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

[Ethan Funk]

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...
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