This volume provides a comprehensive introduction to foundational topics in sound design for linear media, such as listening and recording; audio postproduction; key musical concepts and forms such as harmony, conceptual sound design, electronica, soundscape, and electroacoustic composition; the audio commons; and sound's ontology and phenomenology. The reader will gain a broad understanding of the key concepts and practices that define sound design for its use with moving images as well as important forms of composed sound. The chapters are written by international authors from diverse backgrounds who provide multidisciplinary perspectives on sound in its linear forms. The volume is designed as a textbook for students and teachers, as a handbook for researchers in sound, media and experience, and as a survey of key trends and ideas for practitioners interested in exploring the boundaries of their profession.

Simon Grimm examines new multi-microphone signal processing strategies that aim to achieve noise reduction and dereverberation. Therefore, narrow-band signal enhancement approaches are combined with broad-band processing in terms of directivity based beamforming. Previously introduced formulations of the multichannel Wiener filter rely on the second order statistics of the speech and noise signals. The author analyses how additional knowledge about the location of a speaker as well as the microphone arrangement can be used to achieve further noise reduction and dereverberation.

The author, Andrea Pejrolo, is an experienced musician, composer/arranger, MIDI programmer, sound designer, and engineer. In this illustrated guidebook he focuses on the leading audio sequencers: ProTools, Digital Performer, Cubase SX, and Logic Audio, showing how to get the most out of them. Sequencing techniques are divided into basic, intermediate, and advanced sections, allowing readers with different levels of expertise to access the book at the level appropriate for them. The advice covered includes techniques such as groove quantizing, sounds layering, tap tempo, creative meter, and tempo changes, advanced use of plug-ins automation, and advanced mixing. The companion website includes examples of arrangements and techniques showing, for example, how to avoid common mistakes - and how to fix them when they occur.

This book constitutes the refereed proceedings of the 4th International Conference on Statistical Language and Speech Processing, SLSP 2016, held in Pilsen, Czech Republic, in October 2016. The 11 full papers presented together with two invited talks were carefully reviewed and selected from 38 submissions. The papers cover topics such as anaphora and coreference resolution; authorship identification, plagiarism and spam filtering; computer-aided translation; corpora and language resources; data mining and semantic web; information extraction; information retrieval; knowledge representation and ontologies; lexicons and dictionaries; machine translation; multimodal technologies; natural language understanding; neural representation of speech and language; opinion mining and sentiment analysis; parsing; part-of-speech tagging; question and answering systems; semantic role labeling; speaker identification and verification; speech and language generation; speech recognition; speech synthesis; speech transcription; speech correction; spoken dialogue systems; term extraction; text categorization; test summarization; user modeling.

The two-volume proceedings LNCS 9314 and 9315, constitute the proceedings of the 16th Pacific-Rim Conference on Multimedia, PCM 2015, held in Gwangju, South Korea, in September 2015. The total of 138 full and 32 short papers presented in these proceedings was carefully reviewed and selected from 224 submissions. The papers were organized in topical sections named: image and audio processing; multimedia content analysis; multimedia applications and services; video coding and processing; multimedia representation learning; visual understanding and recognition on big data; coding and reconstruction of multimedia data with spatial-temporal information; 3D image/video processing and applications; video/image quality assessment and processing; social media computing; human action recognition in social robotics and video surveillance; recent advances in image/video processing; new media representation and transmission technologies for emerging UHD services.

This book constitutes the refereed proceedings of the 17th International Conference on Speech and Computer, SPECOM 2015, held in Athens, Greece, in September 2015. The 59 revised full papers presented together with 2 invited talks were carefully reviewed and selected from 104 initial submissions. The papers cover a wide range of topics in the area of computer speech processing such as recognition, synthesis, and understanding and related domains including signal processing, language and text processing, multi-modal speech processing or human-computer interaction.

