Pulse-code modulation
Pulse-code modulation (PCM) is a method used to digitally represent sampled 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.
| Filename extension |
.L16, .WAV, .AIFF, .AU, .PCM |
|---|---|
| Internet media type |
audio/L16, audio/L8, audio/L20, audio/L24 |
| Type code | "AIFF" for L16, none |
| 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 |
| Passband modulation |
|---|
| Analog modulation |
| Digital modulation |
| Hierarchical modulation |
| Spread spectrum |
| See also |
|
Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform. 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.