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Books : Computers & Internet : Software : Business : Speech Processing
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Text provides a catalog of sixty-five patterns, with real-world solutions that demonstrate the formidable power of messaging and help you design effective messaging solutions for your enterprise. DLC: Telecommunication--Message processing.
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Free at last! Finally, someone has come along to free you from your keyboard. With Dragon NaturallySpeaking, the miraculous voice-recognition software in your computer, you can browse the Web, control your applications, control your desktop, write documents, and more without ever once laying finger to plastic. But don’t run out and get yourself fitted for that Star Fleet uniform just yet, cadet. Dragon NaturallySpeaking is the most accurate voice recognition software on the market, and while it really does deliver on all its claims, it can be very finicky, and getting top results can be tricky.The complete guide to the care of feeding or your Dragon, Dragon NaturallySpeaking For Dummies is a must-have companion for voice-recognition trailblazers who are ready to:
- Kiss that keyboard goodbye and say hello to hands-free computing
- Verbally control your Windows desktop and most applications
- Dictate, edit, format and proofread documents in Word and WordPerfect
- Browse the Web and compose and send email by voice
- Use a pocket digital recorder on the run
Here’s all you need to fire up your Dragon and get it dancing to your tune. Your total guide to installing, configuring, fine-tuning and getting the most out of that amazing voice recognition software, Dragon NaturallySpeaking For Dummies covers all the bases, including:
- Installing, configuring, and launching your Dragon
- Dictating, editing, proofreading, and formatting documents in NaturallySpeaking
- Recording speech onto the NaturallySpeaking recorder and transcribing recorded speech
- Dictating into other applications
- Controlling your desktop and windows by voice
- Using NaturalWord for Word and WordPerfect
- Browsing the Web, emailing and faxing by voice
- Managing databases hands-free
- Maximizing voice recognition accuracy
- Having multiple users and vocabularies
- Adding specialized items and verbal shortcuts to Dragon’s vocabulary
With the introduction of Dragon NaturallySpeaking the old dream of hands-free computing has finally become reality. Now let Dragon NaturallySpeaking For Dummies show you how to give your Dragon wings and make it soar.
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An explosion of Web-based language techniques, merging of distinct fields, availability of phone-based dialogue systems, and much more make this an exciting time in speech and language processing. The first of its kind to thoroughly cover language technology – at all levels and with all modern technologies – this book takes an empirical approach to the subject, based on applying statistical and other machine-learning algorithms to large corporations. Builds each chapter around one or more worked examples demonstrating the main idea of the chapter, usingthe examples to illustrate the relative strengths and weaknesses of various approaches. Adds coverage of statistical sequence labeling, information extraction, question answering and summarization, advanced topics in speech recognition, speech synthesis. Revises coverage of language modeling, formal grammars, statistical parsing, machine translation, and dialog processing. A useful reference for professionals in any of the areas of speech and language processing.
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Microsoft Speech Server is becoming increasingly popular. There are three primary components developers wanting to develop speech applications need to be familiar with: the Speech SDK, Telephony, and ASP.NET server controls. Each of these can be used independently, but in many cases, all three need to be used to build truly compelling applications.
Pro Microsoft Speech Server 2007 walks intermediate to advanced developers through the basics of speech and telephony technology. It then addresses Microsofts specific implementations and what it can do for most companies. From there, the specific components are discussed individually in depth. Youll create an application from scratch, building upon an existing web site, but adding brand new functionality as well. All of the issues associated with setup, security and administration, development, debugging, and deployment are included in the walkthroughs.
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This book reflects decades of important research on the mathematical foundations of speech recognition. It focuses on underlying statistical techniques such as hidden Markov models, decision trees, the expectation-maximization algorithm, information theoretic goodness criteria, maximum entropy probability estimation, parameter and data clustering, and smoothing of probability distributions. The author's goal is to present these principles clearly in the simplest setting, to show the advantages of self-organization from real data, and to enable the reader to apply the techniques.
