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Author: David Burke Publisher: John Wiley & Sons ISBN: 9780470060605 Category : Technology & Engineering Languages : en Pages : 368
Book Description
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.
Author: David Burke Publisher: John Wiley & Sons ISBN: 9780470060605 Category : Technology & Engineering Languages : en Pages : 368
Book Description
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.
Author: Zheng-Hua Tan Publisher: Springer Science & Business Media ISBN: 1848001436 Category : Technology & Engineering Languages : en Pages : 408
Book Description
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.
Author: Jacob Benesty Publisher: Springer Science & Business Media ISBN: 3540491252 Category : Technology & Engineering Languages : en Pages : 1170
Book Description
This handbook plays a fundamental role in sustainable progress in speech research and development. With an accessible format and with accompanying DVD-Rom, it targets three categories of readers: graduate students, professors and active researchers in academia, and engineers in industry who need to understand or implement some specific algorithms for their speech-related products. It is a superb source of application-oriented, authoritative and comprehensive information about these technologies, this work combines the established knowledge derived from research in such fast evolving disciplines as Signal Processing and Communications, Acoustics, Computer Science and Linguistics.
Author: Lingfen Sun Publisher: Springer Science & Business Media ISBN: 144714905X Category : Computers Languages : en Pages : 279
Book Description
This book presents a review of the latest advances in speech and video compression, computer networking protocols, the assessment and monitoring of VoIP quality, and next generation network architectures for multimedia services. The book also concludes with three case studies, each presenting easy-to-follow step-by-step instructions together with challenging hands-on exercises. Features: provides illustrative worked examples and end-of-chapter problems; examines speech and video compression techniques, together with speech and video compression standards; describes the media transport protocols RTP and RTCP, as well as the VoIP signalling protocols SIP and SDP; discusses the concepts of VoIP quality of service and quality of experience; reviews next-generation networks based on the IP multimedia subsystem and mobile VoIP; presents case studies on building a VoIP system based on Asterisk, setting up a mobile VoIP system based on Open IMS and Android mobile, and analysing VoIP protocols and quality.
Author: Mohamed Ibnkahla Publisher: CRC Press ISBN: 0203496515 Category : Technology & Engineering Languages : en Pages : 872
Book Description
In recent years, a wealth of research has emerged addressing various aspects of mobile communications signal processing. New applications and services are continually arising, and future mobile communications offer new opportunities and exciting challenges for signal processing. The Signal Processing for Mobile Communications Handbook provi
Author: Daniel Minoli Publisher: John Wiley & Sons ISBN: 0471449350 Category : Technology & Engineering Languages : en Pages : 512
Book Description
Includes new coverage on the advances in signaling protocols,second-generation switching and the development of non-switchedalternatives, and the implementation lessons learned. Contains in-depth coverage of network architectures used tosupport VoIP, performance and voice quality considerations,compression and integration methods for IP tranmissions.
Author: Antonio Peinado Publisher: John Wiley & Sons ISBN: 0470024011 Category : Technology & Engineering Languages : en Pages : 274
Book Description
Automatic speech recognition (ASR) is a very attractive means for human-machine interaction. The degree of maturity reached by speech recognition technologies during recent years allows the development of applications that use them. In particular, ASR shows an enormous potential in mobile environments, where devices such as mobile phones or PDAs are used, and for Internet Protocol (IP) applications. Speech Recognition Over Digital Channels is the first book of its kind to offer a complete system comprehension, addressing the topics of distributed and network-based speech recognition issues and standards, the concepts of speech processing and transmission, and system architectures and robustness. Describes the different client/server architectures for remote speech recognition systems, by means of which the client transmits speech parameters through a digital channel to a remote recognition server Focuses on robustness against both adverse acoustic environments (in the front-end) and bit errors/packet loss Discusses four ETSI standards for distributed speech recognition; the understanding of the standards and the technologies behind them Provides the necessary background for the comprehension of remote speech recognition technologies This book will appeal to a wide-ranging audience: engineers using speech recognition systems, researchers involved in ASR systems and those interested in processing and transmitting speech such as signal processing and communications communities. It will also be of interest to technical experts requiring an understanding of recognition over mobile and IP networks, and postgraduate students working on robust speech processing.
