Speech Coding - Sample Companding Viewed As A Form of Speech Coding

Sample Companding Viewed As A Form of Speech Coding

From this viewpoint, the A-law and μ-law algorithms (G.711) used in traditional PCM digital telephony can be seen as a very early precursor of speech encoding, requiring only 8 bits per sample but giving effectively 12 bits of resolution. The logarithmic companding laws 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.

A wide variety of other algorithms were tried at the time, mostly variants on delta modulation, 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 them an excellent engineering compromise. Their audio performance remains acceptable, and there has been no need to replace them in the stationary phone network.

In 2008, G.711.1 codec, which has a scalable structure, was standardized by ITU-T. The input sampling rate is 16 kHz.