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Automatic Speech Recognition (ASR) is the enabling technology for hands-free dictation and voice-triggered computer menus. It is becoming increasingly prevalent in environments such as private telephone exchanges and real-time information services. Speech Recognition introduces the principles of ASR systems, including the theory and implementation issues behind multi-speaker continuous speech recognition. Focusing on the algorithms employed in commercial and laboratory systems, the treatment enables the reader to devise practical solutions for ASR system problems. It addresses in detail C++ programming techniques used to develop ASR applications, thus offering skills that will prove useful in any large C++ based software project. Possible extensions of the well-established ASR technology are highlighted, based on "Hidden Markov Models" applied to fields such as modelling and prediction of econometric series. Features include:* Accompanying website containing all C++ source code of a complete laboratory multi-speaker continuous-speech ASR system (e.g. Initialisation, Training, Recognition, Evaluation, etc.) www.wiley.com/go/becchetti_speech* Detailed theoretical, mathematical and technical explanations of ASR* A practical account of the functioning of ASRA crucial source of information for researchers, developers and project managers involved with ASR systems, Speech Recognition is also structured for use by students of digital signal processing, speech recognition and C++ programming techniques.

Release Date: June 09, 2009

Speech Recognition Applications: The Basics and Beyond provides step-by-step directions for getting started with speech recognition software. It also provides instruction in developing the basic speech recognition skills needed to dictate, correct, edit and format a variety of documents. Exercises are included for navigating the Internet by voice and creating e-mails; using Microsoft Word to create letters, reports, tables and macros; and using Microsoft Excel for creating spreadsheets. The unique design of this book offers a perfect training solution for students, teachers, and business professionals. It offers easy to follow lessons with step-by step directions and many screen shots and tips. The exercises will help you learn how to use speech recognition as a daily input device and will help you improve your overall speed and accuracy. Speech recognition technology has made numerous advancements over the past decade and has become easier to use and much more efficient. Speech recognition software is now being used by more and more individuals in a wide variety of industries and professional careers every day! Get a head start with this training manual today.

Automatic Speech Recognition (ASR) is the enabling technology for hands-free dictation and voice-triggered computer menus. It is becoming increasingly prevalent in environments such as private telephone exchanges and real-time information services. Speech Recognition introduces the principles of ASR systems, including the theory and implementation issues behind multi-speaker continuous speech recognition. Focusing on the algorithms employed in commercial and laboratory systems, the treatment enables the reader to devise practical solutions for ASR system problems. It addresses in detail C++ programming techniques used to develop ASR applications, thus offering skills that will prove useful in any large C++ based software project. Possible extensions of the well-established ASR technology are highlighted, based on "Hidden Markov Models" applied to fields such as modelling and prediction of econometric series. Features include:* Accompanying website containing all C++ source code of a complete laboratory multi-speaker continuous-speech ASR system (e.g. Initialisation, Training, Recognition, Evaluation, etc.) www.wiley.com/go/becchetti_speech* Detailed theoretical, mathematical and technical explanations of ASR* A practical account of the functioning of ASRA crucial source of information for researchers, developers and project managers involved with ASR systems, Speech Recognition is also structured for use by students of digital signal processing, speech recognition and C++ programming techniques.

MPN: 264713

Cable, Behind the Head

A complete overview of distant automatic speech recognition The performance of conventional Automatic Speech Recognition (ASR) systems degrades dramatically as soon as the microphone is moved away from the mouth of the speaker. This is due to a broad variety of effects such as background noise, overlapping speech from other speakers, and reverberation. While traditional ASR systems underperform for speech captured with far-field sensors, there are a number of novel techniques within the recognition system as well as techniques developed in other areas of signal processing that can mitigate the deleterious effects of noise and reverberation, as well as separating speech from overlapping speakers. Distant Speech Recognitionpresents a contemporary and comprehensive description of both theoretic abstraction and practical issues inherent in the distant ASR problem. Key Features:Covers the entire topic of distant ASR and offers practical solutions to overcome the