Author: Taniar, David Publisher: IGI Global ISBN: 9781605660547 Category : Computers Languages : en Pages : 3721
Book Description
"This multiple-volume publication advances the emergent field of mobile computing offering research on approaches, observations and models pertaining to mobile devices and wireless communications from over 400 leading researchers"--Provided by publisher.
Author: Sen M. Kuo Publisher: John Wiley & Sons ISBN: 1118706684 Category : Technology & Engineering Languages : en Pages : 532
Book Description
Combines both the DSP principles and real-time implementations and applications, and now updated with the new eZdsp USB Stick, which is very low cost, portable and widely employed at many DSP labs. Real-Time Digital Signal Processing introduces fundamental digital signal processing (DSP) principles and will be updated to include the latest DSP applications, introduce new software development tools and adjust the software design process to reflect the latest advances in the field. In the 3rd edition of the book, the key aspect of hands-on experiments will be enhanced to make the DSP principles more interesting and directly interact with the real-world applications. All of the programs will be carefully updated using the most recent version of software development tools and the new TMS320VC5505 eZdsp USB Stick for real-time experiments. Due to its lower cost and portability, the new software and hardware tools are now widely used in university labs and in commercial industrial companies to replace the older and more expensive generation. The new edition will have a renewed focus on real-time applications and will offer step-by-step hands-on experiments for a complete design cycle starting from floating-point C language program to fixed-point C implementation, code optimization using INTRINSICS, and mixed C-and-assembly programming on fixed-point DSP processors. This new methodology enables readers to concentrate on learning DSP fundamentals and innovative applications by relaxing the intensive programming efforts, namely, the traditional DSP assembly coding efforts. The book is organized into two parts; Part One introduces the digital signal processing principles and theories, and Part Two focuses on practical applications. The topics for the applications are the extensions of the theories in Part One with an emphasis placed on the hands-on experiments, systematic design and implementation approaches. The applications provided in the book are carefully chosen to reflect current advances of DSP that are of most relevance for the intended readership. Combines both the DSP principles and real-time implementations and applications using the new eZdsp USB Stick, which is very low cost, portable and widely employed at many DSP labs is now used in the new edition Places renewed emphasis on C-code experiments and reduces the exercises using assembly coding; effective use of C programming, fixed-point C code and INTRINSICS will become the main focus of the new edition. Updates to application areas to reflect latest advances such as speech coding techniques used for next generation networks (NGN), audio coding with surrounding sound, wideband speech codec (ITU G.722.2 Standard), fingerprint for image processing, and biomedical signal processing examples. Contains new addition of several projects that can be used as semester projects; as well as new many new real-time experiments using TI’s binary libraries – the experiments are prepared with flexible interface and modular for readers to adapt and modify to create other useful applications from the provided basic programs. Consists of more MATLAB experiments, such as filter design, algorithm evaluation, proto-typing for C-code architecture, and simulations to aid readers to learn DSP fundamentals. Includes supplementary material of program and data files for examples, applications, and experiments hosted on a companion website. A valuable resource for Postgraduate students enrolled on DSP courses focused on DSP implementation & applications as well as Senior undergraduates studying DSP; engineers and programmers who need to learn and use DSP principles and development tools for their projects.
Author: Tokunbo Ogunfunmi Publisher: Springer ISBN: 1493914561 Category : Technology & Engineering Languages : en Pages : 347
Book Description
This book describes the basic principles underlying the generation, coding, transmission and enhancement of speech and audio signals, including advanced statistical and machine learning techniques for speech and speaker recognition with an overview of the key innovations in these areas. Key research undertaken in speech coding, speech enhancement, speech recognition, emotion recognition and speaker diarization are also presented, along with recent advances and new paradigms in these areas.
Author: David Burke
Author: David Burke
Author: Zheng-Hua Tan
Author: Jacob Benesty
Author: Lingfen Sun
Author: Mohamed Ibnkahla
Author: Daniel Minoli
Author: Antonio Peinado
Author: Taniar, David
Author: Sen M. Kuo
Author: Tokunbo Ogunfunmi