According to the Nyquist sampling theorem, in order to recover a signal via sampling, the sample rate must be greater than or equal to the two time (2x) the highest frequency in the signal. Doing so prevents aliasing due to frequency foldback about the 0 Hz (DC) axis. So, if you were digitally sampling a human voice with a maximum frequency content of, say, 6 kHz, then your sampling rate would need to be at least 12 kHz. In practice, the sample rate used is a little higher to provide a buffer, and a lowpass filter is placed in front of the A-to-D converter to make certain that higher frequency content does not make its way into the converter and subsequently get translated into an aliased (false) component. The concept applies to digital signals as well as with analog signals.