Making its first huge impact in the 1960s through the inventions of Bob Moog, the analog synthesizer sound, riding a wave of later developments in digital and software synthesis, has now become more popular than ever. Analog Synthesizers charts the technology, instruments, designers, and musicians associated with its three major historical phases: invention in the 1960s-1970s and the music of Walter Carlos, Pink Floyd, Gary Numan, Genesis, Kraftwerk, The Human League, Tangerine Dream, and Jean-Michel Jarre; re-birth in the 1980s-1990s through techno and dance music and jazz fusion; and software synthesis. Now updated, this new edition also includes sections on the explosion from 2000 to the present day in affordable, mass market Eurorack format and other analog instruments, which has helped make the analog synthesizer sound hugely popular once again, particularly in the fields of TV and movie music. Major artists interviewed in depth include: Hans Zimmer (Golden Globe and Academy Award nominee and winner, "Gladiator" and "The Lion King") Mike Oldfield (Grammy Award winner, "Tubular Bells") Isao Tomita (Grammy Award nominee, "Snowflakes Are Dancing") Rick Wakeman (Grammy Award nominee, Yes) Tony Banks (Grammy, Ivor Novello and Brit Awards, Genesis) Nick Rhodes (Grammy Award Winner, Duran Duran) and from the worlds of TV and movie music: Kyle Dixon and Michael Stein (Primetime Emmy Award, "Stranger Things") Paul Haslinger (BMI Film and TV Music Awards, "Underworld") Suzanne Ciani (Grammy Award Nominee, "Neverland") Adam Lastiwka ("Travelers") The book opens with a grounding in the physics of sound, instrument layout, sound creation, purchasing, and instrument repair, which will help entry level musicians as well as seasoned professionals appreciate and master the secrets of analog sound synthesis. Analog Synthesizers has a companion website featuring hundreds of examples of analog sound created using dozens of classic and modern instruments.

This book constitutes the refereed proceedings of the 4th International Conference on Text, Speech and Dialogue, TSD 2001, held in Zelezna Ruda, Czech Republic in September 2001.
The 59 revised papers presented were carefully reviewed and selected from 117 submissions. The book presents a wealth of state-of-the-art research and development results from the field of natural language processing with emphasis on text, speech, and spoken language.

This book is based on the workshop "Information Retrieval Techniques for Speech Applications", held as part of the 24th Annual International ACM SIGIR Conference on Research and Development in Information Retrieval in New Orleans, USA, in September 2001.The book presents 10 papers based on workshop presentations. The topics range from traditional information retrieval techniques over adaptations of these techniques to spoken documents and multimedia collections finally to new applications.

Ken Abbott 's "Voice Enabling Web Applications: VoiceXML and Beyond" is a comprehensive introduction to the concepts, architectures, and implementation techniques underlying the development of voice-enabled Internet applications.

This book is divided into three parts, each of which tackles an essential piece of the voice application development puzzle. In Part One, "Retrospective on Voice and the Web," you'll learn how VoiceXML integrates voice recognition and synthesis technologies with markup languages, and you'll see how VoiceXML (VXML) is a powerful vehicle for incorporating voice and graphical interfaces into today's web architectures. In Part Two, "The VoiceXML Language," you'll be introduced to VXML syntax and programming concepts, and you'll quickly learn how to develop dynamic voice applications by following along with the creation of a voice-enabled personal information manager. You'll also learn about Voice User Interface (VUI) design principles, and you'll gain valuable insight into the techniques used to create efficient, user-friendly voice applications. In Part Three, "Incorporating Voice into the Web," you'll be introduced to the architectures and components used to create large-scale web applications, and you'll learn how to use VoiceXML with other web technologies in a multitier, voice-enabled Web application.

