Pulse-code modulation
Filename extension |
.L16, .WAV, .AIFF, .AU, .PCM[1] |
---|---|
Internet media type |
audio/L16, audio/L8, M2TS, VOB, and many others |
Open format? | Yes |
Free format? | Yes[5] |
Passband modulation |
---|
Analog modulation |
Digital modulation |
Hierarchical modulation |
Spread spectrum |
See also |
Pulse-code modulation (PCM) is a method used to
Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform.
A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the
History
Early electrical communications started to sample signals in order to multiplex samples from multiple telegraphy sources and to convey them over a single telegraph cable. The American inventor Moses G. Farmer conceived telegraph time-division multiplexing (TDM) as early as 1853. Electrical engineer W. M. Miner, in 1903, used an electro-mechanical commutator for time-division multiplexing multiple telegraph signals; he also applied this technology to telephony. He obtained intelligible speech from channels sampled at a rate above 3500–4300 Hz; lower rates proved unsatisfactory.
In 1920, the
British engineer
The first transmission of speech by digital techniques, the SIGSALY encryption equipment, conveyed high-level Allied communications during World War II. In 1943 the Bell Labs researchers who designed the SIGSALY system became aware of the use of PCM binary coding as already proposed by Reeves. In 1949, for the Canadian Navy's DATAR system, Ferranti Canada built a working PCM radio system that was able to transmit digitized radar data over long distances.[10]
PCM in the late 1940s and early 1950s used a
In the United States, the National Inventors Hall of Fame has honored Bernard M. Oliver[13] and Claude Shannon[14] as the inventors of PCM,[15] 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 by John 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.[16]
The T-carrier system, introduced in 1961, uses two twisted-pair transmission lines to carry 24 PCM telephone calls sampled at 8 kHz and 8-bit resolution. This development improved capacity and call quality compared to the previous frequency-division multiplexing schemes.
In 1973, adaptive differential pulse-code modulation (ADPCM) was developed, by P. Cummiskey, Nikil Jayant and James L. Flanagan.[17]
Digital audio recordings
In 1967, the first PCM recorder was developed by
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 "with emphasis, making it equivalent to 15.5 bits."[18]
In 1979, the first digital pop album,
The compact disc (CD) brought PCM to consumer audio applications with its introduction in 1982. The CD uses a 44,100 Hz sampling frequency and 16-bit resolution and stores up to 80 minutes of stereo audio per disc.
Digital telephony
The rapid development and wide adoption of PCM
Implementations
PCM is the method of encoding typically used for uncompressed digital audio.[note 3]
- The 4ESS switch introduced time-division switching into the US telephone system in 1976, based on medium scale integrated circuit technology.[22]
- LPCM is used for the lossless encoding of audio data in the compact disc Red Book standard(informally also known as Audio CD), introduced in 1982.
- AES3 (specified in 1985, upon which S/PDIF is based) is a particular format using LPCM.
- LaserDiscs with digital sound have an LPCM track on the digital channel.
- On PCs, PCM and LPCM often refer to the format used in container formats.
- LPCM has been defined as a part of the ).
- LPCM is used by HDMI (defined in 2002), a single-cable digital audio/video connector interface for transmitting uncompressed digital data.
- RF64 container format (defined in 2007) uses LPCM and also allows non-PCM bitstream storage: various compression formats contained in the RF64 file as data bursts (Dolby E, Dolby AC3, DTS, MPEG-1/MPEG-2 Audio) can be "disguised" as PCM linear.[28]
Modulation
In the diagram, a sine 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 single integrated circuit called an analog-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 aggregate data stream, generally for transmission of multiple streams over a single physical link. One technique is called time-division multiplexing (TDM) and is widely used, notably in the modern public telephone system.
