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Voice Codec for Mobile Network
post Jan 11 2013, 04:33
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I am wondering how does the current voice codec Wideband AMR compared to Opus? And why aren't we using Opus on 4G LTE All packet network instead when the Internet has decided on Opus.

Would switching to Opus be hard ( aren't these spec defined, or could software be updated to use it? ) ( If LTE is all packet based i see no reason why not possible )

And on such as low bitrate, would voice cancellation, multiple microphone helps more then a codec change in voice quality?

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post Jan 11 2013, 06:59
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AMR-NB and AMR-WB seem to be competitive with Opus or better for mono speech from about 6 to 16 kbps, and Opus seems to have taken the lead by around 24 kbps. The speech oriented tests reported to the IETF and referenced in the Wikipedia page opus (audio format) are worth a look. Packet Loss Concealment etc. seems to be better in Opus than AMR-WB, as is latency.

Opus is a new standard. AMR-WB has been around a bit longer, as have LTE and true 4G.

The standards adopted for connection to traditional phone networks and phone numbers tend to get set in stone earlier and bodies other than IETF and W3C (which are involved with standards using Opus) are more important for official codecs required for interoperability (e.g. ITU-T), though since 2G mobile, extensions such as the GSM EFR codec have been usable when both handsets (or handset plus base station connecting to POTS) supported them and network capacity was sufficient. I don't know if Opus can be introduced in the same way or whether limitations such as strict CBR requirements might prevent that. As LTE is a step towards 4G, rather than true 4G, there might be scope for introduction of Opus at another stage.
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