

A sound file is an individual sound recording of any length, whose individual parts may or may not have been recorded at different times. With analogue reel-to-reel tape, these recordings will be separated by leader tape, and may be album tracks, takes from a studio session or short sound effects. Similarly, digitally recorded sounds are stored as sound files of different lengths, but one advantage over analogue storage is that the random- and direct-access nature of the storage medium means that one individual sound file has no specific time relationship to another. Older recordings will be found nearer the start of an analogue reel, but this is not the case with digital recordings, since digital audio is stored non-linearly.
During the late 1970s and early 80s, several sites developed UNIX-based sound file systems for the benefit of storing computer music. The formats were generally different to the standard UNIX file system and so required separate disks (or disk partitions) for sound storage.
The root of most research into audio file formats is the csound file system, which first appeared in 1980, and was developed by D. Gareth Loy at the Computer Research Lab (CARL) at UC San Diego. CARL music software distribution contained various tools such as vocoders, configurable reverberators, as well as software such as Csound and the cmusic scripting language (a simple C-based language descended from the Music V language, and written by F Richard Moore).
In the late 1980s, the several variations of software based on csound were merged into the BICSF (Berkeley/IRCAM /CARL Sound File) System and adapted to allow compatibility and interoperability with other computer systems such as NeXt and SPARC.
A standard digital audio file is recognised by its .au extension, and unlike many computer files, is platform-independent, which means that it can be read by any computer with the appropriate software. The main disadvantage of digital audio files is that they consume the largest amount of disk space since they comprise of pure PCM (Pulse Code Modulation), i.e. raw digital data, which makes them the least appropriate format for delivering audio across the Internet .