When sound is recorded digitally, ``snapshots'' are taken of the analog sound at precise intervals. The snapshots are then processed by an analog-to-digital converter (ADC), which translates the analog sound into binary bits that can be stored on a disc, memory chip, or some other digital medium. The snapshots are called ``samples,'' and the process of capturing and converting the samples is called ``sampling.''
Here is another way to look at the concept of sampling. Think of a bouncing ball. As we watch the ball, its motion is continuous and smooth. If we want to capture the image of the bouncing ball to view later or to share with others, we might film it with a movie camera. As we film the ball, the camera takes a rapid sequence of still photographs. In fact, it takes twenty-four photographs each second. Once the film is developed, we can see by looking at the strip of film that each photograph shows the ball frozen at a specific point in its continuous motion, up and down. When the film is run through a projector, these still images are reproduced in their original sequence, twenty-four images per second, recreating, through illusion, the continuous and smooth motion of the bouncing ball.
When we sample sound, we take thousands of ``snapshots'' of the sound each second, and when these samples are played back in sequence, the resulting sound creates an illusion of smoothness and continuity, like the illusion of smooth motion in the film of the bouncing ball.
The two parameters of sampled sound that most directly affect its quality are its sampling rate and its resolution (or bitdepth).