This book constitutes the refereed proceedings of the 18th International Conference on Text, Speech and Dialogue, TSD 2015, held in Pilsen, Czech Republic, in September 2015. The 67 papers presented together with 3 invited papers were carefully reviewed and selected from 138 submissions. They focus on topics such as corpora and language resources; speech recognition; tagging, classification and parsing of text and speech; speech and spoken language generation; semantic processing of text and speech; integrating applications of text and speech processing; automatic dialogue systems; as well as multimodal techniques and modelling.

This book constitutes the refereed proceedings of the 15th International Conference on Speech and Computer, SPECOM 2013, held in Pilsen, Czech Republic. The 48 revised full papers presented were carefully reviewed and selected from 90 initial submissions. The papers are organized in topical sections on speech recognition and understanding, spoken language processing, spoken dialogue systems, speaker identification and diarization, speech forensics and security, language identification, text-to-speech systems, speech perception and speech disorders, multimodal analysis and synthesis, understanding of speech and text, and audio-visual speech processing.

Sound Inventions is a collection of 34 articles taken from Experimental Musical Instruments, the seminal journal published from 1984 through 1999. In addition to the selected articles, the editors have contributed introductory essays, placing the material in cultural and temporal context, providing an overview of the field both before and after the time of original publication. The Experimental Musical Instruments journal contributed extensively to a number of sub-fields, including sound sculpture and sound art, sound design, tuning theory, musical instrument acoustics, timbre and timbral perception, musical instrument construction and materials, pedagogy, and contemporary performance and composition. This book provides a picture of this important early period, presenting a wealth of material that is as valuable and relevant today as it was when first published, making it essential reading for anyone researching, working with or studying sound.

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.

Design and build innovative, custom, data-driven Alexa skills for home or business. Working through several projects, this book teaches you how to build Alexa skills and integrate them with online APIs. If you have basic Python skills, this book will show you how to build data-driven Alexa skills. You will learn to use data to give your Alexa skills dynamic intelligence, in-depth knowledge, and the ability to remember. Data-Driven Alexa Skills takes a step-by-step approach to skill development. You will begin by configuring simple skills in the Alexa Skill Builder Console. Then you will develop advanced custom skills that use several Alexa Skill Development Kit features to integrate with lambda functions, Amazon Web Services (AWS), and Internet data feeds. These advanced skills enable you to link user accounts, query and store data using a NoSQL database, and access real estate listings and stock prices via web APIs. What You Will Learn Set up and configure your development environment properly the first time Build Alexa skills quickly and efficiently using Agile tools and techniques Create a variety of data-driven Alexa skills for home and business Access data from web applications and Internet data sources via their APIs Test with unit-testing frameworks throughout the development life cycle Manage and query your data using the DynamoDb NoSQL database engines Who This Book Is For Developers who wish to go beyond Hello World and build complex, data-driven applications on Amazon's Alexa platform; developers who want to learn how to use Lambda functions, the Alexa Skills SDK, Alexa Presentation Language, and Alexa Conversations; developers interested in integrating with public APIs such as real estate listings and stock market prices. Readers will need to have basic Python skills.

Why don't Guitar Hero players just pick up real guitars? What happens when millions of people play the role of a young black gang member in Grand Theft Auto: San Andreas? How are YouTube-based music lessons changing the nature of amateur musicianship? This book is about play, performance, and participatory culture in the digital age. Miller shows how video games and social media are bridging virtual and visceral experience, creating dispersed communities who forge meaningful connections by "playing along" with popular culture. Playing Along reveals how digital media are brought to bear in the transmission of embodied knowledge: how a Grand Theft Auto player uses a virtual radio to hear with her avatar's ears; how a Guitar Hero player channels the experience of a live rock performer; and how a beginning guitar student translates a two-dimensional, pre-recorded online music lesson into three-dimensional physical practice and an intimate relationship with a distant teacher. Through a series of engaging ethnographic case studies, Miller demonstrates that our everyday experiences with interactive digital media are gradually transforming our understanding of musicality, creativity, play, and participation.

