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.

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.

Discover how to achieve release-quality mixes even in the smallest studios by applying power-user techniques from the world's most successful producers. Mixing Secrets for the Small Studio is the best-selling primer for small-studio enthusiasts who want chart-ready sonics in a hurry. Drawing on the back-room strategies of more than 160 famous names, this entertaining and down-to-earth guide leads you step-by-step through the entire mixing process. On the way, you'll unravel the mysteries of every type of mix processing, from simple EQ and compression through to advanced spectral dynamics and "fairy dust" effects. User-friendly explanations introduce technical concepts on a strictly need-to-know basis, while chapter summaries and assignments are perfect for school and college use. Learn the subtle editing, arrangement, and monitoring tactics which give industry insiders their competitive edge, and master the psychological tricks which protect you from all the biggest rookie mistakes. Find out where you don't need to spend money, as well as how to make a limited budget really count. Pick up tricks and tips from leading-edge engineers working on today's multi-platinum hits, including Derek "MixedByAli" Ali, Michael Brauer, Dylan "3D" Dresdow, Tom Elmhirst, Serban Ghenea, Jacquire King, the Lord-Alge brothers, Tony Maserati, Manny Marroquin, Noah "50" Shebib, Mark "Spike" Stent, DJ Swivel, Phil Tan, Andy Wallace, Young Guru, and many, many more... Now extensively expanded and updated, including new sections on mix-buss processing, mastering, and the latest advances in plug-in technology.

This book lays out all the latest research in the area of multimedia data hiding. The book introduces multimedia signal processing and information hiding techniques. It includes multimedia representation, digital watermarking fundamentals and requirements of watermarking. It moves on to cover the recent advances in multimedia signal processing, before presenting information hiding techniques including steganography, secret sharing and watermarking. The final part of this book includes practical applications of intelligent multimedia signal processing and data hiding systems.

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.

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.

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.

Practical, concise, and approachable, Audio Engineering 101, Second Edition covers everything aspiring audio engineers need to know to make it in the recording industry, from the characteristics of sound to microphones, analog versus digital recording, EQ/compression, mixing, mastering, and career skills. Filled with hand-ons, step-by-step technique breakdowns and all-new interviews with active professionals, this updated edition includes instruction in using digital consoles, iPads for mixing, audio apps, plug-ins, home studios, and audio for podcasts. An extensive companion website features fifteen new video tutorials, audio clips, equipment lists, quizzes, and student exercises.

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 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."

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.

Metal Music Manual shows you the creative and technical processes involved in producing contemporary heavy music for maximum sonic impact. From pre-production to final mastered product, and fundamental concepts to advanced production techniques, this book contains a world of invaluable practical information. Assisted by clear discussion of critical audio principles and theory, and a comprehensive array of illustrations, photos, and screen grabs, Metal Music Manual is the essential guide to achieving professional production standards. The extensive companion website features multi-track recordings, final mixes, processing examples, audio stems, etc., so you can download the relevant content and experiment with the techniques you read about. The website also features video interviews the author conducted with the following acclaimed producers, who share their expertise, experience, and insight into the processes involved: Fredrik Nordstroem (Dimmu Borgir, At The Gates, In Flames) Matt Hyde (Slayer, Parkway Drive, Children of Bodom) Ross Robinson (Slipknot, Sepultura, Machine Head) Logan Mader (Gojira, DevilDriver, Fear Factory) Andy Sneap (Megadeth, Killswitch Engage, Testament) Jens Bogren (Opeth, Kreator, Arch Enemy) Daniel Bergstrand (Meshuggah, Soilwork, Behemoth) Nick Raskulinecz (Mastodon, Death Angel, Trivium) Quotes from these interviews are featured throughout Metal Music Manual, with additional contributions from: Ross "Drum Doctor" Garfield (one of the world's top drum sound specialists, with Metallica and Slipknot amongst his credits) Andrew Scheps (Black Sabbath, Linkin Park, Metallica) Maor Appelbaum (Sepultura, Faith No More, Halford)

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.

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.

Stochastically-Based Semantic Analysis investigates the problem of automatic natural language understanding in a spoken language dialog system. The focus is on the design of a stochastic parser and its evaluation with respect to a conventional rule-based method. Stochastically-Based Semantic Analysis will be of most interest to researchers in artificial intelligence, especially those in natural language processing, computational linguistics, and speech recognition. It will also appeal to practicing engineers who work in the area of interactive speech systems.

The availability of increased computational power and the proliferation of the Internet have facilitated the production and distribution of unauthorized copies of multimedia information. As a result, the problem of copyright protection has attracted the interest of worldwide scientific and business communities. Signal Processing, Perceptual Coding and Watermarking of Digital Audio: Advanced Technologies and Models focuses on watermarking, in which data is marked with hidden ownership information, as a promising solution to copyright protection issues. Compared to embedding watermarks into still images, hiding data in audio is much more challenging due to the extreme sensitivity of the human auditory system to changes in the audio signal. This book focuses on understanding human perception processes and including them in effective psychoacoustic models, as well as synchronization, which is an important component of a successful watermarking system.

Designing Human Interface in Speech Technology bridges a gap between the needs of the technical engineer and cognitive researchers working in the multidisciplinary area of speech technology applications. The approach is systematic and the focus is on the utility of developing and designing speech related products.

Included is coverage of topics such as neuroscience on the multimodal cortex, cognitive theories on multi-task performance, stress and workload, as well as human information process theory and ecological interface design theory for evaluating speech-related human-system interfaces.

