 division multiplexer Figure 3.15 PCM encoder. sampling theorem is satisfied, that is, that the analog signal does not contain frequencies higher than half of the sampling frequency. Then the analog sig- nal is sampled at an 8-kHz sampling frequency and the samples are nonline- arly coded into 8-bit words by a quantizer and an encoder. Words are then converted into serial form and multiplexed with other PCM-coded voice signals into a 2,048-Kbps primary rate signal that contains 30 voice channels according to the European standard. This 2-Mbps rate is a very common data rate in telecommunications networks. For example, digi- tal exchanges build up 2-Mbps streams with 30 PCM-coded subscriber inter- faces for internal transmission inside the equipment. The multiplexing process is described in Chapter 4. In the United States the corresponding data rate is 1.544 Mbps instead of 2.048 Mbps. In this DS1 system, each frame contains 24 speech channels and a framing bit. The sampling rate is the same 8 kHz and we get 8,000.{(8 -24)+1}=1544 Mbps . = (3.10) 3.6.7.2 PCM Decoder At the receiver the demultiplexer separates 64-Kbps individual channels that are then converted into 8-bit parallel sample values, as shown in Figure 3.16. Sample pulses are reconstructed and the resulting series is fil- tered to create a voice signal that closely resembles the original. 3.7 Other Speech-Coding Methods PCM was standardized during the 1970s and implementation of many more efficient coding methods has become feasible. By "more efficient" we mean Signals Carried over the Network Other PCM-coded speech signals Serial to parallel Decoder demultiplexer converter 8 kHz 2,048 Kbps Analog 64 Kbps 8 bits speech serial parallel Timing circuit LPF = Lowpass filter S/H = Sample and hold circuit TDM = Time division multiplexer Figure 3.16 PCM decoder. that we may get better quality at the same data rate or equal quality at a lower data rate. More sophisticated coding schemes are used, for example, in ISDN, where an ISDN telephone may transmit a bett