An ADC takes a continuous analog signal and converts it to a discrete digital signal by taking samples that represent the signal’s amplitude at specific points in time. The sample rate (or sampling rate) is the number of samples taken per second.
The units for sample rate are samples per second (sps) or Hertz (Hz). The two are equivalent since the Hertz is equal to the reciprocal second, [Hz]=[s-1]. Hertz is the unit for frequency, and the sample rate is sometimes referred to as the sampling frequency. Sample rate and sampling frequency represent the same value.
For a sampled signal to be free of distortion known as aliasing, the Nyquist frequency of the sampler must be greater than the highest frequency that needs to be preserved. The Nyquist frequency is equal to half of the sample rate, so increasing sample rate means that higher frequencies can be recorded without aliasing.
The Nyquist criterion sets a theoretical lower limit, and in practice, sample rates must be (sometimes much) greater than twice the highest frequency to be sampled for the signal to be accurately converted. Higher sample rates typically come at the cost of slower speeds and higher power consumption.
Audio signals are subject to the same criteria as other analog signals. Humans can hear sounds in the 20-20,000 Hz range, so music and other sound waves are often sampled at 44.1kHz or 48kHz (slightly over the Nyquist frequency). Audio is also sometimes recorded at 88.2kHz or 96kHz in a process known as oversampling, wherein the sample rate is taken to be well over the Nyquist frequency in order to improve resolution and signal-to-noise ratio.