ied in two main categories: speech waveform- based coding and Analysis-by-Synthesis (AbS) methods. The waveform-based method attempts to reproduce the time domain speech waveform as accurately as possible, while in the AbS method, the goal is to minimize the perceptual distortion by using linear predic- tion techniques. In other words, the coder parameters are evaluated by locally decoding the signal and comparing it to the original speech signal. Pulse Code Modulation (PCM), considered to be the most common voice-coding tech- nique, merely involves sampling and quantizing the input waveform. PCM is described in more detail in the next section. As shown in Figure 1.2, assuming standard, voice-only appli- cations, the spectral extent of the input signal is taken as 300 Hz to 3.4 kHz (as evidenced by the formant/fricative sound generation mechanisms). With strict band limiting at 4 kHz (allowing 600 Hz for filter roll-off), standard sampling theory requires 8 ksamples/s. In contrast with linear PCM, where the sample amplitudes are presented on a linear scale, loga- rithmic PCM (log-PCM) uses a nonlinear quantizer to reproduce low-amplitude signals. There are two major variants of log-PCM: A-law, which is used in North America and Japan; and A-law, which is used mostly in Europe. These companding laws are explained with more details in the next section. Chapter 1 Voice Transmission DPCM Samples PCM Samples Fig. 1.3 Comparison of the histogram of the PCM and DPCM samples. -0.05 -0.03 -0.01 Adjacent speech samples are highly correlated. This means that the compression effi- ciency can be improved by removing this short-term redundancy. Differential PCM (DPCM) predicts the current sample from previous samples and, instead of the original speech sample, the prediction error is quantized and coded. With an accurate prediction, for most of the time, the prediction error is concentrated near zeros and can be coded with fewer bits compared to the original signal. Figure 1.3 compares the histogram of the PCM samples and DPCM