Web Design in a Nutshell, 3rd Edition

Basic Digital Audio Concepts

To distribute recorded speech or music over the Internet, an analog signal must be converted to digital information (described by bits and bytes). This process is called encoding . It is analogous to scanning a photograph to a digital bitmap format, and many of the same concepts regarding quality and file size apply. Some audio file formats (such as MPEG) are compressed in size during encoding using a specialized audio compression algorithm to save disk space. In the encoding process, you may be asked to provide settings for the following aspects of the audio file.

Sampling rate

To convert an analog sound wave into a digital description of that wave, samples of the wave are taken at timed intervals (see Figure 33-1). The number of samples taken per second is called the sampling rate. The more samples taken per second, the more accurately the digital description can recreate the original shape of the sound wave, and therefore the better the quality of the digital audio. In this respect, sampling rate is similar to image resolution for digital images.

Sample rates are typically measured in kilohertz (kHz). On the high end, CD-quality audio has a sampling rate of 44.1 kHz (or 44,100 samples per second). On the low end, 8 kHz produces a grainy sound quality that is equivalent to a transistor radio. Standard sampling rates include 8 kHz, 11.025 kHz, 11.127 kHz, 22.05 kHz, 44.1 kHz, and 48 kHz. The high-end standard is 96K, which may be seen in DVD audio but ...