Adaptive Multi-Rate audio codec: Difference between revisions

Revision as of 19:17, 23 December 2020

Revision as of 16:31, 10 July 2021

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== Usage ==

The frames contain 160 samples and are 20 milliseconds long.<ref name="3gpp-26090" /> AMR uses various techniques, such as [[algebraic code -excited linear prediction|ACELP]], [[discontinuous transmission|DTX]], [[voice activity detection|VAD]] and [[comfort noise|CNG]]. The usage of AMR requires optimized link adaptation that selects the best codec mode to meet the local radio channel and capacity requirements. If the radio conditions are bad, [[source coding]] is reduced and [[channel coding]] is increased. This improves the quality and robustness of the network connection while sacrificing some voice clarity. In the particular case of AMR this improvement is somewhere around S/N = 4–6&nbsp;dB for usable communication. The new intelligent system allows the network operator to prioritize capacity or quality per base station.

There are a total of 14 modes of the AMR codec, eight are available in a [[Full Rate|full rate channel (FR)]] and six on a [[Half Rate|half rate channel (HR)]].

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* AMR is a hybrid speech coder, and as such transmits both speech parameters and a waveform signal

** [[Linear predictive coding]] (LPC) is used to synthesize the speech from a residual waveform. The LPC parameters are encoded as [[line spectral pairs]] (LSP).

** The residual waveform is coded using [[algebraic code -excited linear prediction]] (ACELP).

* The complexity of the algorithm is rated at 5, using a relative scale where [[G.711]] is 1 and [[G.729a]] is 15.

* [[PSQM]] testing under ideal conditions yields [[mean opinion score]]s of 4.14 for AMR (12.2&nbsp;kbit/s), compared to 4.45 for [[G.711]] (μ-law){{citation needed|date=October 2019}}