Filename extension |
.L16, .WAV, .AIFF, .AU, .PCM[1] |
---|---|
Internet media type | |
Type code | "AIFF" for L16,[1] none[3] |
Magic number | Varies |
Type of format | Uncompressed audio |
Contained by | Audio CD, AES3, WAV, AIFF, AU, M2TS, VOB, and many others |
Open format? | Yes |
Free format? | Yes[5] |
Passband modulation |
---|
Analog modulation |
Digital modulation |
Hierarchical modulation |
Spread spectrum |
See also |
Pulse-code modulation (PCM) is a method used to digitally represent analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Alec Reeves, Claude Shannon, Barney Oliver and John R. Pierce are credited with its invention.[6][7][8]
Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform.[5] This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm). Though PCM is a more general term, it is often used to describe data encoded as LPCM.
A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that can be used to represent each sample.
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