Demodulation
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 digital-to-analog converters (DACs). They produce a voltage or current (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, a demodulator 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 a reconstruction filter that suppresses energy outside the expected frequency range (greater than the Nyquist frequency ).[note 4]
Standard sampling precision and rates
Common sample depths for LPCM are 8, 16, 20 or 24 bits per
LPCM encodes a single sound channel. Support for multichannel audio depends on file format and relies on synchronization of multiple LPCM streams.[5][30] 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 48 kHz as used with DVD 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 the benefits have been debated.[31]
Limitations
The Nyquist–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, in telephony, the usable voice frequency band ranges from approximately 300 Hz to 3400 Hz.[32] 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:
- Choosing a discrete value that is near but not exactly at the analog signal level for each sample leads to quantization error.[note 5]
- Between samples no measurement of the signal is made; the sampling theorem guarantees non-ambiguous representation and recovery of the signal only if it has no energy at frequency fs/2 or higher (one half the sampling frequency, known as the Nyquist frequency); higher frequencies will not be correctly represented or recovered and add aliasing distortion to the signal below the Nyquist frequency.
- As samples are dependent on time, an accurate clock is required for accurate reproduction. If either the encoding or decoding clock is not stable, these imperfections will directly affect the output quality of the device.[note 6]
Processing and coding
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-based
- Linear PCM (LPCM) is PCM with linear quantization.[5]
- Differential PCM(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.
- Adaptive differential pulse-code modulation (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 ratio.
- Delta modulation is a form of DPCM that uses one bit per sample to indicate whether the signal is increasing or decreasing compared to the previous sample.
In telephony, a standard audio signal for a single phone call is encoded as 8,000
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 the G.726 standard.
Encoding for serial transmission
PCM 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 zeroes. For binary PCM systems, the density of 1-symbols is called ones-density.[33]
Ones-density is often controlled using precoding techniques such as
Another technique used to control ones-density is the use of a
In other cases, the long term DC value of the modulated signal is important, as building up a DC 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 are
Nomenclature
The word pulse in the term pulse-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 and pulse-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.
See also
- Beta encoder
- Equivalent pulse code modulation noise
- Signal-to-quantization-noise ratio (SQNR), one method of measuring quantization error
Explanatory notes
- ^ Among the first recordings was Uzu: The World Of Stomu Yamash'ta 2 by Stomu Yamashta.
- ^ The first recording with this new system was recorded in Tokyo during April 24–26, 1972.
- ^ Other methods exist such as pulse-density modulation used also on Super Audio CD.
- ^ 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 anti-aliasing filter is much simpler. In some systems, no explicit filtering is done at all; as it is 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.
- uniformly distributedover this interval, with zero mean and variance of q2/12.
- ^ A slight difference between the encoding and decoding clock frequencies is not generally a major concern; a small constant error is not noticeable. Clock error does become a major issue if the clock contains significant jitter, however.
References
- ^ doi:10.17487/RFC2586. Retrieved March 16, 2010.)
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(help - ^ doi:10.17487/RFC4856. Retrieved March 16, 2010.)
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(help - ^ doi:10.17487/RFC3190. Retrieved March 16, 2010.)
{{cite journal}}
: Cite journal requires|journal=
(help - ^ "Audio Media Types". Internet Assigned Numbers Authority. Retrieved March 16, 2010.
- ^ a b c d "Linear Pulse Code Modulated Audio (LPCM)". Library of Congress. April 19, 2022. Retrieved September 5, 2022.
- ^ "The Bartlane Transmission System". DigicamHistory.com. Archived from the original on February 10, 2010. Retrieved January 7, 2010.
- ISBN 0-7506-7841-0.
- ^ IEEE
- ^ US 2272070
- ISBN 9780931761188.[page needed]
- ^ Sears, R. W. (January 1948). Electron Beam Deflection Tube for Pulse Code Modulation. Vol. 27. Bell Labs. pp. 44–57. Retrieved May 14, 2017.
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ignored (help) - ^ Goodall, W. M. (January 1951). Television by Pulse Code Modulation. Vol. 30. Bell Labs. pp. 33–49. Retrieved May 14, 2017.
{{cite book}}
:|work=
ignored (help) - ^ "Bernard Oliver". National Inventor's Hall of Fame. Archived from the original on December 5, 2010. Retrieved February 6, 2011.
- ^ "Claude Shannon". National Inventor's Hall of Fame. Archived from the original on December 6, 2010. Retrieved February 6, 2011.
