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Speech and Audio Coding for Wireless and Network Applications
contains 34 chapters, loosely grouped into six topical areas. The
chapters in this volume reflect the progress and present the state
of the art in low-bit-rate speech coding, primarily at bit rates
from 2.4 kbit/s to 16 kbit/s. Together they represent important
contributions from leading researchers in the speech coding
community. Speech and Audio Coding for Wireless and Network
Applications contains contributions describing technologies that
are under consideration as standards for such applications as
digital cellular communications (the half-rate American and
European coding standards). A brief Introduction is followed by a
section dedicated to low-delay speech coding, a research direction
which emerged as a result of the CCITT requirement for a universal
low-delay 16 kbit/s speech coding technology and now continues with
the objective of achieving toll quality with moderate delay at a
rate of 8 kbit/s. A section on the important topic of speech
quality evaluation is then presented. This is followed by a section
on speech coding for wireless transmission, and a section on audio
coding which covers not only 7 kHz bandwidth speech, but also
wideband coding applicable to high fidelity music. The book
concludes with a section on speech coding for noisy transmission
channels, followed by a section addressing future research
directions. Speech and Audio Coding for Wireless and Network
Applications presents a cross-section of the key contributions in
speech and audio coding which have emerged recently. For this
reason, the book is a valuable reference for all researchers and
graduate students in the speech coding community.
Herb Caen, a popular columnist for the San Francisco Chronicle,
recently quoted a Voice of America press release as saying that it
was reorganizing in order to "eliminate duplication and redundancy.
" This quote both states a goal of data compression and illustrates
its common need: the removal of duplication (or redundancy) can
provide a more efficient representation of data and the quoted
phrase is itself a candidate for such surgery. Not only can the
number of words in the quote be reduced without losing informa
tion, but the statement would actually be enhanced by such
compression since it will no longer exemplify the wrong that the
policy is supposed to correct. Here compression can streamline the
phrase and minimize the em barassment while improving the English
style. Compression in general is intended to provide efficient
representations of data while preserving the essential information
contained in the data. This book is devoted to the theory and
practice of signal compression, i. e., data compression applied to
signals such as speech, audio, images, and video signals (excluding
other data types such as financial data or general purpose computer
data). The emphasis is on the conversion of analog waveforms into
efficient digital representations and on the compression of digital
information into the fewest possible bits. Both operations should
yield the highest possible reconstruction fidelity subject to
constraints on the bit rate and implementation complexity."
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."
Speech and Audio Coding for Wireless and Network Applications
contains 34 chapters, loosely grouped into six topical areas. The
chapters in this volume reflect the progress and present the state
of the art in low-bit-rate speech coding, primarily at bit rates
from 2.4 kbit/s to 16 kbit/s. Together they represent important
contributions from leading researchers in the speech coding
community. Speech and Audio Coding for Wireless and Network
Applications contains contributions describing technologies that
are under consideration as standards for such applications as
digital cellular communications (the half-rate American and
European coding standards). A brief Introduction is followed by a
section dedicated to low-delay speech coding, a research direction
which emerged as a result of the CCITT requirement for a universal
low-delay 16 kbit/s speech coding technology and now continues with
the objective of achieving toll quality with moderate delay at a
rate of 8 kbit/s. A section on the important topic of speech
quality evaluation is then presented. This is followed by a section
on speech coding for wireless transmission, and a section on audio
coding which covers not only 7 kHz bandwidth speech, but also
wideband coding applicable to high fidelity music. The book
concludes with a section on speech coding for noisy transmission
channels, followed by a section addressing future research
directions. Speech and Audio Coding for Wireless and Network
Applications presents a cross-section of the key contributions in
speech and audio coding which have emerged recently. For this
reason, the book is a valuable reference for all researchers and
graduate students in the speech coding community.
Herb Caen, a popular columnist for the San Francisco Chronicle,
recently quoted a Voice of America press release as saying that it
was reorganizing in order to "eliminate duplication and redundancy.
" This quote both states a goal of data compression and illustrates
its common need: the removal of duplication (or redundancy) can
provide a more efficient representation of data and the quoted
phrase is itself a candidate for such surgery. Not only can the
number of words in the quote be reduced without losing informa
tion, but the statement would actually be enhanced by such
compression since it will no longer exemplify the wrong that the
policy is supposed to correct. Here compression can streamline the
phrase and minimize the em barassment while improving the English
style. Compression in general is intended to provide efficient
representations of data while preserving the essential information
contained in the data. This book is devoted to the theory and
practice of signal compression, i. e., data compression applied to
signals such as speech, audio, images, and video signals (excluding
other data types such as financial data or general purpose computer
data). The emphasis is on the conversion of analog waveforms into
efficient digital representations and on the compression of digital
information into the fewest possible bits. Both operations should
yield the highest possible reconstruction fidelity subject to
constraints on the bit rate and implementation complexity."
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."
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