With Computational Thinking in Sound, veteran educators Gena R. Greher and Jesse M. Heines provide the first book ever written for music fundamentals educators which is devoted specifically to music, sound, and technology. The authors demonstrate how the range of mental tools in computer science - for example, analytical thought, system design, and problem design and solution - can be fruitfully applied to music education, including examples of successful student work. While technology instruction in music education has traditionally focused on teaching how computers and software work to produce music, Greher and Heines offer context: a clear understanding of how music technology can be structured around a set of learning challenges and tasks of the type common in computer science classrooms. Using a learner-centered approach that emphasizes project-based experiences, the book provides music educators with multiple strategies to explore, create, and solve problems with music and technology in equal parts. It also provides examples of hands-on activities which encourage students, alone and in interdisciplinary groups, to explore the basic principles that underlie today's music technology and which expose them to current multimedia development tools.

"Emotion Recognition Using Speech Features" provides coverage of emotion-specific features present in speech. The author also discusses suitable models for capturing emotion-specific information for distinguishing different emotions. The content of this book is important for designing and developing natural and sophisticated speech systems. In this Brief, Drs. Rao and Koolagudi lead a discussion of how emotion-specific information is embedded in speech and how to acquire emotion-specific knowledge using appropriate statistical models. Additionally, the authors provide information about exploiting multiple evidences derived from various features and models. The acquired emotion-specific knowledge is useful for synthesizing emotions. Features includes discussion of: * Global and local prosodic features at syllable, word and phrase levels, helpful for capturing emotion-discriminative information; * Exploiting complementary evidences obtained from excitation sources, vocal tract systems and prosodic features in order to enhance the emotion recognition performance; * Proposed multi-stage and hybrid models for improving the emotion recognition performance. This brief is for researchers working in areas related to speech-based products such as mobile phone manufacturing companies, automobile companies, and entertainment products as well as researchers involved in basic and applied speech processing research.

This volume constitutes the refereed proceedings of the Spanish Conference, IberSPEECH 2012: Joint VII "Jornadas en Tecnologia del Habla" and III Iberian SLTech Workshop, held in Madrid, Spain, in November 21-23, 2012. The 29 revised papers were carefully reviewed and selected from 80 submissions. The papers are organized in topical sections on speaker characterization and recognition; audio and speech segmentation; pathology detection and speech characterization; dialogue and multimodal systems; robustness in automatic speech recognition; applications of speech and language technologies.

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.

Refining Sound is a practical roadmap to the complexities of creating sounds on modern synthesizers. As author, veteran synthesizer instructor Brian K. Shepard draws on his years of experience in synthesizer pedagogy in order to peel back the often-mysterious layers of sound synthesis one-by-one. The result is a book which allows readers to familiarize themselves with each individual step in the synthesis process, in turn empowering them in their own creative or experimental work. The book follows the stages of synthesis in chronological progression, starting readers at the raw materials of sound creation and ultimately bringing them to the final "polishing" stage. Each chapter focuses on a particular aspect of the synthesis process, culminating in a last chapter that brings everything together as the reader creates his/her own complex sounds. Throughout the text, the material is supported by copious examples and illustrations as well as by audio files and synthesis demonstrations on a related companion website. Each chapter contains easily digestible guided projects (entitled "Your Turn" sections) that focus on the topics of the corresponding chapter. In addition to this, one complete project will be carried through each chapter of the book cumulatively, allowing the reader to follow - and build - a sound from start to finish. The final chapter includes several sound creation projects in which readers are given types of sound to create as well as some suggestions and tips, with final outcomes is left to readers' own creativity. Perhaps the most difficult aspect of learning to create sounds on a synthesizer is to understand exactly what each synthesizer component does independent of the synthesizer's numerous other components. Not only does this book thoroughly illustrate and explain these individual components, but it also offers numerous practical demonstrations and exercises that allow the reader to experiment with and understand these elements without the distraction of the other controls and modifiers. Refining Sound is essential for all electronic musicians from amateur to professional levels of accomplishment, students, teachers, libraries, and anyone interested in creating sounds on a synthesizer.

"Automatic Speech Signal Analysis for Clinical Diagnosis and Assessment of Speech Disorders "provides a survey of methods designed to aid clinicians in the diagnosis and monitoring of speech disorders such as dysarthria and dyspraxia, with an emphasis on the signal processing techniques, statistical validity of the results presented in the literature, and the appropriateness of methodsthat do not requirespecialized equipment, rigorously controlled recording procedures or highly skilled personnel to interpret results.
Such techniques offer the promise of a simple and cost-effective, yet objective, assessment of a range of medical conditions, which would be of great value to clinicians. The ideal scenario would begin with the collection of examples of the clients' speech, either over the phone or using portable recording devices operated by non-specialist nursing staff.
The recordings could then be analyzed initially to aid diagnosis of conditions, and subsequently to monitor the clients' progress and response to treatment. The automation of this process would allow more frequent and regular assessments to be performed, as well as providing greater objectivity."

