[17:54] <OvenWerks> tastenheld[m]: general testing of autojack: in studio controls, system tweaks tab, set Logging Level to Extra. Then hit either Restart Jack or Stop Jack (depending on where you want things to end up)
[17:55] <OvenWerks> tastenheld[m]: wait a minute of so noting that Jack status should make some changes. It should  say either stopped or running when it is done.
[17:57] <OvenWerks> tastenheld[m]: next, you should find the log file in ~/.log/autojack.log and maybe the file with a .1 on the end if it got too big at the wrong time.
[17:58] <OvenWerks> tastenheld[m]: normally the tail of that file will give an indication of why it quit.
[18:00] <OvenWerks> tastenheld[m]: another thing to try if the log file doesn't give you any ideas, is to run autojack in a terminal and look at the terminal output as there are some things that are "unloggable".
[18:01] <OvenWerks> If you wish to file an issue on the git page above. uploading these few files would be helpful.
[18:03] <OvenWerks> If you catch me when I am around here you can drop them in a paste site and I will look at them here.
[18:48] <pauluzs> Hi folks,
[18:48] <pauluzs> Currently i'm trying out ubuntu-studio 23.04 with pipewire.
[18:48] <pauluzs> After messing arround with different config- and command-line settings, I'm still unable to run 2 jack aplications at different sample rates. 
[18:48] <pauluzs> Goal is to run the sound-hardware and stereotool-jack @ 192Khz to output MPX to a fm transmitter and feed it from mixxx, which can only run up to 96Khz. 
[18:48] <pauluzs> question is this possible? and who can hint me to the right configuration of documention how to achieve this?
[18:55] <OvenWerks> pauluzs: that would not be possible because of how the jack graph works. In order to have jack graphs with two different SR, you would need two jack servers with two different server names.
[18:57] <pauluzs> @Ovenwerks thanks, in 22.04 this was working out of the box. (no pipewire) i'll have a look into running a second jack server
[18:58] <OvenWerks> in order to run two applications with different SR, if they are to connect at all (no matter what audio server you use) you need an SRC step somewhere to convert from one to the other
[19:11] <OvenWerks> pauluzs: I have only a slight idea for what you are trying to do. That you want to run a 192k and mention FM makes me think you are going to output RF? Is that correct? There is no good reason to use anything higher for audio meant for human hearing on it's own.
[19:11] <OvenWerks> anyway, I am going to spend some time with my wife. so maybe talk another time.
[19:43] <pauluzs> @OvenWerks i use mixxx as dj program with microphone input from the sound card. and stereotool to create a mpx multiplex which goes into a fm transmitter to modulate it. the MPX constist of a mono sigal L+R 0-15Khz, a pilot tone @19 Khz to identify a stereo transmission, The stereo signal L-R 0-15Khz centered arround 38 Khz and a RDS(data) subcarrier centered auround 57Khz . This signal gets to the transmitter either analog 
[19:43] <pauluzs> over the soundcard or digital to a gnu-radio script to  a transmit capable SDR and get converted upto RF there . To satisfy nyquist, there is the need to run at least double the sample rate of the highest freqency. which in my case is the RDS just below 60 Khz. For american broadcast there is also the provision for addition subcarriers the highest almost 96Khz. This is why the minimal sample rate schould be 192Khz.
[22:05] <OvenWerks> That makes sense, lots of audio devices might have odd problems if used as mono
[22:05] <pauluzs> i run it 192 128 with no noticble delay for me
[22:06] <OvenWerks> That would be about the same as 48@32/2
[22:06] <OvenWerks> so yeah pretty good for latency\
[22:06] <pauluzs> any faster i get xruns
[22:06] <OvenWerks>  :) no surprise
[22:08] <OvenWerks> The engineers who designed the hardware/software spec for internal audio consider "low latency" to be 30ms
[22:08] <OvenWerks> As a musician, even 20ms has me playing out of time.
[22:11] <pauluzs> i tend to start to stutter as soon as feedback delay hits 26 ms
[22:12] <OvenWerks> You may have to run real (tm?) jack
[22:13] <pauluzs> quitte happy with the setup now, only wordered if i could achieve the same in pipewire
[22:14] <OvenWerks> piepwire, despite being default audio these days, is still in very active development.
[22:15] <OvenWerks> It is not finished, though it does very well for desktop audio and even a lot of pro audio needs if they don't need low latency.
[22:15] <pauluzs> i'm aware of that, still trying out the new stuff whille hoping to also get my second input working on the same device
[22:16] <pauluzs> i'll give that a second try tomorrow, haven't succeeded yet and its getting late over here
[22:16] <OvenWerks> yeah, well alsa_in or zita-a2j may do that if the audio device (or subdevice) is visible
[22:18] <OvenWerks> At least in jack, not so sure about with pipewire
[22:18] <OvenWerks> I have not tried those utilities with pipewire.
[22:19] <OvenWerks> Anyway, good night.