Free Ebook Discrete-Time Processing of Speech Signals, by John R. Deller, John G. Proakis, John H. L. Hansen
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Discrete-Time Processing of Speech Signals, by John R. Deller, John G. Proakis, John H. L. Hansen
Free Ebook Discrete-Time Processing of Speech Signals, by John R. Deller, John G. Proakis, John H. L. Hansen
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Commercial applications of speech processing and recognition are fast becoming a growth industry that will shape the next decade. Now students and practicing engineers of signal processing can find in a single volume the fundamentals essential to understanding this rapidly developing field. IEEE Press is pleased to publish a classic reissue of Discrete-Time Processing of Speech Signals. Specially featured in this reissue is the addition of valuable World Wide Web links to the latest speech data references.
This landmark book offers a balanced discussion of both the mathematical theory of digital speech signal processing and critical contemporary applications. The authors provide a comprehensive view of all major modern speech processing areas: speech production physiology and modeling, signal analysis techniques, coding, enhancement, quality assessment, and recognition. You will learn the principles needed to understand advanced technologies in speech processing -- from speech coding for communications systems to biomedical applications of speech analysis and recognition.
Ideal for self-study or as a course text, this far-reaching reference book offers an extensive historical context for concepts under discussion, end-of-chapter problems, and practical algorithms. Discrete-Time Processing of Speech Signals is the definitive resource for students, engineers, and scientists in the speech processing field.
An Instructor's Manual presenting detailed solutions to all the problems in the book is available upon request from the Wiley Makerting Department.
- Sales Rank: #4035972 in Books
- Published on: 1993-03
- Original language: English
- Number of items: 1
- Dimensions: 9.75" h x 6.50" w x 1.50" l,
- Binding: Hardcover
- 928 pages
From the Publisher
This book covers the essential aspects of modern speech processing-- analysis, synthesis and coding, enhancement and quality assessment, and recognition. It also contains a comprehensive review of fundamental topics such as signal processing, stochastic processes, pattern recognition, and information theory, and includes an extensive background bibliography. The book provides state-of-the-art information for advanced research and development in the computer processing of speech, and gives the reader a thorough understanding of speech processing concepts.
From the Back Cover
Electrical Engineering Discrete-Time Processing of Speech Signals Commercial applications of speech processing and recognition are fast becoming a growth industry that will shape the next decade. Now students and practicing engineers of signal processing can find in a single volume the fundamentals essential to understanding this rapidly developing field. IEEE Press is pleased to publish a classic reissue of Discrete-Time Processing of Speech Signals. Specially featured in this reissue is the addition of valuable World Wide Web links to the latest speech data references. This landmark book offers a balanced discussion of both the mathematical theory of digital speech signal processing and critical contemporary applications. The authors provide a comprehensive view of all major modern speech processing areas: speech production physiology and modeling, signal analysis techniques, coding, enhancement, quality assessment, and recognition. You will learn the principles needed to understand advanced technologies in speech processing—from speech coding for communications systems to biomedical applications of speech analysis and recognition. Ideal for self-study or as a course text, this far-reaching reference book offers an extensive historical context for concepts under discussion, end-of-chapter problems, and practical algorithms. Discrete-Time Processing of Speech Signals is the definitive resource for students, engineers, and scientists in the speech processing field.
About the Author
John R. (Jack) Deller, Jr. is professor of electrical and computer engineering at Michigan State University where he directs the Speech Processing Laboratory. He received the 1998 IEEE Signal Processing Magazine Best Paper Award and the 1997 IEEE Signal Processing Society Meritorious Service Award for his six-year service as editor in chief of the IEEE Signal Processing Magazine. Dr. Deller is the coauthor of Digital Signal Processing and the Microcontroller (Prentice Hall, 1999) and currently serves as associate editor of the IEEE Transactions on Speech and Audio Processing. He is a Fellow of the IEEE.
John H. L. Hansen is an associate professor at the University of Colorado at Boulder; he holds appointments in the Department of Speech, Language, and Hearing Sciences and in the Department of Electrical and Computer Engineering. Dr. Hansen serves as associate director for the Center for Spoken Language Understanding (CSLU) and director of research activities of the Robust Speech Processing Laboratory at CSLU. He has served as technical advisor to the U.S. delegate for NATO, and he is associate editor of the IEEE Transactions on Speech and Audio Processing and of the IEEE Signal Processing Letters. Dr. Hansen received the 1990 National Science Foundation Research Initiation Award and has been named a Lilly Foundation Teaching Fellow for contributions to the advancement of engineering education.
