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Pulse code modulation (PCM) is a digital representation of an analog signal that takes samples of the amplitude of the analog signal at regular intervals. The sampled analog data is changed to, and then represented by, binary data. PCM requires a very accurate clock. The number of samples per second, ranging from 8,000 to 192,000, is usually several times the maximum frequency of the analog waveform in Hertz (Hz), or cycles per second, which ranges from 8 to 192 KHz.
The word pulse refers to pulses found in transmission lines, which are a natural consequence of two other almost simultaneously evolved analog methods: pulse width modulation and pulse position modulation, where each uses discrete signal pulses of varying widths or positions. Otherwise, PCM has little similarity to these other forms of signal encoding. These methodologies were introduced to the U.S. in the early 1960s as telephone companies began converting voice to digital signals to facilitate transmission between cities.
Each sample in a PCM is quantized, approximating a very large set of possible values by a relatively small set of values, which may be integers or even discrete symbols. No matter how complex they are, all analog data may be digitized. This includes analog data such as full-motion video, sound, telemetry and virtual reality.
PCM data is actually raw digital audio samples. Audio files, in formats such as MP3 and AAC, are first converted to PCM data. Then, the PCM data is converted to analog signals for the speakers.
Further processing by digital signal processors may create many streams of data. These streams, in turn, may be multiplexed into larger streams of data transmitted very rapidly over long distances by processes such as time-division multiplexing, frequency division multiplexing, and others. TDM is used more widely because of its natural compatibility with digital communication and its lower bandwidth requirement.
After these data streams reach their destination, they are demultiplexed, broken back down into individual data streams, and demodulated, whereby the modulation procedure is applied in reverse to recreate the original binary numbers. These are further processed to restore the original analog waveform. In the process of transitioning from one sampling period to the next, the signal gains significant high-frequency energy. Analog filters are used to smooth out the signal and remove these undesirable frequencies, called aliasing frequencies. Depending on the requirement for precise output signals, these analog filters may or may not be necessary.