This book contains the collection of papers presented at the Second Workshop on Text, Speech and Dialogue - TSD'99 held in Plzen and Mari ansk eLazn e (Czech Republic) on 13{17 September 1999. The general objective of the workshop was to present state{of{the{art technology and recent achievements in the eld of natural language processing. A total of 57 papers and 19 posters contributed by 128 authors (63 from Central Europe, 11 from Eastern Europe, 33 from Western Europe, 2 from Africa, 13 from America, and 6 from Asia) were included in the workshop proceedings. The workshop is an interdisciplinary forum, which brings together research in speech and language processing as well as research in the Eastern and Western hemisphere. We feel that the mixture of di erent approaches and applications gives all of us a great opportunity to bene t and learn from each other. We would like to gratefully thank the invited speakers and the authors of the papers for their valuable contributions, the Medav GmbH (Uttenreuth, GER) and the SpeechWorks (Boston, USA) for their nancial support, and Prof. V- tracky for greeting the workshop on behalf of the University of West Bohemia.

This original volume describes the Spoken Language Translator (SLT), one of the first major automatic speech translation projects. The SLT system can translate between English, French, and Swedish in the domain of air travel planning, using a vocabulary of about 1500 words, and with an accuracy of about 75%. The authors detail the language processing components, largely built on top of the SRI Core Language Engine, using a combination of general grammars and techniques that allow them to be rapidly customized to specific domains. They base speech recognition on Hidden Markov Mode technology, and use versions of the SRI DECIPHER system. This account of SLT is an essential resource for researchers interested in knowing what is achievable in spoken-language translation today.

This book is based on the author's Ph.D. thesis which was selected during the 1994 ACM Doctoral Dissertation Competition as one of the two co-winning works. T.V. Raman did his Ph.D. work at Cornell University with Professor Davied Gries as thesis advisor.
The author presents the computing system ASTER that audio formats electronic documents to produce audio documents. ASTER can speak both literary texts and highly technical documents containing complex mathematics (presented in (LA)TEX).

Speech Processing, Recognition and Artificial Neural Networks contains papers from leading researchers and selected students, discussing the experiments, theories and perspectives of acoustic phonetics as well as the latest techniques in the field of spe ech science and technology. Topics covered in this book include; Fundamentals of Speech Analysis and Perceptron; Speech Processing; Stochastic Models for Speech; Auditory and Neural Network Models for Speech; Task-Oriented Applications of Automatic Speech Recognition and Synthesis.

The new realities are here. Virtual and Augmented realities and 360 video technologies are rapidly entering our homes and office spaces. Good quality audio has always been important to the user experience, but in the new realities, it is more than important, it's essential. If the audio doesn't work, the immersion of the experience fails and the cracks in the new reality start to show. This practical guide helps you navigate the challenges and pitfalls of designing audio for these new realities. This technology is different from anything we've seen before and requires an entirely new approach; this book will introduce the broad concepts you need to know before delving into the practical detail you need. Key Features This book covers audio for all types of new reality technology. At the moment, VR and 360 video are getting a lot of press, but in a few years we will be hearing a lot more about Augmented and Mixed reality technologies as well. A practical guide to creating, designing and implementing audio for this new technology by a leading sound design and implementation expert. Conceptual explanations address the new approaches necessary to designing effective audio for the new realities. Real-world examples and analysis of what does and does not work including detailed case study discussions.

Designing Interactive Speech Systems describes the design and implementation of spoken language dialogue within the context of SLDS (spoken language dialogue systems) development. Using an applications-oriented SLDS developed through the Danish Dialogue project, the authors describe the complete process involved in designing such a system; and in doing so present several innovative practical tools, such as dialogue design guideline s, in-depth evaluation methodologies, and speech functionality analysis. The approach taken is firmly applications-oriented, describing the results of research applicable to industry and showing how the development of advanced applications drives research rather than the other way around. All those working on the research and development of spoken language services, especially in the area of telecommunications, will benefit from reading this book.

