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Home > Pulse-code modulation


 

Pulse-code modulation (PCM) is a modulation technique. It is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals. Every sample is quantized to a series of symbols in a digital code, which is usually a binary code. PCM is used in digital telephone systems. It is also the standard form for digital audio in computers and various compact disc formats.

Several PCM streams may be multiplexed into a larger aggregate data stream. This technique is called Time-Division Multiplexing, or TDM. TDM was invented by the telephone industry, but today the technique is an integral part of many digital audio workstations such as Pro Tools.


1 Digitization as part of the PCM process

In conventional PCM, the analog signal may be processed (e.g. by amplitude compression) before being digitized. Once the signal is digitized, the PCM signal is not subjected to further processing (e.g. digital data compressionIn computer science, data compression is the process of encoding information using fewer bits, or information units, thanks to specific encoding schemes. For example, this article could be encoded with fewer bits if we accept the convention that the word).

Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog domain as part of the A/D process, newer implementations do so in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-based signal compressionIn telecommunication, the term signal compression has the following meanings: In analog (usually audio) systems, reduction of the dynamic range of a signal by controlling it as a function of the inverse relationship of its instantaneous value relative to techniques.

The default encoding on a DS0 is either mu-law PCM (North America) or a-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 12 or 13 bit linear PCM sample number is mapped into an 8 bit value. This system is described by international standard G.711.

Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm is used to map a series of 8 bit PCM samples into a series of 4 bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard.

Later it was found that even further compression was possible and additional standards were published. Some of these international standards describe systems and ideas which are covered by privately owned patents and thus to use these standards requires payments to the patent holders.

Some ADPCM techniques are used in Voice over IPIP Telephony also called Internet telephony is the technology that makes it possible to have a telephone conversation over the Internet or a dedicated Internet Protocol (IP) network instead of dedicated voice transmission lines. This allows the eliminatio communications.



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