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Linear PCM

Pulse-code modulation
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
Extended from PCM

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 regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.

Linear pulse-code modulation (LPCM) is a specific type of PCM where the quantization levels are linearly uniform. This is in contrast to PCM encodings where 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.

Early electrical communications started to sample signals in order to interlace samples from multiple telegraphy sources and to convey them over a single telegraph cable. The American inventor Moses G. Farmer conveyed telegraph time-division multiplexing (TDM) as early as 1853. Electrical engineer W. M. Miner, in 1903, used an electro-mechanical commutator for time-division multiplexing multiple telegraph signals; he also applied this technology to telephony. He obtained intelligible speech from channels sampled at a rate above 3500–4300 Hz; lower rates proved unsatisfactory. This was TDM, but pulse-amplitude modulation (PAM) rather than PCM.


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