Speech coding: Difference between revisions

Revision as of 21:48, 9 September 2024

Revision as of 22:20, 9 September 2024

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== Sample companding viewed as a form of speech coding ==

The [[A-law algorithm|A-law]] and [[μ-law algorithm]]s used in [[G.711]] PCM [[digital telephony]] can be seen as an earlier precursor of speech encoding, requiring only 8 bits per sample but giving effectively 12 [[audio bit depth|bits of resolution]].<ref>{{cite book|first=N. S. |last=Jayant and |first=P.|last= Noll,|title= Digital coding of waveforms.|___location= Englewood Cliffs:|publisher= Prentice-Hall, |year=1984.}}</ref> Logarithmic companding are consistent with human hearing perception in that a low-amplitude noise is heard along a low-amplitude speech signal but is masked by a high-amplitude one. Although this would generate unacceptable distortion in a music signal, the peaky nature of speech waveforms, combined with the simple frequency structure of speech as a [[periodic function|periodic waveform]] having a single [[fundamental frequency]] with occasional added noise bursts, make these very simple instantaneous compression algorithms acceptable for speech.{{citation needed|date=July 2023}}{{dubious|discuss=Logarithmic companding for music|date=July 2023}}

A wide variety of other algorithms were tried at the time, mostly [[delta modulation]] variants, but after careful consideration, the A-law/μ-law algorithms were chosen by the designers of the early digital telephony systems. At the time of their design, their 33% bandwidth reduction for a very low complexity made an excellent engineering compromise. Their audio performance remains acceptable, and there was no need to replace them in the stationary phone network.{{citation needed|date=July 2023}}