- ^ "National Inventors Hall of Fame announces 2004 class of inventors". Science Blog. February 11, 2004. Retrieved February 6, 2011.
- ^
B. M. Oliver; J. R. Pierce & C. E. Shannon (November 1948). "The Philosophy of PCM". Proceedings of the IRE. 36 (11): 1324–1331. S2CID 51663786.
- ^ P. Cummiskey, N. S. Jayant, and J. L. Flanagan, "Adaptive quantization in differential PCM coding of speech," Bell Syst. Tech. J., vol. 52, pp. 1105—1118, Sept. 1973.
- ^ a b c Thomas Fine (2008). "The dawn of commercial digital recording" (PDF). ARSC Journal. 39 (1): 1–17.
- ^ Roger Nichols. "I Can't Keep Up With All The Formats II". Archived from the original on October 20, 2002.
The Ry Cooder Bop Till You Drop album was the first digitally recorded pop album
- ^ ISBN 9788793609860. Archived from the original(PDF) on September 30, 2021. Retrieved November 29, 2019.
- ^ ISBN 9781420041163.
- ^ Cambron, G. Keith (October 17, 2012). Global Networks: Engineering, Operations and Design. John Wiley & Sons. p. 345.
- ^ Blu-ray Disc Association (March 2005), White paper Blu-ray Disc Format – 2.B Audio Visual Application Format Specifications for BD-ROM (PDF), retrieved July 26, 2009
- ^ "DVD Technical Notes (DVD Video – "Book B") – Audio data specifications". July 21, 1996. Retrieved March 16, 2010.
- ^ Jim Taylor. "DVD Frequently Asked Questions (and Answers) – Audio details of DVD-Video". Retrieved March 20, 2010.
- ^ "How DV works". Archived from the original on December 6, 2007. Retrieved March 21, 2010.
- ^ "AVCHD Information Website – AVCHD format specification overview". Retrieved March 21, 2010.
- ^ EBU (July 2009), EBU Tech 3306 – MBWF / RF64: An Extended File Format for Audio (PDF), archived from the original (PDF) on November 22, 2009, retrieved January 19, 2010
- doi:10.17487/RFC3108. Retrieved March 16, 2010.)
{{cite journal}}
: Cite journal requires|journal=
(help - ^ "PCM, Pulse Code Modulated Audio". Library of Congress. April 6, 2022. Retrieved September 5, 2022.
- ^ Christopher, Montgometry. "24/192 Music Downloads, and why they do not make sense". Chris "Monty" Montgomery. Archived from the original on September 6, 2014. Retrieved March 16, 2013.
- ^ https://www.its.bldrdoc.gov/fs-1037/dir-039/_5829.htm[failed verification]
Further reading
- .
- Ken C. Pohlmann (1985). Principles of Digital Audio (2nd ed.). Carmel, Indiana: Sams/Prentice-Hall Computer Publishing. ISBN 978-0-672-22634-2.
- doi:10.3758/BF03204440.)
{{cite journal}}
: CS1 maint: multiple names: authors list (link - Bill Waggener (1995). Pulse Code Modulation Techniques (1st ed.). New York, NY: Van Nostrand Reinhold. ISBN 978-0-442-01436-0.
- Bill Waggener (1999). Pulse Code Modulation Systems Design (1st ed.). Boston, MA: Artech House. ISBN 978-0-89006-776-5.
External links
- PCM description on MultimediaWiki
- Ralph Miller and Bob Badgley invented multi-level PCM independently in their work at Bell Labs on SIGSALY: U.S. patent 3,912,868 filed in 1943: N-ary Pulse Code Modulation.
- Information about PCM: A description of PCM with links to information about subtypes of this format (for example linear pulse-code modulation), and references to their specifications.
- Summary of LPCM – Contains links to information about implementations and their specifications.
- How to control internal/external hardware using Microsoft's Media Control Interface – Contains information about, and specifications for the implementation of LPCM used in WAV files.
- RFC 4856 – Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences – audio/L8 and audio/L16 (March 2007)
- RFC 3190 – RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio (January 2002)
- RFC 3551 – RTP Profile for Audio and Video Conferences with Minimal Control – L8 and L16 (July 2003)