英文原文
Pulse-code modulation
Pulse-code modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals， then quantized to a series of symbols in a numeric (usually binary) code．PCM has been used in digital telephone systems and 1980s-era electronic musical keyboards．It is also the standard form for digital audio in computers and the compact disc "red book" format．It is also standard in digital video，for example，using ITU-R BT．601．Uncompressed PCM is not typically used for video in standard definition consumer applications such as DVD or DVR because the bit rate required is far too high．
Modulation
In the diagram，a sine wave (red curve) is sampled and quantized for pulse code modulation．The sine wave is sampled at regular intervals，shown as ticks on the x-axis．For each sample，one of the available values (ticks on the y-axis) is chosen by some algorithm (in this case，the floor function is used)．This produces a fully discrete representation of the input signal (shaded area) that can be easily encoded as digital data
for storage or manipulation．For the sine wave example at right，we can verify that the quantized values at the sampling moments are 7，9，11，12，13，14，14，15，15，15，14，etc．
Encoding these values as binary numbers would result in the following set of nibbles:0111，1001，1011，1100，1101，1110，1110，1111，1111，1111，1110，etc．These digital values could then be further processed or analyzed by a purpose-specific digital signal processor or general purpose CPU．Several Pulse Code Modulation streams could also be multiplexed into a larger aggregate data stream，generally for transmission of multiple streams over a single physical link．One technique is called time-division multiplexing，or TDM，and is widely used，notably in the modern public telephone system．Another technique is called Frequency-division multiplexing，where the signal is assigned a frequency in a spectrum，and transmitted along with other signals inside that spectrum．Currently，TDM is much more widely used than FDM because of its natural compatibility with digital communication，and generally lower bandwidth requirements．
There are many ways to implement a real device that performs this task．In real systems，such a device is commonly implemented on a single integrated circuit that lacks only the clock necessary for sampling，and is generally referred to as an ADC (Analog-to-Digital converter)．These devices will produce on their output a binary representation of the input whenever they are triggered by a clock signal，which would then be read by a processor of some sort．
Demodulation
To produce output from the sampled data，the procedure of modulation is applied in reverse．After each sampling period has passed，the next value is read and a signal is shifted to the new value．As a result of these transitions，the signal will have a significant amount of high-frequency energy．To smooth out the signal and remove these undesirable aliasing frequencies，the signal would be passed through analog filters that suppress energy outside the expected frequency range (that is，greater than the Nyquist frequency fs/2)．Some systems use digital filtering to remove some of the aliasing，conv
erting the signal from digital to analog at a higher sample rate such that the analog filter required for anti-aliasing is much simpler．In some systems，no explicit filtering is done at all; as it's impossible for any system to reproduce a signal with infinite bandwidth，inherent losses in the system compensate for the artifacts-or the system simply does not require much precision．The sampling theorem suggests that practical PCM devices，provided a sampling frequency that is sufficiently greater than that of the input signal，can operate without introducing significant distortions within their designed frequency bands．
The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signal．These devices are DACs (digital-to-analog converters)，and operate similarly to ADCs．They produce on their output a voltage or current (depending on type) that represents the value presented on their inputs．This output would then generally be filtered and amplified for use．
Limitations
There are two sources of impairment implicit in any PCM system:
transform中文翻译Choosing a discrete value near the analog signal for each sample ( quantization error ) Between samples no measurement of the signal is made; due to the sampling theorem this results in any frequency above or equal to( Fs being the sampling frequency) being distorted or lost completely ( aliasing error).  (One half the sampling frequencies are known as the Nyquist frequency.)
Digitization as part of the PCM process
In conventional PCM，the analog signal may be processed (eg by amplitude compression)before being digitized．Once the signal is digitized，the PCM signal is usually subjected to further processing (eg digital data compression)．
PCM with linear quantization is known as Linear PCM (LPCM)．
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 i
mplementations do so in the digital domain．These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques．
DPCM encodes the PCM values as differences between the current and the predicted value．An algorithm predicts the next sample based on the previous samples，and the encoder stores only the difference between this prediction and the actual value．If the prediction is reasonable，fewer bits can be used to represent the same information．For audio，this type of encoding reduces the number of bits required per sample by about 25% compared to PCM．

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