In sampling the level of a continuous source signal is measured at regular time intervals to produce a sequence of samples. The source signal is then represented by a sequence of its samples. Sampling is part of the process of analog to digital conversion.
The time between consecutive samples is called the sampling interval. The number of samples taken per second is known as the sampling frequency. There is an inverse relationship between sampling interval and sampling frequency.
In digital audio the common sampling frequencies are 44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz and 192kHz for PCM and 2.8224MHz and 5.6448MHz for DSD.
An important effect of sampling is that the level of the original signal is only known at the instants at which it was sampled. Interpolation is necessary to estimate the level of the signal between two consecutive sampling instants. The shorter the sampling interval, corresponding to higher sampling frequency, the better is the estimate produced by interpolation. This is why higher sampling frequencies are preferred in digital audio.
©Wayne Butcher