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.

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.

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.

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.

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.

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.

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 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.

Classical Recording: A Practical Guide in the Decca Tradition is the authoritative guide to all aspects of recording acoustic classical music. Offering detailed descriptions, diagrams, and photographs of fundamental recording techniques such as the Decca tree, this book offers a comprehensive overview of the essential skills involved in successfully producing a classical recording. Written by engineers with years of experience working for Decca and Abbey Road Studios and as freelancers, Classical Recording equips the student, the interested amateur, and the practising professional with the required knowledge and confidence to tackle everything from solo piano to opera.

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.

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.

Spoken Dialogue Systems Technology and Design covers key topics in the field of spoken language dialogue interaction from a variety of leading researchers. It brings together several perspectives in the areas of corpus annotation and analysis, dialogue system construction, as well as theoretical perspectives on communicative intention, context-based generation, and modelling of discourse structure. These topics are all part of the general research and development within the area of discourse and dialogue with an emphasis on dialogue systems; corpora and corpus tools and semantic and pragmatic modelling of discourse and dialogue.

Based on a NATO Advanced Study Institute held in 1993, this book addresses recent advances in automatic speech recognition and speech coding. The book contains contributions by many of the most outstanding researchers from the best laboratories worldwide in the field. The contributions have been grouped into five parts: on acoustic modeling; language modeling; speech processing, analysis and synthesis; speech coding; and vector quantization and neural nets. For each of these topics, some of the best-known researchers were invited to give a lecture. In addition to these lectures, the topics were complemented with discussions and presentations of the work of those attending. Altogether, the reader is given a wide perspective on recent advances in the field and will be able to see the trends for future work.

A comprehensive reference on the exciting growth area of spoken dialogs with computers, this text describes the components of a computer-based spoken dialog system, and will prove invaluable to researchers in industry and academia working on speech communication systems and for applications developers. This state-of-the-art book reviews the complete chain from microphone to speech synthesis. It provides methods, models, and algorithms for building a working system. Renato De Mori is coauthor of each chapter ensuring coherence and homogeneity throughout the text.
Spoken Dialogs with Computers covers in detail: transducers and microphone arrays, speech analysis and transformation, acoustic modeling and model training, language modeling, and knowledge integration for automatic speech recognition (ASR). The book also presents generation of word hypotheses, speaker adaptation, robustness and telephone application, use of syntactic and semantic knowledge, speech interpretation and dialog strategies, speech generation, and software system architectures for practical implementation.
Key Features
* All the necessary methods and models are provided for building a working systems and there is clear algorithmic presentation of the important components
* A section on automatic interpretation allows the building of a database query system in spoken language
* The book will be invaluable to researchers in industry and academia working on speech communication systems and for application developers in industry

This book provides the first comprehensive overview of the fascinating topic of audio source separation based on non-negative matrix factorization, deep neural networks, and sparse component analysis. The first section of the book covers single channel source separation based on non-negative matrix factorization (NMF). After an introduction to the technique, two further chapters describe separation of known sources using non-negative spectrogram factorization, and temporal NMF models. In section two, NMF methods are extended to multi-channel source separation. Section three introduces deep neural network (DNN) techniques, with chapters on multichannel and single channel separation, and a further chapter on DNN based mask estimation for monaural speech separation. In section four, sparse component analysis (SCA) is discussed, with chapters on source separation using audio directional statistics modelling, multi-microphone MMSE-based techniques and diffusion map methods. The book brings together leading researchers to provide tutorial-like and in-depth treatments on major audio source separation topics, with the objective of becoming the definitive source for a comprehensive, authoritative, and accessible treatment. This book is written for graduate students and researchers who are interested in audio source separation techniques based on NMF, DNN and SCA.

Audio Mastering: The Artists collects more than twenty interviews, drawn from more than 60 hours of discussions, with many of the world's leading mastering engineers. In these exclusive and often intimate interviews, engineers consider the audio mastering process as they, themselves, experience and shape it as the leading artists in their field. Each interview covers how engineers got started in the recording industry, what prompted them to pursue mastering, how they learned about the process, which tools and techniques they routinely use when they work, and a host of other particulars of their crafts. We also spoke with mix engineers, and craftsmen responsible for some of the more iconic mastering tools now on the market, to gain a broader perspective on their work. This book is the first to provide such a comprehensive overview of the audio mastering process told from the point-of-view of the artists who engage in it. In so doing, it pulls the curtain back on a crucial, but seldom heard from, agency in record production at large.

This text is the first published survey of recent research in signal processing for music transcription, edited and authored by authorities in the field. It covers a range of topics, from the structure and decomposition of signals, pitch and multipitch estimation, coding methods for sound separation, automatic sound source identification and sequence transcription, to using computational modeling and neural networks for music transcription. The book targets a growing audience interested in MPEG-7 standardization. It is a reference for researchers and students in signal processing, computer science, acoustics and music.

