Add Opus audio codec support

[KallDrexx]

WebRTC doesn't support AAC codec, so in order to ingest or serve audio via WebRTC we need to support Opus codec.

This involves:

• [ ] - Add Opus to the AudioCodec enumeration
• [ ] - Add code to successfully pass Opus data into gstreamer, so the AAC encoder can convert Opus -> AAC
• [ ] - Add an Opus gstreamer audio encoder to convert AAC to Opus
• [ ] - Verify WebRTC Opus negotiation works as expected.