pres- sion using techniques such as entropy coding. Synthesis coding starts from human physi- ology and voice generation, and how its component parts may be represented by a series of discrete numbers. Major waveform coding techniques include pulse code modulation (PCM) used in ISDN and adaptive differential pulse code modulation (ADPCM) used in a number of standards including the Japanese Personal Handy-phone System (PHS). A number of syn- thesis coding techniques exist which are widely used in VoIP and mobile phone applications. From Analogue to Digital Figure 5.14 Logarithmic voice coding quantisation We will discuss Adaptive Multi-Rate (AMR) coding standards which are used in wireless telecommunication 3G and 4G standards. Pulse Code Modulation ISDN standards used a special modulation technique called Pulse Code Modulation (PCM). This technique is a pure A to D conversion waveform source coding. Voice is sampled at 8 kHz and quantised at 8 bits per sample, resulting in a 64 kbps source coding rate. Quantisation levels are mostly logarithmically distributed with finer (more) levels at smaller amplitudes and coarser (fewer) levels at higher amplitudes as illustrated in Figure 5.14. This quantisation level distribution increases the dynamic range of voice, giving more accuracy to smaller signals and less to larger signals. Such a quantisation technique reduces the signal to distortion ratio as noise impacts smaller signal samples. Adaptive Differential Pulse Code Modulation The PCM described above is an inefficient form of an information theory perspective since it does not take the correlation between neighbouring samples into account. In practice consecutive samples of an analogue signal are highly correlated and therefore a technique that decorrelates this interdependency can reduce the number of bits required to represent the signal. One such method is to only encode the difference between two consecutive samples. This is illustrated in Figure 5.15. The difference between consecutive samples can be encoded w