Speech technology, the automatic processing of (spontaneously) spoken language, is now known to be technically feasible. It will become the major tool for handling the confusion of languages with applications including dictation systems, information retrieval by spoken dialog, and speech-to-speech translation. The book gives a throrough account of prosodic phenomena. The author presents in detail the mathematical and comnputational background of the algorithms and statistical models used and develops algorithms enabling the exploitation of prosodic information on various levels of speech understanding, such as syntax, semantics, dialog, and translation. Then he studies the integration of these algorithms in the speech-to-speech translation system VERBMOBIL and in the dialog system EVAR and analyzes the results.

Principles of Game Audio and Sound Design is a comprehensive introduction to the art of sound for games and interactive media using Unity. This accessible guide encompasses both the conceptual challenges of the artform as well as the technical and creative aspects, such as sound design, spatial audio, scripting, implementation and mixing. Beginning with basic techniques, including linear and interactive sound design, before moving on to advanced techniques, such as procedural audio, Principles of Game Audio and Sound Design is supplemented by a host of digital resources, including a library of ready-to-use, adaptable scripts. This thorough introduction provides the reader with the skills and tools to combat the potential challenges of game audio independently. Principles of Game Audio and Sound Design is the perfect primer for beginner- to intermediate-level readers with a basic understanding of audio production and Unity who want to learn how to gain a foothold in the exciting world of game and interactive audio.

This book constitutes the refereed proceedings of the First International Conference on Audio- and Video-based Biometric Person Authentication, AVBPA'97, held in Crans-Montana, Switzerland, in March 1997.
The 49 revised papers presented were carefully reviewed and selected by the program committee for inclusion in the book; also included are four invited contributions. The papers are organized in sections on facial features localisation, lip and facial motion, visual non-face biometrics, face-based authentication, text-dependent speaker authentication, text-independent authentication, audio-video features and fusion, and systems and applications.

This volume provides a comprehensive introduction to foundational topics in sound design for embedded media, such as physical computing; interaction design; auditory displays and data sonification; speech synthesis; wearables; smart objects and instruments; user experience; toys and playful tangible objects; and the new sensibilities entailed in expanding the concept of sound design to encompass the totality of our surroundings. The reader will gain a broad understanding of the key concepts and practices that define sound design for its use in computational products and design. The chapters are written by international authors from diverse backgrounds who provide multidisciplinary perspectives on sound in its many embedded forms. The volume is designed as a textbook for students and teachers, as a handbook for researchers in sound, programming and design, and as a survey of key trends and ideas for practitioners interested in exploring the boundaries of their profession.

This volume collects together refereed versions of twenty-five papers presented at the 4th Neural Computation and Psychology Workshop, held at University College London in April 1997. The "NCPW" workshop series is now well established as a lively forum which brings together researchers from such diverse disciplines as artificial intelligence, mathematics, cognitive science, computer science, neurobiology, philosophy and psychology to discuss their work on connectionist modelling in psychology. The general theme of this fourth workshop in the series was "Connectionist Repre sentations," a topic which not only attracted participants from all these fields, but from allover the world as well. From the point of view of the conference organisers focusing on representational issues had the advantage that it immediately involved researchers from all branches of neural computation. Being so central both to psychology and to connectionist modelling, it is one area about which everyone in the field has their own strong views, and the diversity and quality of the presentations and, just as importantly, the discussion which followed them, certainly attested to this."

This book constitutes the strictly refereed post-workshop documentation of the ECAI'96 Workshop on Dialogue Processing in Spoken Language Systems, held in Budapest, Hungary, in August 1996, during ECAI'96.
The volume presents 16 revised full papers including a detailed introduction and survey paper by the volume editors. The papers are organized in sections on foundations of spoken language dialogue systems, dialogue systems and prosodic aspects of spoken dialogue processing, spoken dialogue systems-design and implementation, and evaluation of systems. The book reports on work being pursued both in academia and in industry as a crucial issue in speech processing.

A new generation of speech-driven personal computer systems promises to transform the business use of Information Technology. This is not merely a matter of discarding the keyboard, but of rethinking business processes to take advantage of the increased productivity that speech-driven systems can bring.
Malcolm McPherson is one of the pioneers of this fast-moving field, and has been personally involved in the development of systems that have met business needs across many industrial sectors.

