PC Hardware in a Nutshell, 3rd Edition

Waveform Audio

Waveform audio files are created using a process called sampling or digitizing to convert analog sound to digital format. Sampling takes periodic snapshots, or samples, of the instantaneous state of the analog signal, encodes the data, and stores the audio in digital form. Just as digital images can be stored at different resolutions according to their intended use, audio data can be stored at different resolutions to trade off sound quality against file size. Five parameters determine the quality of digital sound files and how much space they occupy:

Sample size

Sample size specifies how much data is stored for each sample. A larger sample size stores more information about each sample, contributing to higher sound quality. Sample size is specified as the number of bits stored for each sample. CD audio, for example, uses 16-bit samples, which allow the waveform amplitude to be specified as one of 65,536 discrete values. All sound cards support at least 16-bit samples.

Sampling rate

Sampling rate specifies how often samples are taken. Sampling rate is specified in Hz (Hertz, or cycles/second) or kHz (kilohertz, 1000 Hertz). Higher-frequency data inherently changes more often. Changes that occur between samples are lost, so the sampling rate determines the highest-frequency sounds that can be sampled. Two samples are required to capture a change, so the highest frequency that can be sampled, called the Nyquist frequency, is half the sampling rate. For example, ...