Unleash your iPod touch and take it to the limit using secret tips and techniques. Fast and fun to read, Taking Your iPod touch 5 to the Max will help you get the most out of iOS 5 on your iPod touch. You'll find all the best undocumented tricks, as well as the most efficient and enjoyable introduction to the iPod touch available. Starting with the basics, you'll quickly move on to discover the iPod touch's hidden potential, like how to connect to a TV and get contract-free VoIP. From e-mail and surfing the Web, to using iTunes, iBooks, games, photos, ripping DVDs and getting free VoIP with Skype or FaceTime--whether you have a new iPod touch, or an older iPod touch with iOS 5, you'll find it all in this book. You'll even learn tips on where to get the best and cheapest iPod touch accessories. Get ready to take iPod touch to the max What you'll learn * How to get your music, videos, and data onto your iPod touch * How to manage your media * Tips for shopping in the App Store and iTunes Store * Getting the most out of iBooks * Using Mail on your iPod touch * Keeping in touch with FaceTime Who this book is for Anyone who wants to get the most out of their iPod touch 5.Table of Contents * Bringing Home the iPod touch * Putting Your Data and Media on the iPod touch * Interacting with Your iPod touch * Browsing with Wi-fi and Safari * Touching Photos and Videos * Touching Your Music * Shopping at the iTunes Store * Shopping at the App Store * Reading and Buying Books with iBooks * Setting Up and Using Mail * Staying on Time and Getting There * Using your Desk Set * Photographing and Recording the World Around You * Video Calling with FaceTime * Customizing Your iPod touch

Cross-Word Modeling for Arabic Speech Recognition utilizes phonological rules in order to model the cross-word problem, a merging of adjacent words in speech caused by continuous speech, to enhance the performance of continuous speech recognition systems. The author aims to provide an understanding of the cross-word problem and how it can be avoided, specifically focusing on Arabic phonology using an HHM-based classifier.

Automatic speech recognition (ASR) systems are finding increasing use in everyday life. Many of the commonplace environments where the systems are used are noisy, for example users calling up a voice search system from a busy cafeteria or a street. This can result in degraded speech recordings and adversely affect the performance of speech recognition systems. As the use of ASR systems increases, knowledge of the state-of-the-art in techniques to deal with such problems becomes critical to system and application engineers and researchers who work with or on ASR technologies. This book presents a comprehensive survey of the state-of-the-art in techniques used to improve the robustness of speech recognition systems to these degrading external influences.

Key features: Reviews all the main noise robust ASR approaches, including signal separation, voice activity detection, robust feature extraction, model compensation and adaptation, missing data techniques and recognition of reverberant speech.Acts as a timely exposition of the topic in light of more widespread use in the future of ASR technology in challenging environments.Addresses robustness issues and signal degradation which are both key requirements for practitioners of ASR.Includes contributions from top ASR researchers from leading research units in the field

Digital technology has changed the ways in which music is perceived, stored, distributed, mediated and created. The world of music is now a vast and complex jungle, teeming with CDs, MP3s, concerts, clubs, festivals, conferences, exhibitions, installations, websites, software programmes, scenes, Ideas and competing theories. In the eye of the storm stands David Toop, shedding light on the most interesting music now being made - on laptops, In downtown bars in Tokyo, wherever he finds it. Haunted Weather is part personal memoir and part travel journal, as well as an intensive survey of recent developments in digital technology, sonic theory and musical practice. Along the way Toop probes into the meaning of sound (and silence), offering fascinating insights into how computers can be used for improvisation. His wealth of musical knowledge provides inspiration for anyone interested in music.

We are surrounded by noise; we must be able to separate the signals we want to hear from those we do not. To overcome this 'cocktail party effect' we have developed various strategies; endowing computers with similar abilities would enable the development of devices such as intelligent hearing aids and robust speech recognition systems. This book describes a system which attempts to separate multiple, simultaneous acoustic sources using strategies based on those used by humans. It is both a review of recent work on the modelling of auditory processes, and a presentation of a new model in which acoustic signals are decomposed into elements. These structures are then re-assembled in accordance with rules of auditory organisation which operate to bind together elements that are likely to have arisen from the same source. The model is evaluated by measuring its ability to separate speech from a wide variety of other sounds, including music, phones and other speech.

