Now that we’re familiar with notion of frequency, we can return to the question which was posed earlier in the section on aliasing. Namely, how does one choose a sampling rate (or period) which will avoid undersampling and oversampling, and still faithfully represent the signal being sampled? As it turns out, there is a theorem which tells us how to precisely calculate the proper sampling rate and it’s known as the NyquistShannon Sampling Theorem.^{1} The theorem is stated as follows,
The NyquistShannon Sampling Theorem
If a signal contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points [samples] spaced 1/(2B) seconds apart.
For example, if we are attempting to sample a 300 Hz sine wave, the NyquistShannon Theorem tells us that we must sample at a rate greater than 600 Hz to faithfully capture the incoming sinusoid. If you know what the highest frequency component in your signal is, you simply set your sampling rate to be greater than twice that frequency.
This formulation of the theorem is quite signalspecific. You can also choose to look at it from a more samplingratecentric viewpoint,
Given a sampling rate of FS hertz, you cannot accurately sample any signal containing frequencies greater than or equal to FS/2.
I sometimes find this interpretation easier to think about, since it can be confusing for the uninitiated to think about determining the "frequency content" of a signal without first understanding the DFT. This formulation is also helpful because it introduces the value of FS/2 (half the sampling rate) as being key. This value is so crucial to sampling theory that it’s often referred to using the special name of the Nyquist Limit or Nyquist Frequency.
Figure 1 shows a set of four continuous sine waves which are all being sampled at the same rate. You can adjust the sampling rate using the slider at the bottom of the visualization. Spend enough time with this visualization and you’ll notice some very strange properties of the sampling process. One such oddity will occur when you set the sampling rate to 4 Hz. In this case, the sampled version of the 3 Hz sinusoid will look exactly like the sampled version of the 1 Hz sinusoid except for being flipped over the xaxis. When the sampling rate is set to 2 Hz, all four sampled signals will look exactly the same.
Figure 1. Playing with the Sampling Period 

Recall for a moment the section on sound and perception. In that section we mentioned that the range of human hearing runs from about 20 Hz to 20,000 Hz. Since humans can’t actually hear frequency content above 20,000 Hz, frequencies above this limit are actively removed from most audio and music signals before they are sampled. Given what we now know about the Sampling Theorem, you won’t be surprised to hear that the most common sampling rate for audio and music signals is around 40,000 Hz, or twice the highest audible frequency.^{2}