This book is a revised version of my doctoral thesis which was submitted in April 1993. The main extension is a chapter on evaluation of the system de scribed in Chapter 8 as this is clearly an issue which was not treated in the original version. This required the collection of data, the development of a concept for diagnostic evaluation of linguistic word recognition systems and, of course, the actual evaluation of the system itself. The revisions made primarily concern the presentation of the latest version of the SILPA system described in an additional Subsection 8. 3, the development environment for SILPA in Sec tion 8. 4, the diagnostic evaluation of the system as an additional Chapter 9. Some updates are included in the discussion of phonology and computation in Chapter 2 and finite state techniques in computational phonology in Chapter 3. The thesis was designed primarily as a contribution to the area of compu tational phonology. However, it addresses issues which are relevant within the disciplines of general linguistics, computational linguistics and, in particular, speech technology, in providing a detailed declarative, computationally inter preted linguistic model for application in spoken language processing. Time Map Phonology is a novel, constraint-based approach based on a two-stage temporal interpretation of phonological categories as events."

Auditory User Interfaces: Toward the Speaking Computer describes a speech-enabling approach that separates computation from the user interface and integrates speech into the human-computer interaction. The Auditory User Interface (AUI) works directly with the computational core of the application, the same as the Graphical User Interface. The author's approach is implemented in two large systems, ASTER - a computing system that produces high-quality interactive aural renderings of electronic documents - and Emacspeak - a fully-fledged speech interface to workstations, including fluent spoken access to the World Wide Web and many desktop applications. Using this approach, developers can design new high-quality AUIs. Auditory interfaces are presented using concrete examples that have been implemented on an electronic desktop. This aural desktop system enables applications to produce auditory output using the same information used for conventional visual output. Auditory User Interfaces: Toward the Speaking Computer is for the electrical and computer engineering professional in the field of computer/human interface design. It will also be of interest to academic and industrial researchers, and engineers designing and implementing computer systems that speak. Communication devices such as hand-held computers, smart telephones, talking web browsers, and others will need to incorporate speech-enabling interfaces to be effective.

This book provides a survey of the state-of-the-art in the practical implementation of Spoken Dialog Systems for applications in everyday settings. It includes contributions on key topics in situated dialog interaction from a number of leading researchers and offers a broad spectrum of perspectives on research and development in the area. In particular, it presents applications in robotics, knowledge access and communication and covers the following topics: dialog for interacting with robots; language understanding and generation; dialog architectures and modeling; core technologies; and the analysis of human discourse and interaction. The contributions are adapted and expanded contributions from the 2014 International Workshop on Spoken Dialog Systems (IWSDS 2014), where researchers and developers from industry and academia alike met to discuss and compare their implementation experiences, analyses and empirical findings.

Robust Speech Recognition in Embedded Systems and PC Applications provides a link between the technology and the application worlds. As speech recognition technology is now good enough for a number of applications and the core technology is well established around hidden Markov models many of the differences between systems found in the field are related to implementation variants. We distinguish between embedded systems and PC-based applications. Embedded applications are usually cost sensitive and require very simple and optimized methods to be viable. Robust Speech Recognition in Embedded Systems and PC Applications reviews the problems of robust speech recognition, summarizes the current state of the art of robust speech recognition while providing some perspectives, and goes over the complementary technologies that are necessary to build an application, such as dialog and user interface technologies. Robust Speech Recognition in Embedded Systems and PC Applications is divided into five chapters. The first one reviews the main difficulties encountered in automatic speech recognition when the type of communication is unknown. The second chapter focuses on environment-independent/adaptive speech recognition approaches and on the mainstream methods applicable to noise robust speech recognition. The third chapter discusses several critical technologies that contribute to making an application usable. It also provides some design recommendations on how to design prompts, generate user feedback and develop speech user interfaces. The fourth chapter reviews several techniques that are particularly useful for embedded systems or to decrease computational complexity. It also presents some case studies for embedded applications and PC-based systems. Finally, the fifth chapter provides a future outlook for robust speech recognition, emphasizing the areas that the author sees as the most promising for the future. Robust Speech Recognition in Embedded Systems and PC Applications serves as a valuable reference and although not intended as a formal University textbook, contains some material that can be used for a course at the graduate or undergraduate level. It is a good complement for the book entitled Robustness in Automatic Speech Recognition: Fundamentals and Applications co-authored by the same author.

The mathematical theory of counterpoint was originally aimed at simulating the composition rules described in Johann Joseph Fux's Gradus ad Parnassum. It soon became apparent that the algebraic apparatus used in this model could also serve to define entirely new systems of rules for composition, generated by new choices of consonances and dissonances, which in turn lead to new restrictions governing the succession of intervals. This is the first book bringing together recent developments and perspectives on mathematical counterpoint theory in detail. The authors include recent theoretical results on counterpoint worlds, the extension of counterpoint to microtonal pitch systems, the singular homology of counterpoint models, and the software implementation of contrapuntal models. The book is suitable for graduates and researchers. A good command of algebra is a prerequisite for understanding the construction of the model.