John G. Proakis is a faculty member of the Electrical and Computer Engineering Department at Northeastern University. His professional experience is in the general area of digital communications and digital signal processing. Dr. Proakis is the author of Digital Communications (McGraw-Hill, 3rd ed., 1995) and the coauthor of Introduction to Digital Signal Processing (Prentice Hall, 3rd ed., 1996); Advanced Digital Signal Processing (Macmillan, 1992); Communication Systems Engineering (Prentice Hall, 1994); Digital Signal Processing Using MATLAB (PWS, 1997); and Contemporary Communication Systems Using MATLAB (PWS, 1998). He is a Life Fellow of the IEEE.
Most helpful customer reviews
8 of 8 people found the following review helpful.
The best comprehensive overview of speech processing
By Optimistix
I've seen a lot of books on Speech processing, but this one's the best by quite some margin. Rabiner & Schafer's classic could've edged this one out, but it's slightly dated now(1978 was a long time ago!) - it remains a great source for the fundamentals, though. However, this book is uptodate, as well as quite exhaustive in its coverage of the basics. One can get a solid grasp of basic speech processing(eg pitch analysis), synthesis, compression and recognition. The chapters on recognition, especially, are quite impressive,with detailed treatment of all the standard techniques in the field,including Dynamic time warping,Hidden Markov models & Neural networks. Everything from isolated word recognition to large vocabulary continuous speech recognition is touched upon.
This book is an ideal starting point for exploring the world of speech processing,which is a very rich area of research and has virtually limitless commercial applications,too.
A word of caution though - this is not for the layman.One requires a bit of a background in mathematics,and a little knowledge of digital signal processing won't hurt either.
I have only nice things to say about this book.The only possible crib could be the hefty price tag,but then this is typical of books in this field.
All in all,it's an excellent textbook,which ought to be read by everyone working in speech processing.
3 of 3 people found the following review helpful.
Good overview
By A Customer
This is an attempt to cover many aspects of speech processing, including modeling, coding, synthesis, enhancement, and recognition. Since each of the mentioned areas is a thick book on its own, the present book does not and cannot provide in depth coverage. I like some stories told in the book, like the development of the mechanical vocoder. But I find it to be wordy and in many instances, somehow impractical, like the exagerated mathematical treatment of the acoustic tube model. Also, the mathematical notations tend to be awkward. Due to technological advances, many topics contained in this book are already outdated.
2 of 2 people found the following review helpful.
A good reference, but too theoretical
By Charles John Gervasi
This book is very thorough, but at times it seems like the authors go out of their way to keep their discussion on a very theoretical level.
Chapter 1: A highly theoretical review of DSP. You need good knowledge of DSP to understand it.
Chapter 2: Goes over the human speech production and recognition systems. Here you get some practical info on the spectral and time-domain properties that distinguish speech sounds.
Chapter 3: Describes a model of the speech production based on a series of pulses passed through filters that correspond to features of the human speech production system. Practical issues such as which zero and pole values work best are left as an exercise to the reader.
Chapter 4: A lot of mathematics relating long-term statistical properties to those of a short frame of speech data. Contains good info on how to find recursive formulas for statistical properties of speech frames. It is a great shame that the authors don't include examples in MATLAB or pseudo code.
Chapter 5: Linear Prediction. Discusses a mathematical algorithm for creating a prediction filter that could be used to predict the next value in a series of data. In speech processing we are interested in using the coefficients of this prediction filter to encapsulate the properties of a speech frame. Examples of 1st,, 2nd, and 3rd order filters would have gone along way to illustrate how to implement this. There are some good formulas to measure the degree of similarities between speech frames based on their LP filter coefficients.
Chapter 6: Introduces the concept of the cepstrum. Cepstral analysis allows you to de-convolve speech data to separate the excitation source from the vocal tract filter. That way you can lifter out (a play on filter out) the excitation (responsible for pitch) and focus only on the vowel sounds. As always, how to make it work is left as an exercise.
The rest of the book is about speech coding and speech recognition. My class did not cover the later chapters, so I only read the parts that applied to projects.
The book helped me understand speech processing journal articles which tend to assume a lot of background knowledge.
The theoretical background that this book provides is necessary; good engineers need to understand the underpinnings of their bag of tricks. It is painful, however, at times. To apply the information in this book, you will certainly need a good mentor and a lot of tinkering in MATLAB.
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