JACK audio transport streams are in Opus format by default. Those of us who have been playing in the audio production field have been spoiled for quite a while now.
I'm puzzled by this. I'll do my homework, but for the moment, are you saying JACK transcodes its input into Opus to move it around from app to app, then transcodes it again on the way out?
Yeah. Opus is the default transport codec, and, should you wish to transcode to mp3 or otherwise, is converted after the fact.
One of the nice side effects of this is that transport streams are really low-bandwidth. If I set up a JACK master server, and, say, six slave laptops recording input, I can stream/mix over 802.11g without latency, whereas other solutions will fully saturate a gigabit switch.
But surely when I use JACK to record the audio output of VLC into Audacity (easiest way to record a short clip of what you're watching, IMHO, because you can apply a quick clean+edit immediately), it sends the audio lossless? (that is, a lossless copy of the VLC decoding of the probably lossy audio stream).