The digital turn has created new opportunities for scholars across disciplines to use sound in their scholarship. This volume's contributors provide a blueprint for making sound central to research, teaching, and dissemination. They show how digital sound studies has the potential to transform silent, text-centric cultures of communication in the humanities into rich, multisensory experiences that are more inclusive of diverse knowledges and abilities. Drawing on multiple disciplines-including rhetoric and composition, performance studies, anthropology, history, and information science-the contributors to Digital Sound Studies bring digital humanities and sound studies into productive conversation while probing the assumptions behind the use of digital tools and technologies in academic life. In so doing, they explore how sonic experience might transform our scholarly networks, writing processes, research methodologies, pedagogies, and knowledges of the archive. As they demonstrate, incorporating sound into scholarship is thus not only feasible but urgently necessary. Contributors. Myron M. Beasley, Regina N. Bradley, Steph Ceraso, Tanya Clement, Rebecca Dowd Geoffroy-Schwinden, W. F. Umi Hsu, Michael J. Kramer, Mary Caton Lingold, Darren Mueller, Richard Cullen Rath, Liana M. Silva, Jonathan Sterne, Jennifer Stoever, Jonathan W. Stone, Joanna Swafford, Aaron Trammell, Whitney Trettien

Traditionally, the European-based biannual international conference "EUROSPEECH" dealing with all aspects of speech science and technology is preceded by an "ESPRIT Speech Projects Days," which presents a particularly well timed opportunity to measure progress in speech technology and ap plications in Europe. The last venue was held in Berlin, Germany, on September 20th, 1993. The success of this workshop encouraged the major European experts in the field to contribute to this volume. Published in the ESPRIT Research Report series, it presents the results of advanced European research on speech technologies and its applications in the multilingual framework of the European Union. Speech is an important factor in building an integrated European communication platform. Strong links exist between speech and natural language processing, and human computer interaction. Recent experimental results on multilingual conversion between both speech and text show the advantage of integrating phonetic, lexical, and syntactic knowledge, and also demonstrate the feasibility of multilingual voice systems in the human-computer interface applications. Multilingual queries use natural language-based co-operative dialogue as an interface to the computer services in the information applications. Continuous and robust speech understanding is here addressed for both speaker-independent and speaker-adaptive processing, together with dialogue modelling and manage ment. Such technologies are then used in the design of computer workstations with a speech-based human interface for a large range variety of information technology applications (e.g. in the office, telecommunications, and computer aided education)."

Learn Audio Electronics with Arduino: Practical Audio Circuits with Arduino Control teaches the reader how to use Arduino to control analogue audio circuits and introduces electronic circuit theory through a series of practical projects, including a MIDI drum controller and an Arduino-controlled two-band audio equalizer amplifier. Learn Audio Electronics with Arduino provides all the theoretical knowledge needed to design, analyse, and build audio circuits for amplification and filtering, with additional topics like C programming being introduced in a practical context for Arduino control. The reader will learn how these circuits work and also how to build them, allowing them to progress to more advanced audio circuits in the future. Beginning with electrical fundamentals and control systems, DC circuit theory is then combined with an introduction to C programming to build Arduino-based systems for audio (tone sequencer) and MIDI (drum controller) output. The second half of the book begins with AC circuit theory to allow analogue audio circuits for amplification and filtering to be analysed, simulated, and built. These circuits are then combined with Arduino control in the final project - an Arduino-controlled two-band equalizer amplifier. Building on high-school physics and mathematics in an accessible way, Learn Audio Electronics with Arduino is suitable for readers of all levels. An ideal tool for those studying audio electronics, including as a component within other fields of study, such as computer science, human-computer interaction, acoustics, music technology, and electronics engineering.