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Winner, 2007 International Communication Association Outstanding Book Award for 2005-2006.
Interfaces that talk and listen are populating computers, cars, call centers, and even home appliances and toys, but voice interfaces invariably frustrate rather than help. In Wired for Speech, Clifford Nass and Scott Brave reveal how interactive voice technologies can readily and effectively tap into the automatic responses all speech—whether from human or machine—evokes. Wired for Speech demonstrates that people are "voice-activated": we respond to voice technologies as we respond to actual people and behave as we would in any social situation. By leveraging this powerful finding, voice interfaces can truly emerge as the next frontier for efficient, user-friendly technology.
Wired for Speech presents new theories and experiments and applies them to critical issues concerning how people interact with technology-based voices. It considers how people respond to a female voice in e-commerce (does stereotyping matter?), how a car's voice can promote safer driving (are "happy" cars better cars?), whether synthetic voices have personality and emotion (is sounding like a person always good?), whether an automated call center should apologize when it cannot understand a spoken request ("To Err is Interface; To Blame, Complex"), and much more. Nass and Brave's deep understanding of both social science and design, drawn from ten years of research at Nass's Stanford laboratory, produces results that often challenge conventional wisdom and common design practices. These insights will help designers and marketers build better interfaces, scientists construct better theories, and everyone gain better understandings of the future of the machines that speak with us.
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Engineer your way to excellence! This professional resource explains in full detail how to build VoiceXML-based applications using real-world programs you can adapt for your own projects. The book includes three full-scale, enterprise-level applications complete with all source code.
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Essential principles, practical examples, current applications, and leading-edge research.In this book, Thomas F. Quatieri presents the field's most intensive, up-to-date tutorial and reference on discrete-time speech signal processing. Building on his MIT graduate course, he introduces key principles, essential applications, and state-of-the-art research, and he identifies limitations that point the way to new research opportunities.
Quatieri provides an excellent balance of theory and application, beginning with a complete framework for understanding discrete-time speech signal processing. Along the way, he presents important advances never before covered in a speech signal processing text book, including sinusoidal speech processing, advanced time-frequency analysis, and nonlinear aeroacoustic speech production modeling. Coverage includes:
Speech production and speech perception: a dual view
Crucial distinctions between stochastic and deterministic problems
Pole-zero speech models
Homomorphic signal processing
Short-time Fourier transform analysis/synthesis
Filter-bank and wavelet analysis/synthesis
Nonlinear measurement and modeling techniques
The book's in-depth applications coverage includes speech coding, enhancement, and modification; speaker recognition; noise reduction; signal restoration; dynamic range compression, and more. Principles of Discrete-Time Speech Processing also contains an exceptionally complete series of examples and Matlab exercises, all carefully integrated into the book's coverage of theory and applications.
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The rich variety of activities for word retrieval and problem solving in the Workbook for Cognitive Skills has made it a favorite of clinicians over the past twenty years. The second edition of the red book builds on the original by adding 70 pages of entirely new exercises and 1,000 rewritten questions. Responding to the comments and suggestions of longtime users, the second edition of the Workbook for Cognitive Skills also features a sturdy ring binder that allows for trouble-free copying of exercises, a new page layout that is easier to read, and divider tabs that make it simple to find different sections.
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Provides a theoretically sound, technically accurate, and complete description of the basic knowledge and ideas that constitute a modern system for speech recognition by machine. Covers production, perception, and acoustic-phonetic characterization of the speech signal; signal processing and analysis methods for speech recognition; pattern comparison techniques; speech recognition system design and implementation; theory and implementation of hidden Markov models; speech recognition based on connected word models; large vocabulary continuous speech recognition; and task- oriented application of automatic speech recognition. For practicing engineers, scientists, linguists, and programmers interested in speech recognition.
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Provides an overview of this emerging field. Explains, both for managers and developers, what the issues, challenges, and opportunities are, and gives a clear sense of what a well-designed system requires. Softcover.
