This book discusses all aspects of computing for expressive performance, from the history of CSEMPs to the very latest research, in addition to discussing the fundamental ideas, and key issues and directions for future research. Topics and features: includes review questions at the end of each chapter; presents a survey of systems for real-time interactive control of automatic expressive music performance, including simulated conducting systems; examines two systems in detail, YQX and IMAP, each providing an example of a very different approach; introduces techniques for synthesizing expressive non-piano performances; addresses the challenges found in polyphonic music expression, from a statistical modelling point of view; discusses the automated analysis of musical structure, and the evaluation of CSEMPs; describes the emerging field of embodied expressive musical performance, devoted to building robots that can expressively perform music with traditional instruments.

In Monitoring Adaptive Spoken Dialog Systems, authors Alexander Schmitt and Wolfgang Minker investigate statistical approaches that allow for recognition of negative dialog patterns in Spoken Dialog Systems (SDS). The presented stochastic methods allow a flexible, portable and accurate use. Beginning with the foundations of machine learning and pattern recognition, this monograph examines how frequently users show negative emotions in spoken dialog systems and develop novel approaches to speech-based emotion recognition using hybrid approach to model emotions. The authors make use of statistical methods based on acoustic, linguistic and contextual features to examine the relationship between the interaction flow and the occurrence of emotions using non-acted recordings several thousand real users from commercial and non-commercial SDS. Additionally, the authors present novel statistical methods that spot problems within a dialog based on interaction patterns. The approaches enable future SDS to offer more natural and robust interactions. This work provides insights, lessons and inspiration for future research and development, not only for spoken dialog systems, but for data-driven approaches to human-machine interaction in general.

This book gives an overview of the research and application of speech technologies in different areas. One of the special characteristics of the book is that the authors take a broad view of the multiple research areas and take the multidisciplinary approach to the topics. One of the goals in this book is to emphasize the application. User experience, human factors and usability issues are the focus in this book.

Digital sound synthesis has long been approached using standard digital filtering techniques. Newer synthesis strategies, however, make use of physical descriptions of musical instruments, and allow for much more realistic and complex sound production and thereby synthesis becomes a problem of simulation.

This book has a special focus on time domain finite difference methods presented within an audio framework. It covers time series and difference operators, and basic tools for the construction and analysis of finite difference schemes, including frequency-domain and energy-based methods, with special attention paid to problems inherent to sound synthesis. Various basic lumped systems and excitation mechanisms are covered, followed by a look at the 1D wave equation, linear bar and string vibration, acoustic tube modelling, and linear membrane and plate vibration. Various advanced topics, such as the nonlinear vibration of strings and plates, are given an elaborate treatment.

Key features: Includes a historical overview of digital sound synthesis techniques, highlighting the links between the various physical modelling methodologies. A pedagogical presentation containing over 150 problems and programming exercises, and numerous figures and diagrams, and code fragments in the MATLAB(R) programming language helps the reader with limited experience of numerical methods reach an understanding of this subject. Offers a complete treatment of all of the major families of musical instruments, including certain audio effects.

"Numerical Sound Synthesis" is suitable for audio and software engineers, and researchers in digital audio, sound synthesis and more general musical acoustics. Graduate students in electrical engineering, mechanical engineering or computer science, working on the more technical side of digital audio and sound synthesis, will also find this book of interest.

Musical robotics is a multi- and trans-disciplinary research area involving a wide range of different domains that contribute to its development, including: computer science, multimodal interfaces and processing, artificial intelligence, electronics, robotics, mechatronics and more. A musical robot requires many different complex systems to work together; integrating musical representation, techniques, expressions, detailed analysis and controls, for both playing and listening. The development of interactive multimodal systems provides advancements which enable enhanced human-machine interaction and novel possibilities for embodied robotic platforms. This volume is focused on this highly exciting interdisciplinary field. This book consists of 14 chapters highlighting different aspects of musical activities and interactions, discussing cutting edge research related to interactive multimodal systems and their integration with robots to further enhance musical understanding, interpretation, performance, education and enjoyment. It is dichotomized into two sections: Section I focuses on understanding elements of musical performance and expression while Section II concentrates on musical robots and automated instruments. Musical Robots and Interactive Multimodal Systems provides an introduction and foundation for researchers, students and practitioners to key achievements and current research trends on interactive multimodal systems and musical robotics.