Of special emphasis are topics such as spoken dialogue system design, in-vehicle communication system design and speech technology in military applications. Also included are tools on how to analyze the design, different design theories and process, methods about how to understand users. The material systematically describes the user-center design process and usability evaluation methods.

Designing Human Interface in Speech Technology is appropriate for designers, engineers, and decision makers working in the area of speech technology research. It is also a good text book for senior university students and postgraduate students in the respective interaction design areas.

Mathematical Music offers a concise and easily accessible history of how mathematics was used to create music. The story presented in this short, engaging volume ranges from ratios in antiquity to random combinations in the 17th century, 20th-century statistics, and contemporary artificial intelligence. This book provides a fascinating panorama of the gradual mechanization of thought processes involved in the creation of music. How did Baroque authors envision a composition system based on combinatorics? What was it like to create musical algorithms at the beginning of the 20th century, before the computer became a reality? And how does this all explain today's use of artificial intelligence and machine learning in music? In addition to discussing the history and the present state of mathematical music, Braguinski also takes a look at what possibilities the near future of music AI might hold for listeners, musicians, and the society. Grounded in research findings from musicology and the history of technology, and written for the non-specialist general audience, this book helps both student and professional readers to make sense of today's music AI by situating it in a continuous historical context.

Both modern mathematical music theory and computer science are strongly influenced by the theory of categories and functors. One outcome of this research is the data format of denotators, which is based on set-valued presheaves over the category of modules and diaffine homomorphisms. The functorial approach of denotators deals with generalized points in the form of arrows and allows the construction of a universal concept architecture. This architecture is ideal for handling all aspects of music, especially for the analysis and composition of highly abstract musical works.

This book presents an introduction to the theory of module categories and the theory of denotators, as well as the design of a software system, called Rubato Composer, which is an implementation of the category-theoretic concept framework. The application is written in portable Java and relies on plug-in components, so-called rubettes, which may be combined in data flow networks for the generation and manipulation of denotators.

The Rubato Composer system is open to arbitrary extension and is freely available under the GPL license. It allows the developer to build specialized rubettes for tasks that are of interest to composers, who in turn combine them to create music. It equally serves music theorists, who use them to extract information from and manipulate musical structures. They may even develop new theories by experimenting with the many parameters that are at their disposal thanks to the increased flexibility of the functorial concept architecture.

Two contributed chapters by Guerino Mazzola and Florian Thalmann illustrate the application of the theory as well as the software in the development of compositional tools and the creation of a musical work with the help of the Rubato framework.

Introduction to Digital Music with Python Programming provides a foundation in music and code for the beginner. It shows how coding empowers new forms of creative expression while simplifying and automating many of the tedious aspects of production and composition. With the help of online, interactive examples, this book covers the fundamentals of rhythm, chord structure, and melodic composition alongside the basics of digital production. Each new concept is anchored in a real-world musical example that will have you making beats in a matter of minutes. Music is also a great way to learn core programming concepts such as loops, variables, lists, and functions, Introduction to Digital Music with Python Programming is designed for beginners of all backgrounds, including high school students, undergraduates, and aspiring professionals, and requires no previous experience with music or code.

This thesis discusses the privacy issues in speech-based applications such as biometric authentication, surveillance, and external speech processing services. Author Manas A. Pathak presents solutions for privacy-preserving speech processing applications such as speaker verification, speaker identification and speech recognition. The author also introduces some of the tools from cryptography and machine learning and current techniques for improving the efficiency and scalability of the presented solutions. Experiments with prototype implementations of the solutions for execution time and accuracy on standardized speech datasets are also included in the text. Using the framework proposed may now make it possible for a surveillance agency to listen for a known terrorist without being able to hear conversation from non-targeted, innocent civilians."

This illuminating, engaging book offers an introduction to the art of sound design and postproduction audio, written especially for for directors, producers, sound designers, and teachers without a technical background in sound. Building on over 50 years of combined expertise in teaching, filmmaking, and sound design, experienced instructor and author Peter Rea and sound designer Matthew Polis offer a cogent, clear, and practical overview of sound design principles and practices, from exploring the language and vocabulary of sound to teaching readers how to work with sound professionals, and later to overseeing the edit, mix, and finishing processes. In this book, Rea and Polis focus on creative and practical ways to utilize sound in order to achieve the filmmaker's vision and elevate their films. Balancing practical, experienced-based insight, numerous examples, and unique concepts like storyboarding for sound, A Filmmaker’s Guide to Sound Design arms students, filmmakers, and educators with the knowledge to creatively and confidently navigate their film through the post audio process.

This book provides various speech enhancement algorithms for digital hearing aids. It covers information on noise signals extracted from silences of speech signal. The description of the algorithm used for this purpose is also provided. Different types of adaptive filters such as Least Mean Squares (LMS), Normalized LMS (NLMS) and Recursive Lease Squares (RLS) are described for noise reduction in the speech signals. Different types of noises are taken to generate noisy speech signals, and therefore information on various noises signals is provided. The comparative performance of various adaptive filters for noise reduction in speech signals is also described. In addition, the book provides a speech enhancement technique using adaptive filtering and necessary frequency strength enhancement using wavelet transform as per the requirement of audiogram for digital hearing aids. Presents speech enhancement techniques for improving performance of digital hearing aids; Covers various types of adaptive filters and their advantages and limitations; Provides a hybrid speech enhancement technique using wavelet transform and adaptive filters.