[LAD] Jack audio questions to aid in my development of audiorack

I am in the process of porting a radio automation system that I originally started writing for OSX/CoreAudio some 10 years ago. The OS X version is currently in use at a radio station I am co-owner of, and 2 area LPFM radio station. But I have become progressively frustrated with Apple as each new OS release is made, and find that that's with the continued evolution of jack-audio and gstreamer, I can now ditch my dependency on Apple's audio API. This a huge project, with GUI elements, audio play-out/mixing elements, library management elements, etc, to be ported. I decided to start with the CLI audio/automation engine first, making use of jack to break the original CLI program up into pieces that interconnect with jack: the mixer/automation/control application, media players and media recorders/encoders. I haven't decided what to code the GUI elements with... but I'm not at that point yet. Without the GUI, the CLI pieces are a monster for a human to operate. But again, I needed to start someplace. My "alpha" release of the CLI pieces, so far tested on Ubuntu 19.10, can be found here: https://github.com/eafunk/audiorack NOTE: there is a database bug in the libdbi mysql driver that I have fixed locally. Without the libdbi fix, there are some multi-threading stability problems. I'm working to get my fix into the libdbi code base. With the demise of the jack-audio email list, I have no place to go with questions regarding jack. So is this list a reasonable place for me to ask questions related to jack? By way of full disclosure, I am developing on Ubuntu, with an eye on making the project build-able on OS X and FreeBSD as well. For example, I have some audio problem with low-latency jack audio on Ubuntu right now that I could use some insight on: 1. With the low-latency kernel, jack dropouts/underruns are a problem when jack is configured to bridge to pulseaudio. Without a pulseaudio bridge, the dropout are nonexistent. Is there a way to increase the "buffer/latency" on the bridge between pulseaudio and jack? Latency on pulseaudio is already poor, so audio coming from/to pulseaudio wouldn't be hurt by extra buffering to keep jack's low latency paths and realtime thread happy. 2. I have a similar issue with zita-a2j and zita-j2a with lost of dropouts on the ALAS side, even though the jack side shows no overruns. Can extra buffering be applied to the zita programs? Maybe that is what the -n option is for? 3. I'm using Ubuntu Studio Control to make jack integration with Ubuntu easy. Does anyone know where I can find the source code for Ubuntu Studio Control? 4. Can anyone recommend an audio effects host that doesn't also try to manage jack connections? I can't have a "studio manager" host my audio processing (compression, de-essing, etc) and decides to make connections between my audio engine application and the media handler's it spawns. I need my audio engine to make it's own connection decisions. Thanks, Ethan...