PCM is an abbreviation for "Pulse Code Modulation". PCM is how digital audio is typically represented in a computer.
The audio signal is represented by samples of its instantaneous amplitude taken at regular intervals (the sample period which is the inverse of the sampling frequency). The representation of each sample can take several forms. On CDs, a 16-bit integer is used. Plain old telephony uses 8 bits with a non-linear coding (either A-law or u-law). Studio equipment often used 24 or more bits. Floating point representation is also possible. Alsa can use the following formats: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE FLOAT64_LE FLOAT64_BE IEC958_SUBFRAME_LE IEC958_SUBFRAME_BE MU_LAW A_LAW IMA_ADPCM MPEG GSM SPECIAL S24_3LE S24_3BE U24_3LE U24_3BE S20_3LE S20_3BE U20_3LE U20_3BE S18_3LE S18_3BE U18_3LE
Other representations than PCM are possible. For example it is not necessary to have a one-to-one correspondance between the samples of the audio signal and the numbers in the data stream. (Example: mp3)
Retrieved from "http://alsa.opensrc.org/PCM"