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The remarkable advances in computing and networking have sparked an enormous interest in deploying automatic speech recognition on mobile devices and over communication networks. This trend is accelerating.
This book brings together leading 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, which are expected to co-exist in the future. It offers a wide-ranging, unified approach to the topic and its latest development, also covering the most up-to-date standards and several off-the-shelf systems.
Key features:
• Provides an in-depth review of network speech recognition, distributed speech recognition, embedded speech recognition, systems and applications
• Begins with a comprehensive overview of the subject, discussing the pros and cons of the presented approaches, and guiding the reader through the following chapters
• Includes platforms like mobile phones, PDAs and automobiles
• Presents state-of-the-art methods, advanced systems, and the latest standards
• Offers working knowledge needed for both research and practice
• References supplemental material at associated complementary website at: http://asr.es.aau.dk
This all-inclusive text/reference is an essential read for graduate students, scientists and engineers working or researching in the field of speech recognition and processing. It offers a self-contained approach to this hot research topic.
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Markov models are used to solve challenging pattern recognition problems on the basis of sequential data as, e.g., automatic speech or handwriting recognition. This comprehensive introduction to the Markov modeling framework describes both the underlying theoretical concepts of Markov models - covering Hidden Markov models and Markov chain models - as used for sequential data and presents the techniques necessary to build successful systems for practical applications.
Additionally, the actual use of the technology in the three main application areas of pattern recognition methods based on Markov- Models - namely speech recognition, handwriting recognition, and biological sequence analysis - are demonstrated.
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The first book to provide comprehensive and up-to-date coverage of all major speech enhancement algorithms proposed in the last two decades, Speech Enhancement: Theory and Practice is a valuable resource for experts and newcomers in the field. The book covers traditional speech enhancement algorithms, such as spectral subtraction and Wiener filtering algorithms as well as state-of-the-art algorithms including minimum mean-squared error algorithms that incorporate signal-presence uncertainty and subspace algorithms that incorporate psychoacoustic models. The coverage includes objective and subjective measures used to evaluate speech quality and intelligibility. Divided into three parts, the book presents the digital-signal processing and speech signal fundamentals needed to understand speech enhancement algorithms, the various classes of speech enhancement algorithms proposed over the last two decades, and the methods and measures used to evaluate the performance of speech enhancement algorithms. The text is supplemented with examples and figures designed to help readers understand the theory. MATLAB® implementations of all major speech enhancement algorithms and a speech database that can be used for evaluation of noise reduction algorithms are included in an accompanying DVD-ROM. Providing clear and concise coverage of the subject, the author brings together a large body of knowledge about how human listeners compensate for acoustic noise when in noisy environments. This book is a valuable resource not only for engineers who want to implement the latest speech enhancement algorithms but also for speech practitioners who want to incorporate some of these algorithms into hearing aid applications for speech intelligibility and/or quality improvement.
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Techniques in Speech Acoustics provides an introduction to the acoustic analysis and characteristics of speech sounds. The first part of the book covers aspects of the source-filter decomposition of speech, spectrographic analysis, the acoustic theory of speech production and acoustic phonetic cues. The second part is based on computational techniques for analysing the acoustic speech signal including digital time and frequency analyses, formant synthesis, and the linear predictive coding of speech. There is also an introductory chapter on the classification of acoustic speech signals which is relevant to aspects of automatic speech and talker recognition. Included with the book is a CD-ROM containing extensive speech corpora, the EMU speech analysis tools, extensions to the X-LISP-STAT programming language that are adapted to speech analysis, and numerous exercises that are linked to the major themes of the book and which can be run on Windows-95 and UNIX platforms.
The book and CD-ROM are intended for use as teaching materials on undergraduate and postgraduate speech acoustics and experimental phonetics courses; they are also aimed at researchers from phonetics, linguistics, computer science, psychology and engineering who wish to gain an understanding of the basis of speech acoustics and its application to fields such as speech synthesis and automatic speech recognition.