Modern communication devices, such as mobile phones, teleconferencing systems, VoIP, etc., are often used in noisy and reverberant environments. Therefore, signals picked up by the microphones from telecommunication devices contain not only the desired near-end speech signal, but also interferences such as the background noise, far-end echoes produced by the loudspeaker, and reverberations of the desired source. These interferences degrade the fidelity and intelligibility of the near-end speech in human-to-human telecommunications and decrease the performance of human-to-machine interfaces (i.e., automatic speech recognition systems).

The proposed book deals with the fundamental challenges of speech processing in modern communication, including speech enhancement, interference suppression, acoustic echo cancellation, relative transfer function identification, source localization, dereverberation, and beamforming in reverberant environments.

Enhancement of speech signals is necessary whenever the source signal is corrupted by noise. In highly non-stationary noise environments, noise transients, and interferences may be extremely annoying. Acoustic echo cancellation is used to eliminate the acoustic coupling between the loudspeaker and the microphone of a communication device. Identification of the relative transfer function between sensors in response to a desired speech signal enables to derive a reference noise signal for suppressing directional or coherent noise sources. Source localization, dereverberation, and beamforming in reverberant environments further enable to increase the intelligibility of the near-end speech signal.

This book constitutes the refereed proceedings of the 16th International Conference on Text, Speech and Dialogue, TSD 2013, held in Pilsen, Czech Republic, in September 2013. The 65 papers presented together with 5 invited talks were carefully reviewed and selected from 148 submissions. The main topics of this year's conference was corpora, texts and transcription, speech analysis, recognition and synthesis, and their intertwining within NL dialogue systems. The topics also included speech recognition, corpora and language resources, speech and spoken language generation, tagging, classification and parsing of text and speech, semantic processing of text and speech, integrating applications of text and speech processing, as well as automatic dialogue systems, and multimodal techniques and modelling.

Database of Piano Chords: An Engineering View of Harmony includes a unique database of piano chords developed exclusively for music research purposes, and outlines the key advantages to using this dataset to further one's research. The book also describes the physical bases of the occidental music chords and the influence used in the detection and transcription of the music, enabling researchers to intimately understand the construction of each occidental chord. The online database contains more than 275,000 chords with different degrees of polyphony and with different playing styles. Together, the database and the book are an invaluable tool for researchers in this field.

Audio Signal Processing for Next-Generation Multimedia Communication Systems presents cutting-edge digital signal processing theory and implementation techniques for problems including speech acquisition and enhancement using microphone arrays, new adaptive filtering algorithms, multichannel acoustic echo cancellation, sound source tracking and separation, audio coding, and realistic sound stage reproduction. This book's focus is almost exclusively on the processing, transmission, and presentation of audio and acoustic signals in multimedia communications for telecollaboration where immersive acoustics will play a great role in the near future.

Auralization is the creation of audible acoustic sceneries from computer-generated data. The term "auralization" is to be understood as being analogue to the well-known technique of "visualization." In visual illustration of scenes, data or any other meaningful information, in movie animation and in computer graphics, we describe the process of "making visible" as visualization. In acoustics, auralization is taking place when acoustic effects, primary sound signals or means of sound reinforcement or sound transmission, are processed to be presented by using electro-acoustic equipment. This book is organized as comprehensive collection of basics, methodology and strategies of acoustic simulation and auralization. With mathematical background of advanced students the reader will be able to follow the main strategy of auralization easily and work own implementations of auralization in various fields of applications in acoustic engineering, sound design and virtual reality. For readers interested in basic research the technique of auralization may be useful to create sound stimuli for specific investigations in linguistic, medical, neurological and psychological research and in the field of human-machine interaction.

New synths with unique features and layers of complexity are released frequently, with hundreds of different synths currently available in the marketplace. How do you know which ones to use and how do you get the most out of the ones you already own? The Musical Art of Synthesis　presents synthesizer programming　with a　specific focus on synthesis as a musical tool.　Through its innovative design, this title offers an applied approach by providing a breakdown of synthesis methods by type, the inclusion of step-by-step patch recipes, and extensive web-based media content including tutorials, demonstrations, and additional background information. Sam McGuire and Nathan van der Rest guide you to master synthesis and transcend the technical aspects as a musician and artist.

　

Synths are presented using a multi-tiered system beginning with basic instructions for all common synth techniques. Historical information is included for each type of synth, which is designed to help you understand how each instrument relates to the bigger picture. Advanced level instruction focuses on modern implementations and on mobile devices, with special focus on performing and practical usage.　The goal The Musical Art of Synthesis is to bring all of the different types of together in the same discussion and encourage you to see the similarities and differences that force you to gain a better overall understanding of the synthesis process.

