er quality 7-kHz speech band at 64 Kbps than before. Another example that we will briefly review is GSM, where speech requires only 13 or 7 Kbps. In the following sections, we review some methods that are used in tele- communications networks in addition to the PCM discussed in the previous section. We can divide voice coding methods into two categories: waveform and voice coding (vocoders) [1]. In waveform coding, such as PCM, we transmit information that describes a signal waveform in time domain. In vocoders we use characteristics of human voice. To understand the basic principle of vocoders, imagine that we have a set of signal models each identified by a code. We divide speech into, for example, 50-ms segments and choose one of the models that is closest to the signal to be encoded and send its identification code to the other end. The decoder reproduces the sig- nal corresponding to the received code. Vocoders may also split voice signals into several "components" in the frequency domain, each of them modeled separately for better quality. Vocoders introduce additional delay because each speech segment has to be analyzed before encoding. Waveform coding does not add delay and it usually give better quality but requires a higher data rate than vocoders. To achieve a suitable compromise between the quality and the data rate, the two basic principles are sometimes combined into hybrid coders. Introduction to Telecommunications Network Engineering In conventional PCM we encode all samples independently. We can improve encoding performance by assuming that the next sample value is not independent from the previous one, which is the case in practice. 3.7.1 Adaptive PCM (APCM) APCM is a variation of conventional PCM in which signal strength informa- tion is transmitted periodically in addition to sample values. Now a smaller number of bits is needed for samples and they define the quantum level inside a given scale. If the signal level is high, the quantizing error is high because the same number of levels is used