But what sort of parameters are adjusted?

This is HN. A link to an example would be appreciated.

Edit: To clarify, I work with audio codecs too, and can't really think of parameters (other than the compression level?) that would make much sense to adjust on the fly.

If "AI" is used for more than just a buzzword here, then I imagine the answer must be quite interesting.

They probably adjust the incoming / outgoing buffer sizes (and therefore the audio delay, since it's live) to account for packet loss.

They might also prioritize traffic depending on how full your buffers are.

I can only assume Youtube and Netflix do similar parameter tweaks to optimize their video delivery based on the connection (totally filling the buffer to a max size all the time would waste bandwidth, but if the client has lots of packet loss they need a larger safety net).

Right, looks like we're up to maybe 6 parameters. The claim was dozens which i take to mean at least 24, possibly 36 as the lower bound.