Cutting-edge perspectives on a hot topic, with few competing titles on the market Contributor list includes some very well known professionals, as well as diverse academics from different disciplines Accessible and interdisciplinary introductory volume

Strike a balance between theory and practice! With this text, you'll, find a balance between theory and practice that allows you to build your understanding of the basic concepts, assumptions, and limitations of the theory of speech analysis and synthesis. The methods for data analysis as well as the theoretical background are provided to help you comprehend the analysis results. And you'll be able to study the features and properties of speech as a signal without having to record data and write software to analyze the data. The text includes two CDs that contain stand-alone and MATLAB software and speech and electroglottographic data. The CDs illustrate the effects that speech models and speech analysis procedures have on the quality of synthesized speech. An extensive speech database provides numerous speech files and other data. Examples included in each chapter demonstrate how to use the software. The CDs allow you to:
* Calculate the parameters of linear prediction speech models.
* Examine procedures for converting the speech of one speaker to sound like that of another speaker (i.e., voice conversion).
* Analyze and alter the temporal structure of the speech signal. This allows you to automatically parse speech into various features, such as voiced segments, unvoiced segments, nasal and non-nasal segments, fricatives, stops, and more.
* Create speech with a "high speaking rate" or generate speech with a "slow speaking rate."
* Adjust the parameters of the vocal fold model to change the vocal fold tension, length, thickness, mass, etc., in order to observe the effects of these parameters on the vibratory motion of the vocal folds.

This work addresses this problem in the short-time Fourier transform (STFT) domain. We divide the general problem into five basic categories depending on the number of microphones being used and whether the interframe or interband correlation is considered. The first category deals with the single-channel problem where STFT coefficients at different frames and frequency bands are assumed to be independent. In this case, the noise reduction filter in each frequency band is basically a real gain. Since a gain does not improve the signal-to-noise ratio (SNR) for any given subband and frame, the noise reduction is basically achieved by liftering the subbands and frames that are less noisy while weighing down on those that are more noisy. The second category also concerns the single-channel problem. The difference is that now the interframe correlation is taken into account and a filter is applied in each subband instead of just a gain. The advantage of using the interframe correlation is that we can improve not only the long-time fullband SNR, but the frame-wise subband SNR as well. The third and fourth classes discuss the problem of multichannel noise reduction in the STFT domain with and without interframe correlation, respectively. In the last category, we consider the interband correlation in the design of the noise reduction filters. We illustrate the basic principle for the single-channel case as an example, while this concept can be generalized to other scenarios. In all categories, we propose different optimization cost functions from which we derive the optimal filters and we also define the performance measures that help analyzing them.

"Speech Processing and Soft Computing" includes coverage of synergy between speech technology and bio-inspired soft computing methods. Through practical cases, the author explores, dissects and examines how soft computing may complement conventional techniques in speech enhancement and speech recognition in order to provide robust systems. The material is especially useful to graduate students and experienced researchers who are interested in expanding their horizons and investigating new research directions through review of the theoretical and practical settings of soft computing methods in very recent speech applications.

Modern Recording Techniques is the bestselling, authoritative guide to sound and music recording. Whether you're just starting out or are looking for a step-up in the industry, Modern Recording Techniques provides an in-depth read on the art and technologies of music production. It's a must-have reference for all audio bookshelves. Using its familiar and accessible writing style, this ninth edition has been fully updated, presenting the latest production technologies and includes an in-depth coverage of the DAW, networked audio, MIDI, signal processing and much more. A robust companion website features video tutorials, web-links, an online glossary, flashcards, and a link to the author's blog. Instructor resources include a test bank and an instructor's manual. The ninth edition includes:Updated tips, tricks and insights for getting the best out of your studio; An introduction to the Apple iOS in music production; Introductions to new technologies and important retro studio techniques; The latest advancements in DAW systems, signal processing, mixing and mastering.