PCM脉冲编码中英文翻译1份.doc

英文原文pulse-code modulationpulse-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) codepcm has been used in digital telephone systems and 1980s-era electronic musical keyboardsit is also the standard form for digital audio in computers and the compact disc red book formatit is also standard in digital video，for example，using itu-r bt601uncompressed 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 highmodulation in the diagram，a sine wave (red curve) is sampled and quantized for pulse code modulationthe sine wave is sampled at regular intervals，shown as ticks on the x-axisfor 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 manipulationfor 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，etcencoding these values as binary numbers would result in the following set of nibbles:0111，1001，1011，1100，1101，1110，1110，1111，1111，1111，1110，etcthese digital values could then be further processed or analyzed by a purpose-specific digital signal processor or general purpose cpuseveral pulse code modulation streams could also be multiplexed into a larger aggregate data stream，generally for transmission of multiple streams over a single physical linkone technique is called time-division multiplexing，or tdm，and is widely used，notably in the modern public telephone systemanother technique is called frequency-division multiplexing，where the signal is assigned a frequency in a spectrum，and transmitted along with other signals inside that spectrumcurrently，tdm is much more widely used than fdm because of its natural compatibility with digital communication，and generally lower bandwidth requirementsthere are many ways to implement a real device that performs this taskin 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 sortdemodulation to produce output from the sampled data，the procedure of modulation is applied in reverseafter each sampling period has passed，the next value is read and a signal is shifted to the new valueas a result of these transitions，the signal will have a significant amount of high-frequency energyto 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，converting the signal from digital to analog at a higher sample rate such that the analog filter required for anti-aliasing is much simplerin some systems，no explicit filtering is done at all; as its 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 precisionthe 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 bandsthe electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signalthese devices are dacs (digital-to-analog converters)，and operate similarly to adcsthey produce on their output a voltage or current (depending on type) that represents the value presented on their inputsthis output would then generally be filtered and amplified for uselimitationsthere are two sources of impairment implicit in any pcm system: 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 digitizedonce 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 codingolder 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 domainthese simple techniques have been largely rendered obsolete by modern transform-based audio compression techniquesdpcm encodes the pcm values as differences between the current and the predicted valuean algorithm predicts the next sample based on the previous samples，and the encoder stores only the difference between this prediction and the actual valueif the prediction is reasonable，fewer bits can be used to represent the same informationfor audio，this type of encoding reduces the number of bits required per sample by about 25% compared to pcmadaptive dpcm (adpcm) is a variant of dpcm that varies the size of the quantization step，to allow further reduction of the required bandwidth for a given signal-to-noise ratiodelta modulation is a form of dpcm which uses one bit per samplein telephony，a standard audio signal for a single phone call is encoded as 8000 analog samples per second，of 8 bits each，giving a 64 kbit/s digital signal known as ds0the default signal compression encoding on a ds0 is either -law (mu-law) pcm (north america and japan) 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 valuethis system is described by international standard g711an alternative proposal for a floating point representation，with 5-bit mantissa and 3-bit