feat: register Opus codec in media-sdk opus package

[thetillhoff]

Summary

• Mirrors the g711/g722/dtmf pattern: init() in the codec's own package calls RegisterCodec so importers get the codec registered by side-effect
• CGo-gated (//go:build cgo) — requires libopus
• Disabled: true — opt-in; callers enable via msdk.CodecSetEnabled("opus/48000/2", true)
• Exports SDPName = "opus/48000/2" constant for callers

Related

• Add optional Opus codec support for SIP sip#712 — prior Opus PR; feedback requested codec registration move to media-sdk
• feat: add opt-in Opus codec support for SIP media negotiation sip#738 — companion sip PR implementing that feedback

[inechay]

👍 for this approach. I independently ran into #281, built the same media-sdk registration plus a companion sip change, and only then found this was already in flight — I've closed my duplicate (#72) and won't open a competing sip PR against #738.

One data point in case it's useful: I built this registration into a custom livekit-sip image and ran it on a live inbound call from a softphone (Zoiper, Opus offered above G722). The SIP leg negotiated Opus cleanly:

• SDP answer a=rtpmap:106 opus/48000/2, log using codecs … audio-codec=opus/48000/2
• clean 48 kHz media over a ~5 min call: expected_pcm_hz=48000, packets_failed=0, write_errors=0, no decode errors
• DTMF / telephone-event intact

So the opus package registration + the generic RTP path (rtp.EncodePCM / rtp.DecodePCM) work end to end on the SIP leg with no per-codec wiring.

One optional enhancement for the SIP encode path, as a follow-up if you think it fits: a telephony-tuned encoder — inband FEC + a packet-loss estimate so lossy mobile paths (wifi↔LTE handoff, phone in pocket) degrade gracefully, plus higher bitrate/complexity for a richer narrator/music-bed sound. I kept it in a separate sipEncode from the shared Encode so the WebRTC leg (its own bandwidth estimation) stays untouched. Glad to send it on top of this once it merges.