Key features of this title:

•　This book will teach you how to put synthesizers to use with easy-to-use synth patch recipes

•　Using a unique, multi-tiered approach applicable to the level of equipment in use, this publication introduces concepts that apply to a wide range of hardware/software synthesizers.

•　A robust companion website, featuring video demonstrations by synthesizer experts, further supports the book: www.focalpress.com/cw/mcguire

Table of Contents

Introduction

Subtractive Synthesis

Sampling and Sample Based Synthesis

Modulation Synthesis

Additive Synthesis

Wavetable Synthesis

Granular Synthesis

Vector Synthesis

Combination Synthesis and Modelling

Performing Live

Glossary

Appendices

This book constitutes the proceedings of the First Indo-Japanese conference on Perception and Machine Intelligence, PerMIn 2012, held in Kolkata, India, in January 2012. The 41 papers, presented together with 1 keynote paper and 3 plenary papers, were carefully reviewed and selected for inclusion in the book. The papers are organized in topical sections named perception; human-computer interaction; e-nose and e-tongue; machine intelligence and application; image and video processing; and speech and signal processing.

The author covers the fundamentals of both information and communication security including current developments in some of the most critical areas of automatic speech recognition. Included are topics on speech watermarking, speech encryption, steganography, multilevel security systems comprising speaker identification, real transmission of watermarked or encrypted speech signals, and more. The book is especially useful for information security specialist, government security analysts, speech development professionals, and for individuals involved in the study and research of speech recognition at advanced levels.

In Mixing with Impact: Learning to Make Musical Choices, Wessel Oltheten discusses the creative and technical concepts behind making a mix. Whether you're a dance producer in your home studio, a live mixer in a club, or an engineer in a big studio, the mindset is largely the same. The same goes for the questions you run into: where do you start? How do you deal with a context in which all the different parts affect each other? How do you avoid getting lost in technique? How do you direct your audience's attention? Why doesn't your mix sound as good as someone else's? How do you maintain your objectivity when you hear the same song a hundred times? How do your speakers affect your perception? What's the difference between one compressor and another? Following a clear structure, this book covers these and many other questions, bringing you closer and closer to answering the most important question of all: how do you tell a story with sound?

Sure, you can import and mix a track in Pro Tools. You can work with MIDI and you know your way around the Edit window. The UI is as familiar as your most broken-in pair of jeans. We get it you don t need another button-pushing guide starting from the ground floor.

Get uniquely in-depth coverage instead with In the Box Music Production: Advanced Tools and Techniques for Pro Tools. Author Mike Collins splits the book into three distinct sections covering how you use Pro Tools now whether you re working with the synths and samplers or loops and beats of a dance or hip-hop project, the soaring vocals of the next pop sensation, or the lush layers of an instrumental world music track. Use Pro Tools to its full potential with advice on studio techniques and full exploration of its internal capabilities. Learn to leverage Pro Tools and make it work for you with this guide that is fully grounded in real-world applications and process.

This book assumes that the user has some music production experience and has worked through the basics in Pro Tools. "

Automated Speaking Assessment: Using Language Technologies to Score Spontaneous Speech provides a thorough overview of state-of-the-art automated speech scoring technology as it is currently used at Educational Testing Service (ETS). Its main focus is related to the automated scoring of spontaneous speech elicited by TOEFL iBT Speaking section items, but other applications of speech scoring, such as for more predictable spoken responses or responses provided in a dialogic setting, are also discussed. The book begins with an in-depth overview of the nascent field of automated speech scoring-its history, applications, and challenges-followed by a discussion of psychometric considerations for automated speech scoring. The second and third parts discuss the integral main components of an automated speech scoring system as well as the different types of automatically generated measures extracted by the system features related to evaluate the speaking construct of communicative competence as measured defined by the TOEFL iBT Speaking assessment. Finally, the last part of the book touches on more recent developments, such as providing more detailed feedback on test takers' spoken responses using speech features and scoring of dialogic speech. It concludes with a discussion, summary, and outlook on future developments in this area. Written with minimal technical details for the benefit of non-experts, this book is an ideal resource for graduate students in courses on Language Testing and Assessment as well as teachers and researchers in applied linguistics.

The advances in computing and networking have sparked an enormous interest in deploying automatic speech recognition on mobile devices and over communication networks. This book brings together academic researchers and industrial practitioners to address the issues in this emerging realm and presents the reader with a comprehensive introduction to the subject of speech recognition in devices and networks. It covers network, distributed and embedded speech recognition systems.

This book constitutes the refereed proceedings of the 4th International Workshop on Haptic and Audio Interaction Design, HAID 2009 held in Dresden, Germany in September 2009.

