Editing Speech coding (section)

== Sample companding viewed as a form of speech coding ==
The [[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|first1=N. S. |last1=Jayant|first2=P.|last2= 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 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}}

In 2008, [[G.711.1]] codec, which has a scalable structure, was standardized by ITU-T. The input sampling rate is 16&nbsp;kHz.<ref name="g711-1-2012">{{citation |publisher=ITU-T |date=2012 |url=http://www.itu.int/rec/T-REC-G.711.1/en |title=G.711.1 : Wideband embedded extension for G.711 pulse code modulation |access-date=2022-12-24}}</ref>