Multimodal Behavioral Analysis in the Wild: Advances and Challenges presents the state-of- the-art in behavioral signal processing using different data modalities, with a special focus on identifying the strengths and limitations of current technologies. The book focuses on audio and video modalities, while also emphasizing emerging modalities, such as accelerometer or proximity data. It covers tasks at different levels of complexity, from low level (speaker detection, sensorimotor links, source separation), through middle level (conversational group detection, addresser and addressee identification), and high level (personality and emotion recognition), providing insights on how to exploit inter-level and intra-level links. This is a valuable resource on the state-of-the- art and future research challenges of multi-modal behavioral analysis in the wild. It is suitable for researchers and graduate students in the fields of computer vision, audio processing, pattern recognition, machine learning and social signal processing.

In the literature of information science, a number of studies have been carried out attempting to model cognitive, affective, behavioral, and contextual factors associated with human information seeking and retrieval. On the other hand, only a few studies have addressed the exploration of creative thinking in music, focusing on understanding and describing individuals' information seeking behavior during the creative process. Trends in Music Information Seeking, Behavior, and Retrieval for Creativity connects theoretical concepts in information seeking and behavior to the music creative process. This publication presents new research, case studies, surveys, and theories related to various aspects of information retrieval and the information seeking behavior of diverse scholarly and professional music communities. Music professionals, theorists, researchers, and students will find this publication an essential resource for their professional and research needs.

Tanja Schultz and Katrin Kirchhoff have compiled a comprehensive overview of speech processing from a multilingual perspective. By taking this all-inclusive approach to speech processing, the editors have included theories, algorithms, and techniques that are required to support spoken input and output in a large variety of languages. This book presents a comprehensive introduction to research problems and solutions, both from a theoretical as well as a practical perspective, and highlights technology that incorporates the increasing necessity for multilingual applications in our global community.
Current challenges of speech processing and the feasibility of sharing data and system components across different languages guide contributors in their discussions of trends, prognoses and open research issues. This includes automatic speech recognition and speech synthesis, but also speech-to-speech translation, dialog systems, automatic language identification, and handling non-native speech. The book is complemented by an overview of multilingual resources, important research trends, and actual speech processing systems that are being deployed in multilingual human-human and human-machine interfaces.
Researchers and developers in industry and academia with different backgrounds but a common interest in multilingual speech processing will find an excellent overview of research problems and solutions detailed from theoretical and practical perspectives.
* State-of-the-art research with a global perspective by authors from the USA, Asia, Europe, and South Africa
* The only comprehensive introduction to multilingual speech processing currently available
* Detailed presentation of technological advances integral to security, financial, cellular and commercial applications

The future of music archiving and search engines lies in deep learning and big data. Music information retrieval algorithms automatically analyze musical features like timbre, melody, rhythm or musical form, and artificial intelligence then sorts and relates these features. At the first International Symposium on Computational Ethnomusicological Archiving held on November 9 to 11, 2017 at the Institute of Systematic Musicology in Hamburg, Germany, a new Computational Phonogram Archiving standard was discussed as an interdisciplinary approach. Ethnomusicologists, music and computer scientists, systematic musicologists as well as music archivists, composers and musicians presented tools, methods and platforms and shared fieldwork and archiving experiences in the fields of musical acoustics, informatics, music theory as well as on music storage, reproduction and metadata. The Computational Phonogram Archiving standard is also in high demand in the music market as a search engine for music consumers. This book offers a comprehensive overview of the field written by leading researchers around the globe.

Learn how to program JavaScript while creating interactive audio applications with JavaScript for Sound Artists: Learn to Code With the Web Audio API! William Turner and Steve Leonard showcase the basics of JavaScript language programing so that readers can learn how to build browser based audio applications, such as music synthesizers and drum machines. The companion website offers further opportunity for growth. Web Audio API instruction includes oscillators, audio file loading and playback, basic audio manipulation, panning and time. This book encompasses all of the basic features of JavaScript with aspects of the Web Audio API to heighten the capability of any browser. Key Features Uses the readers existing knowledge of audio technology to facilitate learning how to program using JavaScript. The teaching will be done through a series of annotated examples and explanations. Downloadable code examples and links to additional reference material included on the books companion website. This book makes learning programming more approachable to nonprofessional programmers The context of teaching JavaScript for the creative audio community in this manner does not exist anywhere else in the market and uses example-based teaching

Learn how to program JavaScript while creating interactive audio applications with JavaScript for Sound Artists: Learn to Code With the Web Audio API! William Turner and Steve Leonard showcase the basics of JavaScript language programing so that readers can learn how to build browser based audio applications, such as music synthesizers and drum machines. The companion website offers further opportunity for growth. Web Audio API instruction includes oscillators, audio file loading and playback, basic audio manipulation, panning and time. This book encompasses all of the basic features of JavaScript with aspects of the Web Audio API to heighten the capability of any browser. Key Features Uses the readers existing knowledge of audio technology to facilitate learning how to program using JavaScript. The teaching will be done through a series of annotated examples and explanations. Downloadable code examples and links to additional reference material included on the books companion website. This book makes learning programming more approachable to nonprofessional programmers The context of teaching JavaScript for the creative audio community in this manner does not exist anywhere else in the market and uses example-based teaching

Provides a comprehensive description and analysis into the use of music information retrieval, from the data management perspective.

This book presents works from world-class experts from academia, industry, and national agencies representing countries from across the world focused on automotive fields for in-vehicle signal processing and safety. These include cutting-edge studies on safety, driver behavior, infrastructure, and human-to-vehicle interfaces. Vehicle Systems, Driver Modeling and Safety is appropriate for researchers, engineers, and professionals working in signal processing for vehicle systems, next generation system design from driver-assisted through fully autonomous vehicles.

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.

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.

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.

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 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.

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.

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.