A complete overview of distant automatic speech recognitionThe performance of conventional Automatic Speech Recognition (ASR) systems degrades dramatically as soon as the microphone is moved away from the mouth of the speaker. This is due to a broad variety of effects such as background noise, overlapping speech from other speakers, and reverberation. While traditional ASR systems underperform for speech captured with far-field sensors, there are a number of novel techniques within the recognition system as well as techniques developed in other areas of signal processing that can mitigate the deleterious effects of noise and reverberation, as well as separating speech from overlapping speakers.Distant Speech Recognitionpresents a contemporary and comprehensive description of both theoretic abstraction and practical issues inherent in the distant ASR problem.Key Features:Covers the entire topic of distant ASR and offers practical solutions to overcome the problems related to itProvides documentation and sample scripts to enable readers to construct state-of-the-art distant speech recognition systemsGives relevant background information in acoustics and filter techniques, Explains the extraction and enhancement of classification relevant speech featuresDescribes maximum likelihood as well as discriminative parameter estimation, and maximum likelihood normalization techniquesDiscusses the use of multi-microphone configurations for speaker tracking and channel combinationPresents several applications of the methods and technologies described in this bookAccompanying website with open source software and tools to construct state-of-the-art distant speech recognition systemsThis reference will be an invaluable resource for researchers, developers, engineers and other professionals, as well as advanced students in speech technology, signal processing, acoustics, statistics and artificial intelligence fields.

A complete overview of distant automatic speech recognitionThe performance of conventional Automatic Speech Recognition (ASR) systems degrades dramatically as soon as the microphone is moved away from the mouth of the speaker. This is due to a broad variety of effects such as background noise, overlapping speech from other speakers, and reverberation. While traditional ASR systems underperform for speech captured with far-field sensors, there are a number of novel techniques within the recognition system as well as techniques developed in other areas of signal processing that can mitigate the deleterious effects of noise and reverberation, as well as separating speech from overlapping speakers.Distant Speech Recognitionpresents a contemporary and comprehensive description of both theoretic abstraction and practical issues inherent in the distant ASR problem.Key Features:Covers the entire topic of distant ASR and offers practical solutions to overcome the problems related to itProvides documentation and sample scripts to enable readers to construct state-of-the-art distant speech recognition systemsGives relevant background information in acoustics and filter techniques, Explains the extraction and enhancement of classification relevant speech featuresDescribes maximum likelihood as well as discriminative parameter estimation, and maximum likelihood normalization techniquesDiscusses the use of multi-microphone configurations for speaker tracking and channel combinationPresents several applications of the methods and technologies described in this bookAccompanying website with open source software and tools to construct state-of-the-art distant speech recognition systemsThis reference will be an invaluable resource for researchers, developers, engineers and other professionals, as well as advanced students in speech technology, signal processing, acoustics, statistics and artificial intelligence fields.

New SpeechWare 3-in-1 TableMike (TK03) This is the newly released TableMike from SpeechWare and the result of more than a year of the best Belgian engineering and input from a number experienced speech recognition users/resellers including Chuck Runquist, Larry Allen, Lunis Orcutt and Speech Recognition Solutions. While the first generation product was good and clearly on par with other table-mounted microphones, the soon to be released "second generation" 3-in-1 is looking to be nothing less than extraordinary. Features/Specifications: USB interface (v.1 and 2.0 compliant) with audio output and input Sound ports Detachable boom with professional hi-fi mini-XLR connectors for easier removal, portability and storage Speech Equaliser for optimum Speech recognition performance (like a headset!) and Normal or HD VoIP mode Operating System: Windows® 7 / Vista® / XP® / 2000 / MacOS® 10.0 or higher / Linux without additional USB Sound drivers Compatible voice applications: Dragon NaturallySpeaking®, Microsoft Speech

Full Version, Platform(s): Mac; CD-ROM, MPN: 806709E11

This book provides a cross-disciplinary reference to speech in mobile and pervasive environmentsSpeech in Mobile and Pervasive Environments  addresses the issues related to speech processing on resource-constrained mobile devices. These include speech recognition in noisy environments, specialised hardware for speech recognition and synthesis, the use of context to enhance recognition and user experience, and the emerging software standards required for interoperability.  This book takes a multi-disciplinary look at these matters, while offering an insight into the opportunities and challenges of speech processing in mobile environs. In developing regions, speech-on-mobile is set to play a momentous role, socially and economically; the authors discuss how voice-based solutions and applications offer a compelling and natural solution in this setting.Key FeaturesProvides a holistic overview of all speech technology related topics in the context of mobilityBrings together the latest research in a logically connected way in a single volumeCovers hardware, embedded recognition and synthesis, distributed speech recognition, software technologies, contextual interfacesDiscusses multimodal dialogue systems and their evaluationIntroduces speech in mobile and pervasive environments for developing regionsThis book provides a comprehensive overview for beginners and experts alike. It can be used as a textbook for advanced undergraduate and postgraduate students in electrical engineering and computer science. Students, practitioners or researchers in the areas of mobile computing, speech processing, voice applications, human-computer interfaces, and information and communication technologies will also find this reference insightful. For experts in the above domains, this book complements their strengths. In addition, the book will serve as a guide to practitioners working in telecom-related industries.