Audio signals to and from the real (analog) world have a less immediately obvious mapping to the digital world. Audio signals vary depending on the application. Human speech has a well understood spectrum, and set of characteristics, whereas musical input is much more varied, and the human ear and perception and cognition systems behave rather differently in each case. For example, when a speech signal degrades badly, humans make use of comprehension to interpolate. This may be harder or easier to do with music depending on levels of expertise and familiarity with the style/idiom.
Basically, for speech, the analog signal from a microphone is passed through several stages. Firstly a band pass filter is applied eliminating frequencies in the signal that we are not interested in (e.g. for telephone quality speech, above 3.6Khz).4.1 Then the signal is sampled, converting the analog signal into a sequence of values, each of which represents the amplitude of the analogue signal over a small discrete time interval. This is then quantised, or mapped into one of a set of fixed values - e.g. for telephone quality speech, one of 2**8, or 256 possible values. These values are then coded (represented in some standard form) for transmission or storage.
The process at the receiver is simply the reverse.
There are a number of particular features of the speech signal that make it particularly possible to compress, and we look at some of these in chapter four.