The 17 revised full papers presented were carefully reviewed and selected for inclusion in the book. The papers are organized in topical sections on haptic communication and perception, navigation and guidance, visual impairment, vibrotactile feedback and music, multimodal user interfaces: design and evaluation, and multimodal gaming.

Developing Virtual Synthesizers with VCV Rack takes the reader step by step through the process of developing synthesizer modules, beginning with the elementary and leading up to more engaging examples. Using the intuitive VCV Rack and its open-source C++ API, this book will guide even the most inexperienced reader to master efficient DSP coding to create oscillators, filters, and complex modules. Examining practical topics related to releasing plugins and managing complex graphical user interaction, with an intuitive study of signal processing theory specifically tailored for sound synthesis and virtual analog, this book covers everything from theory to practice. With exercises and example patches in each chapter, the reader will build a library of synthesizer modules that they can modify and expand. Supplemented by a companion website, this book is recommended reading for undergraduate and postgraduate students of audio engineering, music technology, computer science, electronics, and related courses; audio coding and do-it-yourself enthusiasts; and professionals looking for a quick guide to VCV Rack. VCV Rack is a free and open-source software available online.

This volume contains the proceedings of NOLISP 2009, an ISCA Tutorial and Workshop on Non-Linear Speech Processing held at the University of Vic (- talonia, Spain) during June 25-27, 2009. NOLISP2009wasprecededbythreeeditionsofthisbiannualeventheld2003 in Le Croisic (France), 2005 in Barcelona, and 2007 in Paris. The main idea of NOLISP workshops is to present and discuss new ideas, techniques and results related to alternative approaches in speech processing that may depart from the mainstream. In order to work at the front-end of the subject area, the following domains of interest have been de?ned for NOLISP 2009: 1. Non-linear approximation and estimation 2. Non-linear oscillators and predictors 3. Higher-order statistics 4. Independent component analysis 5. Nearest neighbors 6. Neural networks 7. Decision trees 8. Non-parametric models 9. Dynamics for non-linear systems 10. Fractal methods 11. Chaos modeling 12. Non-linear di?erential equations The initiative to organize NOLISP 2009 at the University of Vic (UVic) came from the UVic Research Group on Signal Processing and was supported by the Hardware-Software Research Group. We would like to acknowledge the ?nancial support obtained from the M- istry of Science and Innovation of Spain (MICINN), University of Vic, ISCA, and EURASIP. All contributions to this volume are original. They were subject to a doub- blind refereeing procedure before their acceptance for the workshop and were revised after being presented at NOLISP 2009.

This book guides nonfiction storytellers in the art of creatively and strategically using sound to engage their audience and bring stories to life. Sound is half of film and video storytelling, and yet its importance is often overlooked until a post-production emergency arises. Written by two experienced creators-one a seasoned nonfiction producer/director with a background in music, and one a sound designer who owns a well-regarded mix studio-this book teaches nonfiction producers, filmmakers, and branded content creators how to reimagine their storytelling by improving sound workflow from field to post. In addition to real-world examples from the authors' own experiences, interviews with and examples from industry professionals across many genres of nonfiction production are included throughout. Written in a conversational style, the book pinpoints practical topics and considerations like 360 video and viewer accessibility. As such, it is a vital point of reference for all nonfiction filmmakers, directors, and producers, or anyone wanting to learn how to improve their storytelling. An accompanying Companion Website offers listening exercises, production sound layout diagrams, templates, and other resources.

Recording Classical Music presents the fundamental principles of digitally recording and editing acoustic music in ambient spaces, focusing on stereo microphone techniques that will help musicians understand how to translate "live" environments into recorded sound. The book covers theory and the technical aspects of recording from sound source to delivery: the nature of soundwaves and their behavior in rooms, microphone types and the techniques of recording in stereo, proximity and phase, file types, tracking and critical listening, loudness, meters, and the post-production processes of EQ, control of dynamic range (compressors, limiters, dynamic EQ, de-essers), and reverberation (both digital reflection simulation and convolution), with some discussion of commercially available digital plugins. The final part of the book applies this knowledge to common recording situations, showcasing not only strategies for recording soloists and small ensembles, along with case studies of several recordings, but also studio techniques that can enhance or replace the capture of performances in ambient spaces, such as close miking and the addition of artificial reverberation. Recording Classical Music provides the tools necessary for anyone interested in classical music production to track, mix, and deliver audio recordings themselves or to supervise the work of others.

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 new Springer volume provides a comprehensive and detailed look at current approaches to automated question answering. The level of presentation is suitable for newcomers to the field as well as for professionals wishing to study this area and/or to build practical QA systems. The book can serve as a "how-to" handbook for IT practitioners and system developers. It can also be used to teach graduate courses in Computer Science, Information Science and related disciplines.