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Digital Speech Processing

Digital Speech Processing

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Foundations and Trends
inSignal ProcessingVol. 1, Nos. 1–2 (2007) 1–194c
2007 L. R. Rabiner and R. W. SchaferDOI: 10.1561/2000000001
Introduction to Digital Speech Processing
Lawrence R. Rabiner
and Ronald W. Schafer
Rutgers University and University of California, Santa Barbara, USA,rabiner@ece.ucsb.edu 
Hewlett-Packard Laboratories, Palo Alto, CA, USA
Since even before the time of Alexander Graham Bell’s revolution-ary invention, engineers and scientists have studied the phenomenonof speech communication with an eye on creating more efficient andeffective systems of human-to-human and human-to-machine communi-cation. Starting in the 1960s, digital signal processing (DSP), assumeda central role in speech studies, and today DSP is the key to realizingthe fruits of the knowledge that has been gained through decades of research. Concomitant advances in integrated circuit technology andcomputer architecture have aligned to create a technological environ-ment with virtually limitless opportunities for innovation in speechcommunication applications. In this text, we highlight the central roleof DSP techniques in modern speech communication research and appli-cations. We present a comprehensive overview of digital speech process-ing that ranges from the basic nature of the speech signal, through avariety of methods of representing speech in digital form, to applica-tions in voice communication and automatic synthesis and recognitionof speech. The breadth of this subject does not allow us to discuss any
aspect of speech processing to great depth; hence our goal is to pro-vide a useful introduction to the wide range of important concepts thatcomprise the field of digital speech processing. A more comprehensivetreatment will appear in the forthcoming book,
Theory and Application of Digital Speech Processing 
The fundamental purpose of speech is communication, i.e., the trans-mission of messages. According to Shannon’s information theory [116],a message represented as a sequence of discrete symbols can be quanti-fied by its
information content 
in bits, and the rate of transmission of information is measured in bits/second (bps). In speech production, aswell as in many human-engineered electronic communication systems,the information to be transmitted is encoded in the form of a contin-uously varying (analog) waveform that can be transmitted, recorded,manipulated, and ultimately decoded by a human listener. In the caseof speech, the fundamental analog form of the message is an acous-tic waveform, which we call the
speech signal 
. Speech signals, as illus-trated in Figure 1.1, can be converted to an electrical waveform bya microphone, further manipulated by both analog and digital signalprocessing, and then converted back to acoustic form by a loudspeaker,a telephone handset or headphone, as desired. This form of speech pro-cessing is, of course, the basis for Bell’s telephone invention as well astoday’s multitude of devices for recording, transmitting, and manip-ulating speech and audio signals. Although Bell made his inventionwithout knowing the fundamentals of information theory, these ideas

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