-day reference for cable engineers, this book illuminates all the technologies involved in building and maintaining a cable system. But it's also a great study guide for candidates for SCTE certification, and its careful explanations will benefit any technician whose work involves connecting to a cable system or building products that consume cable services.
*Written by four of the most highly-esteemed cable engineers in the industry with a wealth of experience in cable, consumer electronics, and telecommunications.
*All new material on digital technologies, new practices for delivering high speed data, home networking, IP-based voice technology, optical dense wavelength division multiplexing (DWDM), new video compression techniques, and integrated voice/video/data transport.
*Covers the latest on emerging digital standards for voice, data, video, and multimedia.
*Presents distribution systems, from drops through fiber optics, an covers everything from basic principles to network architectures.

The vision of a world in which privacy persists and security is ensured but the full potential of the technology is nevertheless tapped guides this work. It is argued that security and privacy can be ensured using technical safeguards if the whole RFID system is designed properly. The challenge is immense since many constraints exist for providing security and privacy in RFID systems: technically and economically but also ethically and socially. Not only security and privacy needs to be provided but the solutions also need to be inexpensive, practical, reliable, scalable, flexible, inter-organizational, and lasting.

After analyzing the problem area in detail, this work introduces a number of new concepts and protocols that provide security and ensure privacy in RFID systems by technical means. The classic RFID model is extended and considerations in new directions are taken. This leads to innovative solutions with advantageous characteristics. Finally, a comprehensive framework including required protocols for operation is proposed. It can be used within a global scope, supports inter-organizational cooperation and data sharing, and adheres to all the architectural guidelines derived in this work. Security and privacy is provided by technical means in an economic manner. Altogether, the goal of building scalable and efficient RFID systems on a global, inter-organizational scale without neglecting security and privacy has been achieved well.

This volume comprises eight well-versed contributed chapters devoted to report the latest findings on the intelligent approaches to multimedia data analysis. Multimedia data is a combination of different discrete and continuous content forms like text, audio, images, videos, animations and interactional data. At least a single continuous media in the transmitted information generates multimedia information. Due to these different types of varieties, multimedia data present varied degrees of uncertainties and imprecision, which cannot be easy to deal by the conventional computing paradigm. Soft computing technologies are quite efficient to handle the imprecision and uncertainty of the multimedia data and they are flexible enough to process the real-world information. Proper analysis of multimedia data finds wide applications in medical diagnosis, video surveillance, text annotation etc. This volume is intended to be used as a reference by undergraduate and post graduate students of the disciplines of computer science, electronics and telecommunication, information science and electrical engineering. THE SERIES: FRONTIERS IN COMPUTATIONAL INTELLIGENCE The series Frontiers In Computational Intelligence is envisioned to provide comprehensive coverage and understanding of cutting edge research in computational intelligence. It intends to augment the scholarly discourse on all topics relating to the advances in artifi cial life and machine learning in the form of metaheuristics, approximate reasoning, and robotics. Latest research fi ndings are coupled with applications to varied domains of engineering and computer sciences. This field is steadily growing especially with the advent of novel machine learning algorithms being applied to different domains of engineering and technology. The series brings together leading researchers that intend to continue to advance the fi eld and create a broad knowledge about the most recent state of the art.