Audio sampling is rooted in digital-audio technology, the underlying principles of which were established as long ago as 1928 by electronics engineer Harry Nyquist and perfected in the late 1940s by mathematician, engineer and cryptographer Claude Shannon. The ideas these men established are now known as the Nyquist-Shannon Sampling Theorem, which is all about converting a continuous waveform into a series of discrete values from which the original waveform can be recreated.
The basic principle of signal sampling is very simple: it’s just a case of measuring a signal’s amplitude at regular time intervals. But the process of sampling and digitising an analogue waveform requires two significant approximations to be made. Firstly, the value that’s measured isn’t an instantaneous value, but is an average of the signal amplitude during the measurement time period. This means that finer details within that period will be lost.
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