| Pulse-code modulation | |
|---|---|
| Filename extension | .L16, .WAV, .AIFF, .AU, .PCM[1] |
| Internet media type | |
| Type code | "AIFF" for L16,[1] none[3] |
| Magic number | Varies |
| Type of format | Uncompressedaudio |
| Contained by | Audio CD,AES3,WAV,AIFF,AU,M2TS,VOB, and many others |
| Open format? | Yes |
| Free format? | Yes[5] |
| Passbandmodulation |
|---|
 |
| Analog modulation |
| Digital modulation |
| Hierarchical modulation |
| Spread spectrum |
| See also |
Pulse-code modulation (PCM) is a method used todigitally representanalog signals. It is the standard form ofdigital audio in computers,compact discs,digital telephony and other digital audio applications. In a PCMstream, theamplitude of the analog signal issampled at uniform intervals, and each sample isquantized to the nearest value within a range of digital steps. Shannon, Oliver, and Pierce were inducted into the National Inventors Hall of Fame for their PCM patent granted in 1952.[6][7][8]
Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform.[5] This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with theA-law algorithm or theμ-law algorithm). ThoughPCM is a more general term, it is often used to describe data encoded as LPCM.
A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: thesampling rate, which is the number of times per second that samples are taken; and thebit depth, which determines the number of possible digital values that can be used to represent each sample.
Early electrical communications started tosample signals in order tomultiplex samples from multipletelegraphy sources and to convey them over a single telegraph cable. The American inventorMoses G. Farmer conceived telegraphtime-division multiplexing (TDM) as early as 1853. Electrical engineer W. M. Miner, in 1903, used an electro-mechanicalcommutator for time-division multiplexing multiple telegraph signals; he also applied this technology totelephony. He obtained intelligible speech from channels sampled at a rate above 3500–4300 Hz; lower rates proved unsatisfactory.
In 1920, theBartlane cable picture transmission system used telegraph signaling of characters punched in paper tape to send samples of imagesquantized to 5 levels.[9] In 1926, Paul M. Rainey ofWestern Electric patented afacsimile machine that transmitted its signal using 5-bit PCM, encoded by an opto-mechanicalanalog-to-digital converter.[10] The machine did not go into production.[11]
British engineerAlec Reeves, unaware of previous work, conceived the use of PCM for voice communication in 1937 while working forInternational Telephone and Telegraph in France. He described the theory and its advantages, but no practical application resulted. Reeves filed for a French patent in 1938, and his US patent was granted in 1943.[12] By this time Reeves had started working at theTelecommunications Research Establishment.[11]
The first transmission ofspeech by digital techniques, theSIGSALY encryption equipment, conveyed high-levelAllied communications duringWorld War II. In 1949, for the Canadian Navy'sDATAR system,Ferranti Canada built a working PCM radio system that was able to transmit digitized radar data over long distances.[13]
PCM in the late 1940s and early 1950s used acathode-raycoding tube with aplate electrode having encoding perforations.[14] As in anoscilloscope, the beam was swept horizontally at the sample rate while the vertical deflection was controlled by the input analog signal, causing the beam to pass through higher or lower portions of the perforated plate. The plate collected or passed the beam, producing current variations in binary code, one bit at a time. Rather than natural binary, the grid of Goodall's later tube was perforated to produce a glitch-freeGray code and produced all bits simultaneously by using a fan beam instead of a scanning beam.[15]
In the United States, theNational Inventors Hall of Fame has honoredBernard M. Oliver[16]andClaude Shannon[17]as the inventors of PCM,[18]as described in "Communication System Employing Pulse Code Modulation",U.S. patent 2,801,281 filed in 1946 and 1952, granted in 1956. Another patent by the same title was filed byJohn R. Pierce in 1945, and issued in 1948:U.S. patent 2,437,707. The three of them published "The Philosophy of PCM" in 1948.[19]
TheT-carrier system, introduced in 1961, uses two twisted-pair transmission lines to carry 24 PCMtelephone calls sampled at 8 kHz and 8-bit resolution. This development improved capacity and call quality compared to the previousfrequency-division multiplexing schemes.
In 1973,adaptive differential pulse-code modulation (ADPCM) was developed, by P. Cummiskey,Nikil Jayant andJames L. Flanagan.[20]
In 1967, the first PCM recorder was developed byNHK's research facilities in Japan.[21] The 30 kHz 12-bit device used acompander (similar toDBX Noise Reduction) to extend the dynamic range, and stored the signals on avideo tape recorder. In 1969, NHK expanded the system's capabilities to 2-channelstereo and 32 kHz 13-bit resolution. In January 1971, using NHK's PCM recording system, engineers atDenon recorded the first commercial digital recordings.[note 1][21]
In 1972, Denon unveiled the first 8-channel digital recorder, the DN-023R, which used a 4-head open reel broadcast video tape recorder to record in 47.25 kHz, 13-bit PCM audio.[note 2] In 1977, Denon developed the portable PCM recording system, the DN-034R. Like the DN-023R, it recorded 8 channels at 47.25 kHz, but it used 14-bits "withemphasis, making it equivalent to 15.5 bits."[21]
In 1979, the first digital pop album,Bop till You Drop, was recorded. It was recorded in 50 kHz, 16-bit linear PCM using a 3M digital tape recorder.[22]
Thecompact disc (CD) brought PCM to consumer audio applications with its introduction in 1982. The CD uses a44,100 Hz sampling frequency and 16-bit resolution and stores up to 80 minutes of stereo audio per disc.
