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complex process, implemented completely in silicon, which allows for voice to be encoded at rates as low as 2,400 bps. The resulting quality is far from toll quality, but it is certainly intelligible and its low bit-rate capability gives it a distinct advantage over other systems. LPC relies on the fact that each sound created by the human voice has unique attributes, such as frequency range, resonance, and loudness, among others. When voice samples are created in LPC, these attributes are used to generate prediction coefficients. These predictive coefficients represent linear combinations of previous samples, hence the name Linear Predictive Coding. Prediction coefficients are created by taking advantage of the known formants of speech, which are the resonant characteristics of the mouth and throat, which give speech its characteristic timbre and sound. This sound, referred to by speech pathologists as the buzz, can be described by both its pitch and its intensity. LPC, therefore, models the behavior of the vocal cords and the vocal tract itself. To create the digitized voice samples, the buzz is passed through an inverse filter that is selected based upon the value of the coefficients. The remaining signal, after the buzz has been removed, is called the residue. In the most common form of LPC, the residue is encoded as either a voiced or unvoiced sound. Voiced sounds are those that require vocal cord vibration, such as the g in glare, the b in boy, the d and g in dog. Unvoiced sounds such as the h in how, the sh in shoe, the f in frog require no vocal cord vibration. The transmitter creates and sends the prediction coefficients, which include measures of pitch, intensity and whatever voiced and unvoiced coefficients are required. The receiver undoes the process: It converts the voice residue, pitch, and intensity coefficients into a representation of the source signal, using a filter similar to the one used by the transmitter to synthesize the original signal.
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Human speech has many measurable (and therefore predictable) characteristics, one of which is a tendency to have embedded pauses. As a rule, people do not spew out a series of uninterrupted sounds; they tend to pause for emphasis, to collect their thoughts, to reword a phrase while the other person listens quietly on the other end of the line. When speech technicians monitor these pauses, they discover that during considerably more than half of the total connect time, the line is silent. Digital Speech Interpolation (DSI) takes advantage of this characteristic silence to drastically reduce the bandwidth required for a single
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Downloaded from Digital Engineering Library @ McGraw-Hill (www.digitalengineeringlibrary.com) Copyright 2004 The McGraw-Hill Companies. All rights reserved. Any use is subject to the Terms of Use as given at the website.
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channel. Whereas 24 channels can be transported over a typical T1 facility, DSI allows for as many as 120 conversations to be carried over the same circuit. The format is proprietary and requires the setting aside of a certain amount of bandwidth for overhead. A form of statistical multiplexing lies at the heart of DSI s functionality. The standard T-Carrier is a time-division multiplexed scheme in which channel ownership is assured: A user assigned to channel three will always own channel three, regardless of whether they are actually using the line. In DSI, channels are not owned. Instead, large numbers of users share a pool of available channels. When a user starts to talk, the DSI system assigns an available timeslot to that user and notifies the receiving end of the assignment. This system works well when the number of users is large, because statistical probabilities are more accurate and indicative of behavior in larger populations than in smaller ones. There is a downside to DSI, of course, and it comes in several forms. Competitive clipping occurs when more people start to talk than there are available channels, resulting in someone being unable to talk. Connection clipping occurs when the receiving end fails to learn what channel a conversation has been assigned within a reasonable amount of time, resulting in signal loss. Two approaches have been created to address these problems. In the case of competitive clipping, the system intentionally clips off the front end of the initial word of the second person who speaks. This technique is not optimal, but does prevent loss of the conversation and also obviates the problem of clipping out the middle of a conversation, which would be more difficult for the speakers to recover from. The loss of an initial syllable or two can be mentally reconstructed far more easily than sounds in the middle of a sentence. A second technique used to recover from clipping problems is to temporarily reduce the encoding rate. The typical encoding rate for DSI is 32 Kbps; in certain situations, the encoding rate may be reduced to 24 Kbps, thus freeing up significant bandwidth for additional channels. Both techniques are widely utilized in DSI systems.
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