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Speech processing and speech transmission technology are expanding fields of active research. New challenges arise from the 'anywhere, anytime' paradigm of mobile communications, the ubiquitous use of voice communication systems in noisy environments and the convergence of communication networks toward Internet based transmission protocols, such as Voice over IP. As a consequence, new speech coding, new enhancement and error concealment, and new quality assessment methods are emerging. Advances in Digital Speech Transmission provides an up-to-date overview of the field, including topics such as speech coding in heterogeneous communication networks, wideband coding, and the quality assessment of wideband speech. Provides an insight into the latest developments in speech processing and speech transmission, making it an essential reference to those working in these fields Offers a balanced overview of technology and applications Discusses topics such as speech coding in heterogeneous communications networks, wideband coding, and the quality assessment of the wideband speech Explains speech signal processing in hearing instruments and man-machine interfaces from applications point of view Covers speech coding for Voice over IP, blind source separation, digital hearing aids and speech processing for automatic speech recognition Advances in Digital Speech Transmission serves as an essential link between the basics and the type of technology and applications (prospective) engineers work on in industry labs and academia. The book will also be of interest to advanced students, researchers, and other professionals who need to brush up their knowledge in this field.
The enormous advances in digital signal processing (DSP) technology have contributed to the wide dissemination and success of speech communication devices – be it GSM and UMTS mobile telephones, digital hearing aids, or human-machine interfaces. Digital speech transmission techniques play an important role in these applications, all the more because high quality speech transmission remains essential in all current and next generation communication networks. Enhancement, coding and error concealment techniques improve the transmitted speech signal at all stages of the transmission chain, from the acoustic front-end to the sound reproduction at the receiver. Advanced speech processing algorithms help to mitigate a number of physical and technological limitations such as background noise, bandwidth restrictions, shortage of radio frequencies, and transmission errors. Digital Speech Transmission provides a single-source, comprehensive guide to the fundamental issues, algorithms, standards, and trends in speech signal processing and speech communication technology. The authors give a solid, accessible overview of fundamentals of speech signal processing speech coding, including new speech coders for GSM and UMTS error concealment by soft decoding artificial bandwidth extension of speech signals single and multi-channel noise reduction acoustic echo cancellation This text is an invaluable resource for engineers, researchers, academics, and graduate students in the areas of communications, electrical engineering, and information technology.
This book constitutes the refereed proceedings of the Third International Conference on Advances in Information Security and Its Applications, ISA 2009, held in Seoul, Korea, in June 2009. The 41 revised full papers presented were carefully reviewed and selected from 137 submissions. The papers are organized in topical sections on cryptographic algorithms, authentication and identity management, authorization and access control, biometrics and computer forensics, cryptographic protocols, data integrity and privacy, key management and recovery, mobile and RFID network security, firewall, IDs, anti-virus, and other security products, internet and web services security, cyber-attack and cyber-terrorism, other security research, together with the articles from the workshops MoWiN 2009, NASSUE 2009, IAWSN 2009, WNGS 2009 & CGMS 2009, SHCI-ISA 2009.
Speech coding has been an ongoing area of research for several decades, yet the level of activity and interest in this area has expanded dramatically in the last several years. Important advances in algorithmic techniques for speech coding have recently emerged and excellent progress has been achieved in producing high quality speech at bit rates as low as 4.8 kb/s. Although the complexity of the newer more sophisticated algorithms greatly exceeds that of older methods (such as ADPCM), today's powerful programmable signal processor chips allow rapid technology transfer from research to product development and permit many new cost-effective applications of speech coding. In particular, low bit rate voice technology is converging with the needs of the rapidly evolving digital telecom munication networks. The IEEE Workshop on Speech Coding for Telecommunications was held in Vancouver, British Columbia, Canada, from September 5 to 8, 1989. The objective of the workshop was to provide a forum for discussion of recent developments and future directions in speech coding. The workshop attracted over 130 researchers from several countries and its technical program included 51 papers.
We are delighted to welcome readers to the proceedings of the 6th Pacific-Rim Conference on Multimedia (PCM). The first PCM was held in Sydney, Australia, in 2000. Since then, it has been hosted successfully by Beijing, China, in 2001, Hsinchu, Taiwan, in 2002, Singapore in 2003, and Tokyo, Japan, in 2004, and finally Jeju, one of the most beautiful and fantastic islands in Korea. This year, we accepted 181 papers out of 570 submissions including regular and special session papers. The acceptance rate of 32% indicates our commitment to ensuring a very high-quality conference. This would not be possible without the full support of the excellent Technical Committee and anonymous reviewers that provided timely and insightful reviews. We would therefore like to thank the Program Committee and all reviewers. The program of this year reflects the current interests of the PCM’s. The accepted papers cover a range of topics, including, all aspects of multimedia, both technical and artistic perspectives and both theoretical and practical issues. The PCM 2005 program covers tutorial sessions and plenary lectures as well as regular presentations in three tracks of oral sessions and a poster session in a single track. We have tried to expand the scope of PCM to the artistic papers which need not to be strictly technical.
This book brings together many advanced topics in network and acoustic echo cancellation aimed towards enhancing the echo cancellation performance of next-generation telecommunication systems. The resulting compendium provides a coherent treatment of such topics not found otherwise in journals or other books.
"This book provides a comprehensive approach of signal processing tools regarding the enhancement, recognition, and protection of speech and audio signals. It offers researchers and practitioners the information they need to develop and implement efficient signal processing algorithms in the enhancement field"--Provided by publisher.

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