The rapid development and wide adoption of PCMdigital telephony was enabled bymetal–oxide–semiconductor (MOS)switched capacitor (SC) circuit technology, developed in the early 1970s.[23] This led to the development of PCM codec-filter chips in the late 1970s.[23][24] Thesilicon-gateCMOS (complementary MOS) PCM codec-filter chip, developed byDavid A. Hodges and W.C. Black in 1980,[23] has since been the industry standard for digital telephony.[23][24] By the 1990s,telecommunication networks such as thepublic switched telephone network (PSTN) had been largelydigitized withvery-large-scale integration (VLSI) CMOS PCM codec-filters, widely used inelectronic switching systems fortelephone exchanges, user-endmodems and a wide range ofdigital transmission applications such as theintegrated services digital network (ISDN),cordless telephones andcell phones.[24]
PCM is the method of encoding typically used for uncompressed digital audio.[note 3]
In the diagram, asine wave (red curve) is sampled and quantized for PCM. The sine wave is sampled at regular intervals, shown as vertical lines. For each sample, one of the available values (on the y-axis) is chosen. The PCM process is commonly implemented on a singleintegrated circuit called ananalog-to-digital converter (ADC). This produces a fully discrete representation of the input signal (blue points) that can be easily encoded as digital data for storage or manipulation. Several PCM streams could also be multiplexed into a larger aggregatedata stream, generally for transmission of multiple streams over a single physical link. One technique is calledtime-division multiplexing (TDM) and is widely used, notably in the modern public telephone system.
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 aredigital-to-analog converters (DACs). They produce avoltage orcurrent (depending on type) that represents the value presented on their digital inputs. This output would then generally be filtered and amplified for use.
To recover the original signal from the sampled data, ademodulator can apply the procedure of modulation in reverse. After each sampling period, the demodulator reads the next value and transitions the output signal to the new value. As a result of these transitions, the signal retains a significant amount of high-frequency energy due to imaging effects. To remove these undesirable frequencies, the demodulator passes the signal through areconstruction filter that suppresses energy outside the expected frequency range (greater than theNyquist frequency).[note 4]
Common sample depths for LPCM are 8, 16, 20 or 24 bits persample.[1][2][3][32]
LPCM encodes a single sound channel. Support for multichannel audio depends on file format and relies on synchronization of multiple LPCM streams.[5][33] While two channels (stereo) is the most common format, systems can support up to 8 audio channels (7.1 surround)[2][3] or more.
Common sampling frequencies are 48kHz as used withDVD format videos, or 44.1 kHz as used in CDs. Sampling frequencies of 96 kHz or 192 kHz can be used on some equipment, but thebenefits have been debated.[34]
TheNyquist–Shannon sampling theorem shows PCM devices can operate without introducing distortions within their designed frequency bands if they provide a sampling frequency at least twice that of the highest frequency contained in the input signal. For example, intelephony, the usablevoice frequency band ranges from approximately 300 to 3400 Hz.[35] For effective reconstruction of the voice signal, telephony applications therefore typically use an 8000 Hz sampling frequency which is more than twice the highest usable voice frequency.
Regardless, there are potential sources of impairment implicit in any PCM system:
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 analog-to-digital process; newer implementations do so in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-basedaudio compression techniques, such asmodified discrete cosine transform (MDCT) coding.
In telephony, a standard audio signal for a single phone call is encoded as 8,000samples per second, of 8 bits each, giving a 64 kbit/s digital signal known asDS0. The defaultsignal compression encoding on a DS0 is eitherμ-law (mu-law) PCM (North America and Japan) orA-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 standardG.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 μ-law or A-law 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 theG.726 standard.
Audio coding formats andaudio codecs have been developed to achieve further compression. Some of these techniques have been standardized and patented. Advanced compression techniques, such asmodified discrete cosine transform (MDCT) andlinear predictive coding (LPC), are now widely used inmobile phones,voice over IP (VoIP) andstreaming media.
PCM can be eitherreturn-to-zero (RZ) ornon-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 zeroes. For binary PCM systems, the density of 1-symbols is calledones-density.[36]
Ones-density is often controlled using precoding techniques such asrun-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 channel. In other cases, extraframing bits are added into the stream, which guarantees at least occasional symbol transitions.
Another technique used to control ones-density is the use of ascrambler on the data, which will tend to turn the data stream into a stream that lookspseudo-random, but where the data can be recovered exactly by a complementary descrambler. In this case, long runs of zeroes or ones are still possible on the output but are considered unlikely enough to allow reliable synchronization.
In other cases, the long term DC value of the modulated signal is important, as building up aDC bias will tend to move communications circuits out of their operating range. In this case, special measures are taken to keep a count of the cumulative DC bias and to modify the codes if necessary to make the DC bias always tend back to zero.
Many of these codes arebipolar codes, where the pulses can be positive, negative or absent. In the typicalalternate mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to generate special symbols used for framing or other special purposes.
The wordpulse in the termpulse-code modulation refers to the pulses to be found in the transmission line. This perhaps is a natural consequence of this technique having evolved alongside two analog methods,pulse-width modulation andpulse-position modulation, in which the information to be encoded is represented by discrete signal pulses of varying width or position, respectively.[citation needed] In this respect, PCM bears little resemblance to these other forms of signal encoding, except that all can be used in time-division multiplexing, and the numbers of the PCM codes are represented as electrical pulses.
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Line coding (digital baseband transmission) | ||
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| Main articles |  | |
| Basicline codes | ||
| Extended line codes | ||
| Optical line codes | ||
