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PAM

(1) (Pulse Amplitude Modulation) The conversion of audio wave samples to pulses (voltages). PAM is the first step in pulse code modulation (PCM), which is followed by converting the pulses to digital numbers. See PCM.

(2) (Pluggable Authentication Modules) A programming interface that enables third-party security methods to be used in Unix. For example, smart cards, Kerberos and RSA technologies can be integrated with various Unix functions such as rlogin, telnet and ftp.



PCM

(1) See phase change memory.

(2) See also PMC (programmable metallization cell).

(3) (Plug Compatible Manufacturer) An organization that makes a computer or electronic device that is compatible with an existing machine.

(4) (Pulse Code Modulation) The primary way analog audio signals are converted into digital form by taking samples of the waveforms from 8 to 192 thousand times per second (8 to 192 kHz) and recording each sample as a digital number from 8 to 24 bits long (see sampling). PCM data are pure digital audio samples, and they are the underlying data in several music file formats (see WAV, FLAC and AIFF).

Sound Cards Support PCM
The audio-out port on a sound card provides an analog signal to the speakers; however, compressed formats such as MP3 and AAC are converted to PCM, and the PCM data are converted to analog (see D/A converter). Sound cards may also output PCM and other digital signals such as Dolby Digital (see S/PDIF). With regard to input, an analog microphone is plugged into the audio-in port, and the sound card converts the analog signals to PCM.

PCM Ports on A/V Equipment
When ports on set-top boxes and Blu-ray/DVD players are labeled PCM or linear PCM (LPCM), they refer to uncompressed audio channels rather than encoded formats such as Dolby Digital, TrueHD, DTS and DTS-HD. PCM can be mono, stereo or have multiple channels for surround sound. See Bitstream mode and linear PCM.

It Started With the Telcos
PCM was introduced in the U.S. in the early 1960s when the telephone companies began converting voice to digital for transport over intercity trunks. See mu-Law.








It Starts as PCM
PCM is the primary way analog waves are converted into digital form for voice conversations as well as music. Codecs such as MP3 and AAC that compress the digital data further apply algorithms to the PCM samples in order to eliminate overlapping frequencies as well as sounds that are deemed inaudible to the human ear.






Another Approach to Sampling
Sony's SACD audio format uses Direct Stream Digital (DSD), a dramatic departure from PCM. Instead of turning samples into a number with a range of values, DSD samples are only 1-bit long (0 or 1), depending on whether the wave is moving up or down from the previous sample point (see DSD).






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