The accurate determination of the speech spectrum, particularly for short frames, is commonly pursued in diverse areas including speech processing, recognition, and acoustic phonetics. With this book the author makes the subject of spectrum analysis understandable to a wide audience, including those with a solid background in general signal processing and those without such background. In keeping with these goals, this is not a book that replaces or attempts to cover the material found in a general signal processing textbook. Some essential signal processing concepts are presented in the first chapter, but even there the concepts are presented in a generally understandable fashion as far as is possible. Throughout the book, the focus is on applications to speech analysis; mathematical theory is provided for completeness, but these developments are set off in boxes for the benefit of those readers with sufficient background. Other readers may proceed through the main text, where the key results and applications will be presented in general heuristic terms, and illustrated with software routines and practical "show-and-tell" discussions of the results. At some points, the book refers to and uses the implementations in the Praat speech analysis software package, which has the advantages that it is used by many scientists around the world, and it is free and open source software. At other points, special software routines have been developed and made available to complement the book, and these are provided in the Matlab programming language. If the reader has the basic Matlab package, he/she will be able to immediately implement the programs in that platform---no extra "toolboxes" are required.

In this work, the authors present a fully statistical approach to model non--native speakers' pronunciation. Second-language speakers pronounce words in multiple different ways compared to the native speakers. Those deviations, may it be phoneme substitutions, deletions or insertions, can be modelled automatically with the new method presented here.

The methods is based on a discrete hidden Markov model as a word pronunciation model, initialized on a standard pronunciation dictionary. The implementation and functionality of the methodology has been proven and verified with a test set of non-native English in the regarding accent.

The book is written for researchers with a professional interest in phonetics and automatic speech and speaker recognition.

This 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.

This book focuses on speech processing in the presence of low-bit rate coding and varying background environments. The methods presented in the book exploit the speech events which are robust in noisy environments. Accurate estimation of these crucial events will be useful for carrying out various speech tasks such as speech recognition, speaker recognition and speech rate modification in mobile environments. The authors provide insights into designing and developing robust methods to process the speech in mobile environments. Covering temporal and spectral enhancement methods to minimize the effect of noise and examining methods and models on speech and speaker recognition applications in mobile environments.

"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.

This anthology examines the various facets of video game music. Contributors from the fields of science and practice document its historical development, discuss the music's composition techniques, interactivity and function as well as attending to its performative aspects.

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.

"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."

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.

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.

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?

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.

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.

Understanding Game Scoring explores the unique collaboration between gameplay and composition that defines musical scoring for video games. Using an array of case studies reaching back into the canon of classic video games, this book illuminates the musical flexibility, user interactivity and sound programming that make game scoring so different from traditional modes of composition. Mack Enns explores the collaboration between game scorers and players to produce the final score for a game, through case studies of the Nintendo Entertainment System sound hardware configuration, and game scores, including the canonic scores for Super Mario Bros. (1985) and The Legend of Zelda (1986). This book is recommended reading for students and researchers interested in the composition and production of video game scores, as well as those interested in ludo-musicology.

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.

The new iOS 5-driveniPod touch devicesare much more than just music. These have all the features of a PDAincluding email, calendar, Google Maps, the App Store, and even phone capabilitiesas well as the ability to watch movies and play your favorite games, all packed into Apple's sleek design. With iPod touch Made Simple, iOS 5 Edition, you'll learn how to take advantage of all these features and more, now available using the new iOS 5. Packed with over 1,000 visuals and screenshots, this book will help you master all the functions of the iPod touch devices that run iOS 5and teach you time-saving techniques and tips along the way. What you'll learn Support for both Windows and Mac users Sync and manage all your music on the iPod touch or your computer Find the best App Store applications and games Save time with copy/paste & Spotlight search Play music, videos, TV shows, and podcasts Sync playlists, videos, contacts, calendar, and notes Fast email, phone, calendar, and browser tips Use Google Maps to find just about anything Bluetooth and Wi-Fi network setup & security All the best tips and tricks for the touch screen Who this book is for

This book is for those new to theiPod touch or the iPod touch devices running the new and latest iOS 5and even for seasoned users who want to learn new tips and techniques. Table of Contents Getting Started with iPod touch Typing Tips, Copy/Paste, and Search Sync Your iPod touch with iTunes Other Sync Methods Wi-Fi Connectivity Organize Your iPod touch Icons and Folders Personalize and Secure Your iPod touch Multitasking and Voice Control Playing Music Viewing Videos, TV Shows, and More iBooks and E-Books Surfing the Web with Safari FaceTime Video Messaging and Skype Email on Your iPod touch Working with Contacts Your Calendar iPod touch Photography Recording and Editing Videos iTunes on Your iPod touch The Amazing App Store Games and Fun Social Networking Eliminate Your Paper Notes Bluetooth on the iPod touch Utilities: Clock, Calculator, and Weather New Media: Reading Newspapers, Magazines, and More Find Your Way with Maps Troubleshooting Your iPod touch Your iTunes User Guide

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. "

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.