radix，was abandonedwhere circuit costs are high and loss of voice quality is acceptable，it sometimes makes sense to compress the voice signal even furtheran adpcm algorithm is used to map a series of 8-bit -law or a-law pcm samples into a series of 4-bit adpcm samplesin this way，the capacity of the line is doubledthe technique is detailed in the g726 standardlater it was found that even further compression was possible and additional standards were publishedsome of these international standards describe systems and ideas which are covered by privately owned patents and thus use of these standards requires payments to the patent holderssome adpcm techniques are used in voice over ip communicationsencoding for transmission pulse-code modulation can be either return-to-zero (rz) or non-return-to-zero (nrz)for a nrz system to be synchronized using in-band information，there must not be long sequences of identical symbols，such as ones or zeroesfor binary pcm systems，the density of 1-symbols is called ones-densityones-density is often controlled using precoding techniques such as run length limited encoding，where the pcm code is expanded into a slightly longer code with a guaranteed bound on ones-density before modulation into the channelin other cases，extra framing bits are added into the stream which guarantee at least occasional symbol transitionsanother technique used to control ones-density is the use of a scrambler polynomial on the raw data which will tend to turn the raw data stream into a stream that looks pseudo-random，but where the raw stream can be recovered exactly by reversing the effect of the polynomial in this case，long runs of zeroes or ones are still possible on the output， but are considered unlikely enough to be within normal engineering tolerancein other cases， the long term dc value of the modulated signal is important，as building up a dc offset will tend to bias detector circuits out of their operating rangein this case special measures are taken to keep a count of the cumulative dc offset，and to modify the codes if necessary to make the dc offset always tend back to zeromany of these codes are bipolar codes， where the pulses can be positive，negative or absentin the typical alternate mark inversion code，non-zero pulses alternate between being positive and negativethese rules may be violated to generate special symbols used for framing or other special purposes中文译文脉冲编码调制pulse-code modulation脉冲编码调制（pulse-code modulation，pcm）是一种模拟讯号的数码化方法。pcm将讯号的强度依照同样的间距分成数段，然后用独特的数码记号（通常是二进制）来量化。pcm常被用于数码电信系统上，也是电脑和cd红皮书中的标准形式。在数码视讯中它也是标准，例如使用 itu-r bt.601。但是pcm并不流行于诸如dvd或dvr的消费性商品上，因为它需要相当大的位元率（dvd格式虽然支援pcm，不过很少使用）；与之相较，压缩过的音讯较符合效率。不过，许多蓝光光碟使用pcm作音讯编码。非常频繁地，pcm编码以一种序列通讯的形式，使数码传讯由一点至下一点变得更容易不论在已给定的系统内，或物理位置。调制 模拟讯号转换至4-bit pcm的取样和量化在图示中，一个正弦波（红色曲线）被取样和量化为pcm。正弦波在每段固定时间内被取一次样，即x轴的刻度。而每一个样本则依照某种运算法（在这个例子中是ceiling function），选定它们在y轴上的位置。这样便产生完全离散的输入讯号的替代物，很容易编码成为数码资料，以作保存或操纵。以图为例，很清楚看出样本为9、11、12、13、14、14、15、15、15、14等，将它们以二进制编码，就得到一组一组的数字：1001、1011、1100、1101、1110、1111、1111、1111、1110等，这些数码资料之后就可以被特定用途的dsp或者一般的cpu所处理。有一些pcm资料流可以和较大的聚合资料流作多工传输（multiplex），通常在物理层传输资料时都会这么作。这个技术称作 分时多工 time-division multiplexing（tdm），非常广泛地使用，例如现代的公共电话系统。有许多方法可以内置一个处理调制的真实装置。在真实系统中，这种装置一般被放在单一个芯片中，并搭配一个振荡器，称作“模拟至数码转换器（analog-to-digital converter，adc）”。这些装置透过振荡器触动输入讯号的接受，并且输出数码化的讯号至某种处理器。解调从数码讯号回制成模拟讯号的过程，就如同把调制的过程逆转一样，称作解调制（demodulation）。在理想的系统上，每经过取样的固定时间而读取新的资料时，输出会即时改变到该强度。经过这样的即时转换，离散的讯号本质上会有大量的高频率能量，出现与取样频率的倍数相关的谐波（见方波）。要消灭这些谐波并使讯号流畅，讯号必须通过一些模拟滤波器，压制任何在预期频域外的人造物（例如大于的频域，这是理论上最高的分辨率）。有些系统使用数码滤波器来移除最低和最高的谐波，而在有些系统中不使用任何外部的滤波器，因为不可能有系统重制出无限大的带宽，系统本身的不足补足了讯号重制上的瑕疵，或者该系统根本就不要求准确度。取样原理说明，任何一种pcm装置，只要提供相对于输入讯号足够大的取样频率，在期望频域中就不会有显著的失真因素。从离散的资料重制回模拟讯号所使用的电子学，与从模拟至数码是相似的。这些装置被称作“模拟至数码转换器（digital-to-analog converters, dac）”，与adc的运作相似。它们依照输入的数码讯号，输出电压或电流（看情况则种类不同），这个输出然后经过滤波器和放大器，达成回放。限制可注意的是，在任何pcm系统中，本质上有两种损害的来源：在量化时，取样必须迫于选择接近哪一个整数值（即量化误差）。 在样本与样本之间没有任何资料，根据取样原理，这代表任何频率大于或等于fs(即取样频率）的讯号，会产生失真，或者完全消失（aliasing error）。这又称作nyquist frequency。 由于所有样本都依据时间取样，重制时至关重要的便是一个准确的振荡器。如果编码或解码时，任何一方的振荡器不稳定，频率漂移就会使输出装置的品质降低。如果两方的频率具有些微的差异，稳定的误差对于品质而言并非巨大的问题。但一旦振荡器并非稳定的（即脉动的间距不相等），不论是音讯或者视讯上，都将造成巨大的失真。数字化在一般的pcm中，模拟讯号在数码化之前会经过一些处理（如振幅压缩）。一旦经数码化，pcm讯号通常会再进一步处理（如数码资料压缩）。有些形式的pcm把讯号处理结合在编码过程中。老一点的系统会把讯号的处理放在模拟回路中，当作模拟至数码转换（a/d）的一部分，新的系统则放在数码回路中。不过由于现代基础于转换的音讯压缩技术，这些简单的技术大部分已被认为过时。differential（差异）