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Media Resource Control Protocol (MRCP) is a new IETF protocol, providing a key enabling technology that eases the integration of speech technologies into network equipment and accelerates their adoption resulting in exciting and compelling interactive services to be delivered over the telephone. MRCP leverages IP telephony and Web technologies such as SIP, HTTP, and XML (Extensible Markup Language) to deliver an open standard, vendor-independent, and versatile interface to speech engines.Speech Processing for IP Networks brings these technologies together into a single volume, giving the reader a solid technical understanding of the principles of MRCP, how it leverages other protocols and specifications for its operation, and how it is applied in modern IP-based telecommunication networks. Focusing on the MRCPv2 standard developed by the IETF SpeechSC Working Group, this book will also provide an overview of its precursor, MRCPv1.
Speech Processing for IP Networks:
- Gives a complete background on the technologies required by MRCP to function, including SIP (Session Initiation Protocol), RTP (Real-time Transport Protocol), and HTTP (Hypertext Transfer Protocol).
- Covers relevant W3C data representation formats including Speech Synthesis Markup Language (SSML), Speech Recognition Grammar Specification (SRGS), Semantic Interpretation for Speech Recognition (SISR), and Pronunciation Lexicon Specification (PLS).
- Describes VoiceXML - the leading approach for programming cutting-edge speech applications and a key driver to the development of many of MRCP’s features.
- Explains advanced topics such as VoiceXML and MRCP interworking.
This text will be an invaluable resource for technical managers, product managers, software developers, and technical marketing professionals working for network equipment manufacturers, speech engine vendors, and network operators. Advanced students on computer science and engineering courses will also find this to be a useful guide.
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Voice over IP Technologies provides solid technical information on how to successfully design and implement a converged network, combining voice, data, fax and video transmissions into a cohesive networking infrastructure centered on the Internet Protocol.Converged networks, which combine voice, data, fax and video transmissions into a cohesive networking infrastructure -- all centered on the Internet Protocol, or IP -- promise a number of advantages over existing, separate networking environments. But to successfully design and implement a converged network requires expertise on both the voice and data networking sides of the house. Unfortunately, few individuals have these credentials -- either you are a voice networking expert, and familiar with circuit switching and connections between PBXs, or you are a data networking expert, familiar with packet switching and connections between routers and servers.
The objective of this text is to bridge the gap between the voice and data networking sides, and provide the reader with the opportunity to fill in their areas of weakness with solid technical information. In addition, this text presents a number of case studies, from architectural, financial and technical perspectives that illustrate real-world applications for these technologies.
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Finite-state devices, which include finite-state automata, graphs, and finite-state transducers, are in wide use in many areas of computer science. Recently, there has been a resurgence of the use of finite-state devices in all aspects of computational linguistics, including dictionary encoding, text processing, and speech processing. This book describes the fundamental properties of finite-state devices and illustrates their uses. Many of the contributors pioneered the use of finite-automata for different aspects of natural language processing. The topics, which range from the theoretical to the applied, include finite-state morphology, approximation of phrase-structure grammars, deterministic part-of-speech tagging, application of a finite-state intersection grammar, a finite-state transducer for extracting information from text, and speech recognition using weighted finite automata. The introduction presents the basic theoretical results in finite-state automata and transducers. These results and algorithms are described and illustrated with simple formal language examples as well as natural language examples.
Contributors: Douglas Appelt, John Bear, David Clemenceau, Maurice Gross, Jerry R. Hobbs, David Israel, Megumi Kameyama, Lauri Karttunen, Kimmo Koskenniemi, Mehryar Mohri, Eric Laporte, Fernando C. N. Pereira, Michael D. Riley, Emmanuel Roche, Yves Schabes, Max D. Silberztein, Mark Stickel, Pasi Tapanainen, Mabry Tyson, Atro Voutilainen, Rebecca N. Wright.
Language, Speech, and Communication series