Digital audio has many advantages over analogue audio. Digital audio can be stored and transferred much more easily. It can also be processed by digital computers. The simplest way of encoding audio signals digitally is called pulse code modulation (PCM) (Pan 1). Basically, the signal is split up into discrete time intervals. Each time interval is assigned a value. The value is the binary value of a certain size and precision closest to the actual average value of the signal in that interval (the actual value for audio signals is the air pressure at a given time) (Pan 1). Recovering a signal from its digital representation is only possible if the frequency is less than half the sampling frequency (how often you assign a digital value) (Pan 2). Since audible frequencies range up to 20 kHz, there must be at least 40,000 samples a second. This makes PCM very inefficient. For instance, if there were 256 different values for the signal, that would require 8 bits per sample, so PCM data for one second would be 320,000 (8×40,000) bits. A better way to encode audio signals is called adaptive pulse code modulation (ADPCM).
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