PCM脈沖編碼中英文翻譯1份

1、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

2、 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

3、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) i

4、s 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

5、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

6、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 mo

7、dern 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 digita

8、l 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

9、 (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

10、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 woul

11、d 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 fi

12、lter 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 sampl

13、ing 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 t

14、he 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 th

15、en 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 resu

16、lts 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

17、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 sys

18、tems 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 curre

19、nt 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

20、 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

21、 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 (e

22、urope 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-bi

23、t 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 th

24、e 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 stan

25、dards 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

26、 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 bou

27、nd 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 d

28、ata 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 othe

29、r 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 offs

30、et 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 o

31、ther special purposes中文譯文脈沖編碼調(diào)制pulse-code modulation脈沖編碼調(diào)制（pulse-code modulation，pcm）是一種模擬訊號的數(shù)碼化方法。pcm將訊號的強度依照同樣的間距分成數(shù)段，然后用獨特的數(shù)碼記號（通常是二進制）來量化。pcm常被用于數(shù)碼電信系統(tǒng)上，也是電腦和cd紅皮書中的標準形式。在數(shù)碼視訊中它也是標準，例如使用 itu-r bt.601。但是pcm并不流行于諸如dvd或dvr的消費性商品上，因為它需要相當大的位元率（dvd格式雖然支援pcm，不過很少使用）；與之相較，壓縮過的音訊較符合效率。不過，許多藍光光碟使用pcm作音訊編

32、碼。非常頻繁地，pcm編碼以一種序列通訊的形式，使數(shù)碼傳訊由一點至下一點變得更容易不論在已給定的系統(tǒng)內(nèi)，或物理位置。調(diào)制 模擬訊號轉換至4-bit pcm的取樣和量化在圖示中，一個正弦波（紅色曲線）被取樣和量化為pcm。正弦波在每段固定時間內(nèi)被取一次樣，即x軸的刻度。而每一個樣本則依照某種運算法（在這個例子中是ceiling function），選定它們在y軸上的位置。這樣便產(chǎn)生完全離散的輸入訊號的替代物，很容易編碼成為數(shù)碼資料，以作保存或操縱。以圖為例，很清楚看出樣本為9、11、12、13、14、14、15、15、15、14等，將它們以二進制編碼，就得到一組一組的數(shù)字：1001、1011、1

33、100、1101、1110、1111、1111、1111、1110等，這些數(shù)碼資料之后就可以被特定用途的dsp或者一般的cpu所處理。有一些pcm資料流可以和較大的聚合資料流作多工傳輸（multiplex），通常在物理層傳輸資料時都會這么作。這個技術稱作 分時多工 time-division multiplexing（tdm），非常廣泛地使用，例如現(xiàn)代的公共電話系統(tǒng)。有許多方法可以內(nèi)置一個處理調(diào)制的真實裝置。在真實系統(tǒng)中，這種裝置一般被放在單一個芯片中，并搭配一個振蕩器，稱作“模擬至數(shù)碼轉換器（analog-to-digital converter，adc）”。這些裝置透過振蕩器觸動輸入訊號的

34、接受，并且輸出數(shù)碼化的訊號至某種處理器。解調(diào)從數(shù)碼訊號回制成模擬訊號的過程，就如同把調(diào)制的過程逆轉一樣，稱作解調(diào)制（demodulation）。在理想的系統(tǒng)上，每經(jīng)過取樣的固定時間而讀取新的資料時，輸出會即時改變到該強度。經(jīng)過這樣的即時轉換，離散的訊號本質上會有大量的高頻率能量，出現(xiàn)與取樣頻率的倍數(shù)相關的諧波（見方波）。要消滅這些諧波并使訊號流暢，訊號必須通過一些模擬濾波器，壓制任何在預期頻域外的人造物（例如大于的頻域，這是理論上最高的分辨率）。有些系統(tǒng)使用數(shù)碼濾波器來移除最低和最高的諧波，而在有些系統(tǒng)中不使用任何外部的濾波器，因為不可能有系統(tǒng)重制出無限大的帶寬，系統(tǒng)本身的不足補足了訊號重制上

35、的瑕疵，或者該系統(tǒng)根本就不要求準確度。取樣原理說明，任何一種pcm裝置，只要提供相對于輸入訊號足夠大的取樣頻率，在期望頻域中就不會有顯著的失真因素。從離散的資料重制回模擬訊號所使用的電子學，與從模擬至數(shù)碼是相似的。這些裝置被稱作“模擬至數(shù)碼轉換器（digital-to-analog converters, dac）”，與adc的運作相似。它們依照輸入的數(shù)碼訊號，輸出電壓或電流（看情況則種類不同），這個輸出然后經(jīng)過濾波器和放大器，達成回放。限制可注意的是，在任何pcm系統(tǒng)中，本質上有兩種損害的來源：在量化時，取樣必須迫于選擇接近哪一個整數(shù)值（即量化誤差）。 在樣本與樣本之間沒有任何資料，根據(jù)取樣

36、原理，這代表任何頻率大于或等于fs(即取樣頻率）的訊號，會產(chǎn)生失真，或者完全消失（aliasing error）。這又稱作nyquist frequency。 由于所有樣本都依據(jù)時間取樣，重制時至關重要的便是一個準確的振蕩器。如果編碼或解碼時，任何一方的振蕩器不穩(wěn)定，頻率漂移就會使輸出裝置的品質降低。如果兩方的頻率具有些微的差異，穩(wěn)定的誤差對于品質而言并非巨大的問題。但一旦振蕩器并非穩(wěn)定的（即脈動的間距不相等），不論是音訊或者視訊上，都將造成巨大的失真。數(shù)字化在一般的pcm中，模擬訊號在數(shù)碼化之前會經(jīng)過一些處理（如振幅壓縮）。一旦經(jīng)數(shù)碼化，pcm訊號通常會再進一步處理（如數(shù)碼資料壓縮）。有些形式的pcm把訊號處理結合在編碼過程中。老一點的系統(tǒng)會把訊號的處理放在模擬回路中，當作模擬至數(shù)碼轉換（a/d）的一部分，新的系統(tǒng)則放在數(shù)碼回路中。不過由于現(xiàn)代基礎于轉換的音訊壓縮技術，這些簡單的技術大部分已被認為過時。differential（差異）