This computer science lesson describes how sound is digitally encoded and stored by a computer. It begins with a discussion of the nature of sound in air, namely, a longitudinal wave of compressions and rarefactions. It then explains how sound can be captured by a dynamic microphone, which generates an analogue electrical signal that represents the original analogue sound wave. The digitisation of sound is then described, a process known as analogue to digital conversion. The effect of the sample rate on the quality of a digital sound recording is illustrated, as well as its effect on the size of the sound file. The importance of the bit depth used to digitise a sound wave is also covered, along with the impact of quantisation errors that might result from a low bit depth. This video tutorial mentions some typical sample rates such as speech quality (8KHz) which is used by VIOP applications and telephone voice calls, and CD quality (44.1 KHz) which is used for music recordings. The video concludes with a mention of some different audio file formats including wrappers for uncompressed LPCM data (linear pulse code modulation) such as .WAV and .AIFF, and some compressed audio file formats such as MP3, WMA and M4A.

Chapters:
00:00 The nature of sound
01:22 A microphone to capture sound
01:50 Representing sound with a transverse wave
03:12 Sample rate
07:47 Bit depth
09:51 Summary