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In 1950, [[Bell Labs]] filed the patent on [[differential pulse-code modulation]] (DPCM).<ref name="DPCM">{{US patent reference|inventor=C. Chapin Cutler|title=Differential Quantization of Communication Signals|number=2605361|A-Datum=1950-06-29|issue-date=1952-07-29}}</ref> [[Adaptive DPCM]] (ADPCM) was introduced by P. Cummiskey, [[Nikil Jayant|Nikil S. Jayant]] and [[James L. Flanagan]] at [[Bell Labs]] in 1973.<ref>P. Cummiskey, Nikil 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</ref><ref>{{cite journal |last1=Cummiskey |first1=P. |last2=Jayant |first2=Nikil S. |last3=Flanagan |first3=J. L. |title=Adaptive quantization in differential PCM coding of speech |journal=The Bell System Technical Journal |date=1973 |volume=52 |issue=7 |pages=1105–1118 |doi=10.1002/j.1538-7305.1973.tb02007.x |issn=0005-8580}}</ref>
[[Perceptual coding]] was first used for [[speech coding]] compression, with [[linear predictive coding]] (LPC).<ref name="Schroeder2014">{{cite book |last1=Schroeder |first=Manfred R. |title=Acoustics, Information, and Communication: Memorial Volume in Honor of Manfred R. Schroeder |date=2014 |publisher=Springer |isbn=9783319056609 |chapter=Bell Laboratories |page=388 |chapter-url=https://books.google.com/books?id=d9IkBAAAQBAJ&pg=PA388}}</ref> Initial concepts for LPC date back to the work of [[Fumitada Itakura]] ([[Nagoya University]]) and Shuzo Saito ([[Nippon Telegraph and Telephone]]) in 1966.<ref>{{cite journal |last1=Gray |first1=Robert M. |title=A History of Realtime Digital Speech on Packet Networks: Part II of Linear Predictive Coding and the Internet Protocol |journal=Found. Trends Signal Process. |date=2010 |volume=3 |issue=4 |pages=203–303 |doi=10.1561/2000000036 |url=https://ee.stanford.edu/~gray/lpcip.pdf |issn=1932-8346}}</ref> During the 1970s, [[Bishnu S. Atal]] and [[Manfred R. Schroeder]] at [[Bell Labs]] developed a form of LPC called [[adaptive predictive coding]] (APC), a perceptual coding algorithm that exploited the masking properties of the human ear, followed in the early 1980s with the [[code-excited linear prediction]] (CELP) algorithm which achieved a significant
[[Discrete cosine transform]] (DCT), developed by [[N. Ahmed|Nasir Ahmed]], T. Natarajan and [[K. R. Rao]] in 1974,<ref name="DCT">{{cite journal |author1=Nasir Ahmed |author1-link=N. Ahmed |author2=T. Natarajan |author3=Kamisetty Ramamohan Rao |journal=IEEE Transactions on Computers|title=Discrete Cosine Transform|volume=C-23|issue=1|pages=90–93|date=January 1974 |doi=10.1109/T-C.1974.223784 |url=https://www.ic.tu-berlin.de/fileadmin/fg121/Source-Coding_WS12/selected-readings/Ahmed_et_al.__1974.pdf}}</ref> provided the basis for the [[modified discrete cosine transform]] (MDCT) used by modern audio compression formats such as MP3<ref name="Guckert">{{cite web |last1=Guckert |first1=John |title=The Use of FFT and MDCT in MP3 Audio Compression |url=http://www.math.utah.edu/~gustafso/s2012/2270/web-projects/Guckert-audio-compression-svd-mdct-MP3.pdf |website=[[University of Utah]] |date=Spring 2012 |accessdate=14 July 2019}}</ref> and AAC. MDCT was proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987,<ref>J. P. Princen, A. W. Johnson und A. B. Bradley: ''Subband/transform coding using filter bank designs based on time ___domain aliasing cancellation'', IEEE Proc. Intl. Conference on Acoustics, Speech, and Signal Processing (ICASSP), 2161–2164, 1987.</ref> following earlier work by Princen and Bradley in 1986.<ref>John P. Princen, Alan B. Bradley: ''Analysis/synthesis filter bank design based on time ___domain aliasing cancellation'', IEEE Trans. Acoust. Speech Signal Processing, ''ASSP-34'' (5), 1153–1161, 1986.</ref> The MDCT is used by modern audio compression formats such as [[Dolby Digital]],<ref name="Luo">{{cite book |last1=Luo |first1=Fa-Long |title=Mobile Multimedia Broadcasting Standards: Technology and Practice |date=2008 |publisher=[[Springer Science & Business Media]] |isbn=9780387782638 |page=590 |url=https://books.google.com/?id=l6PovWat8SMC&pg=PA590}}</ref><ref>{{cite journal |last1=Britanak |first1=V. |title=On Properties, Relations, and Simplified Implementation of Filter Banks in the Dolby Digital (Plus) AC-3 Audio Coding Standards |journal=IEEE Transactions on Audio, Speech, and Language Processing |date=2011 |volume=19 |issue=5 |pages=1231–1241 |doi=10.1109/TASL.2010.2087755}}</ref> [[MP3]],<ref name="Guckert">{{cite web |last1=Guckert |first1=John |title=The Use of FFT and MDCT in MP3 Audio Compression |url=http://www.math.utah.edu/~gustafso/s2012/2270/web-projects/Guckert-audio-compression-svd-mdct-MP3.pdf |website=[[University of Utah]] |date=Spring 2012 |accessdate=14 July 2019}}</ref> and [[Advanced Audio Coding]] (AAC).<ref name=brandenburg>{{cite web|url=http://graphics.ethz.ch/teaching/mmcom12/slides/mp3_and_aac_brandenburg.pdf|title=MP3 and AAC Explained|last=Brandenburg|first=Karlheinz|year=1999|url-status=live|archiveurl=https://web.archive.org/web/20170213191747/https://graphics.ethz.ch/teaching/mmcom12/slides/mp3_and_aac_brandenburg.pdf|archivedate=2017-02-13}}</ref>
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