t at the communications receiver or output device are the same as the bits at the input device, then the signal is transmitted accurately. For audio transmission, the code present at the end of the process does not have to be a perfect match for the original code. Some loss or distortion is acceptable. In this chapter we will look at several types of pulse code modulation (PCM), a com- mon type of encoding for telephony that is also used in the production of digital music CDs. Then we will look at linear predictive coding, including code-excited linear pre- diction (CELP), the coding scheme used for code division multiple access (CDMA) cel- lular telephony. We will close the chapter with a discussion of how the quality perfor- mance of speech coders is evaluated. 18.1 Pulse Code Modulation (PCM) The most basic way to send an audio signal as a sequence of bits is called pulse code modulation (PCM). Traditional circuit-switched telephone lines use PCM, as do audio compact disks. To perform PCM encoding, we start with an audio signal in electrical form, a voltage level as a function of time, as shown in Fig. 18.1. Volts Figure 18.1 An audio signal, voltage as a function of time. 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 First we sample the timeline by dividing it into discrete intervals, which we call sam- ples. The number of samples per unit time is called the sampling rate. For each sam- ple, the audio input measures the signal voltage at that instant in time. The audio out- put starts with the sample measurements and reconstructs the audio waveform by filling in the gaps, as shown in Fig. 18.2. The sample reconstruction loses wiggles in the audio signal that are smaller than the sampling interval. There is a fundamental limit called the Nyquist frequency, half the sampling rate. Since a CD samples its audio 