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Both use a 4-wire circuit with regenerators The end user connects the line to a digital multiplexer, which also can act as a concentrator for telephone and data traffic For lower-speed lines (64 kbps or less), such as Digital Data Service, the digital multiplexer may be sited at the exchange To reconfigure leased lines, analog lines are normally terminated on a wiring or distribution frame Digital lines are connected to a Time Division Multiplex switch called a digital crossconnect
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323 Pulse Code Modulation (PCM)
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In the digital communication network, the analog voice signal is converted to digital at the earliest point possible In the existing network, most local loops are analog, but when the access line is terminated at the exchange, the analog voice signal is immediately converted to digital PCM In private or ISDN networks, the voice signal will be digitized either at the telephone handset or in the customer s premises Thus PCM in the modern network is really part of the access technology and over time will move further towards the end-user equipment in a fully digital access network Pulse Code Modulation (PCM) is the mechanism used for converting the analog voice channel to a digital signal Figure 35 shows the three stages of the process With a maximum input frequency of 34 kHz, Nyquist s sampling theorem indicates a minimum sampling rate of 68 kHz In fact, practical systems use a stable ( 50 ppm) 8 kHz sampling frequency The filter before the sampler removes signals above 4 kHz, which could cause aliasing
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Quantizing error The final stage of the PCM process is the quantization of the sampled values into one of several discrete levels This process results in a quantizing error (see Figure 36) In this example, the analog signal samples are encoded as one of eight possible levels midway between the decision thresholds (The analog signal could lie anywhere between three thresholds and be encoded as the same digital value) Therefore there is a maximum disparity equal to 05 times the quantizing interval between
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Figure 35 The PCM process digitizes the analog telephone signal To prevent aliasing, the analog bandwidth is restricted to a maximum frequency of 34 kHz before being sampled with a very stable 8 kHz clock These samples are then digitized into an 8-bit coded word, resulting in a 64 kbps signal
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Downloaded from Digital Engineering Library @ McGraw-Hill (wwwdigitalengineeringlibrarycom) 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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Figure 36 Digitizing a continuously variable analog signal always results in quantization error or noise,
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because the discrete digital values never exactly match the analog signal
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the true value of the signal and its quantized level This random difference is called quantization noise or quantization distortion The smaller the signal, the more severe the problem Quantization error can by reduced by using smaller intervals and encoding each sample with a longer digital word (A 12-bit ADC would be required for good quality) For a given sampling rate, however, this technique increases the bit rate of the system A better solution is to compand (compress and expand) the signal to improve the signal-to-noise ratio at low levels
Compression and expansion Figure 37 shows a typical companding curve that encodes the sample value (horizontal axis) into 128 levels using the logarithmic curve Notice how many of the levels are used for small signals to maintain an acceptable signal-to-noise ratio The 128 levels are represented by an 8-bit word (a byte or octet) As bytes are produced at the sampling rate of 8 kHz, the result is a bit stream at the familiar 64 kbps Slightly different companding equations are used in the US and Europe However, all aspects of the PCM process are fully standardized by the ITU-T (Recommendation G711, first issued in 1972) Figure 38 shows the variation of the signal-to-noise ratio as a function of signal level The linear portion at low signal levels is equivalent to the Lease Significant Bits (LSBs) of a 12-bit analog-to-digital (A/D) converter The flat portion (constant signal-to-noise) is the companded 8-bit conversion Companding is the simplest technique to reduce the data rate of digitally encoded analog signals such as telephony, video signals, and audio More sophisticated techDownloaded from Digital Engineering Library @ McGraw-Hill (wwwdigitalengineeringlibrarycom) 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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