This volume comprises a collection of papers presented at the Workshop on Information Protection, held in Moscow, Russia in December 1993. The 16 thoroughly refereed papers by internationally known scientists selected for this volume offer an exciting perspective on error control coding, cryptology, and speech compression. In the former Soviet Union, research related to information protection was often shielded from the international scientific community. Therefore, the results presented by Russian researchers and engineers at this first international workshop on this topic are of particular interest; their work defines the cutting edge of research in many areas of error control, cryptology, and speech recognition.

Record, arrange, mix, produce, and polish your audio files with this best-selling, Apple-certified guide to Logic Pro X 10.4. Veteran producer and composer David Nahmani uses step-bystep, project-based instructions and straightforward explanations to teach everything from basic music creation to sophisticated production techniques. Using the book's downloadable lesson files and Logic Pro X, you'll begin making music in the first lesson. From there, learn to record audio and MIDI data, create and edit sequences, and master mixing and automation techniques such as submixing with track stacks. Create both acoustic and electronic virtual drum performances using Drummer tracks with Drum Kit Designer and Drum Machine Designer. Use Logic Pro X MIDI FX and Smart Controls to control software synthesizers from a MIDI controller or an iPad. Harness the power of Smart Tempo to make sure all recordings, imported audio files, and samples play in time. Flex Time allows you to precisely edit the timing of notes inside an audio recording, and you'll explore Flex Pitch to correct the pitch of a vocal recording. Finally, you mix, automate, and master the song, using plug-ins to process only selected sections or entire tracks, giving your audio creations the final polish needed to achieve a professional sound. Downloadable lesson and media files allow you to perform the hands-on exercises. Focused lessons take you step by step through practical, real-world tasks. Accessible writing style puts an expert instructor at your side Ample illustrations help you master techniques fast. Lesson goals and time estimates help you plan your time. Chapter review questions summarize what you've learned and help you prepare for the Apple certification exam.

Innovation in Music: Performance, Production, Technology and Business is an exciting collection comprising of cutting-edge articles on a range of topics, presented under the main themes of artistry, technology, production and industry. Each chapter is written by a leader in the field and contains insights and discoveries not yet shared. Innovation in Music covers new developments in standard practice of sound design, engineering and acoustics. It also reaches into areas of innovation, both in technology and business practice, even into cross-discipline areas. This book is the perfect companion for professionals and researchers alike with an interest in the Music industry. Chapter 31 of this book is freely available as a downloadable Open Access PDF under a Creative Commons Attribution-Non Commercial-No Derivatives 4.0 license. https://tandfbis.s3-us-west-2.amazonaws.com/rt-files/docs/Open+Access+Chapters/9781138498211_oachapter31.pdf

This book is intended to give an overview of the major results achieved in the field of natural speech understanding inside ESPRIT Project P. 26, "Advanced Algorithms and Architectures for Speech and Image Processing." The project began as a Pilot Project in the early stage of Phase 1 of the ESPRIT Program launched by the Commission of the European Communities. After one year, in the light of the preliminary results that were obtained, it was confirmed for its 5-year duration. Even though the activities were carried out for both speech and image understand ing we preferred to focus the treatment of the book on the first area which crystallized mainly around the CSELT team, with the valuable cooperation of AEG, Thomson-CSF, and Politecnico di Torino. Due to the work of the five years of the project, the Consortium was able to develop an actual and complete understanding system that goes from a continuously spoken natural language sentence to its meaning and the consequent access to a database. When we started in 1983 we had some expertise in small-vocabulary syntax-driven connected-word speech recognition using Hidden Markov Models, in written natural lan guage understanding, and in hardware design mainly based upon bit-slice microprocessors."

During the last two decades, the field of music production has attracted considerable interest from the academic community, more recently becoming established as an important and flourishing research discipline in its own right. Producing Music presents cutting-edge research across topics that both strengthen and broaden the range of the discipline as it currently stands. Bringing together the academic study of music production and practical techniques, this book illustrates the latest research on producing music. Focusing on areas such as genre, technology, concepts, and contexts of production, Hepworth-Sawyer, Hodgson, and Marrington have compiled key research from practitioners and academics to present a comprehensive view of how music production has established itself and changed over the years.