ls. 18.1.1 Linear pulse code modulation We can quantize the voltage levels by picking a discrete set of values and assigning each value a number. When these levels are uniformly spaced from a minimum to a maximum voltage, we call the process linear PCM. Figure 18.3 shows the sampling process for the audio signal that we saw in Fig. 18.1. Each of the circles represents the closest level of the voltage at the sampling time. The figure shows nine distinct levels, whereas a CD is linear PCM with 65,536 levels (16 bits). When telephone systems use linear PCM, they use 8192 levels (13 bits). Each sample is converted to a binary number that goes to the audio output, which converts it back into analog electrical signals, as shown in Fig. 18.4. Linear PCM has 2This issue comes up in digital audio design regarding the CD, where designers try to achieve a top end of 20 kHz and make compromises in other parts of audio performance. 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. Speech Coding Speech Coding Volts Figure 18.4 Reconstructing the audio wave from linear PCM. voltage quantization errors as well, in addition to the time-sampling errors we saw in Fig. 18.2. The bit rate required for linear PCM is the sampling rate times the number of bits required for each sample. Thus a CD audio stream requires 44,100 samples times 16 bits per sample times 2 for stereo for a grand total of 1,411,200 bits per second. 3 Using 8000 samples times 13 bits per sample, we get 104,000 bits per second for a linear PCM telephone link. The 104,000 0-bit-per-second linear PCM link described here is never used in tele- phone transport. 4 The linear PCM is converted to a logarithmic scheme (u-law or A- law) or to adaptive differential PCM (ADPCM), as described in next two subsections. 18.1.2 u-Law and A-law PCM Looking at the reconstruction in Fig. 18.4, we notice that the s