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You should look into using a JACK audio server for low latency stuff. I would guess you run on PulseAudio currently.
install jack and qjackctl (control dialog for JACK).
sudo apt-get install jackd qjackctl
The system sound needs to be suspended for JACK to work properly (it needs full access to your audio hardware). Some JACK versions can suspend pulse automatically via dbus nowadays as far as i remember, but i don't know how things on ubuntu are currently. To be sure you can start qjackctrl wit pasuspender:
pasuspender qjackctl
This way pulse audio will release the audio hardware as long as JACK is running. You will now have to configure JACK in qjackctl. For basic functionality there should not be much to do the most important part is buffer size.
If the server starts successfully you can try to use it in Bitwig.
There might be more stuff you need to do e.g. JACK should have Realtime access rights and your user might be missing them. According to this page your user has to be in the "audio" group:
https://jackaudio.org/faq/linux_rt_config.html
Another good resource for further optimization is the arch wiki: https://wiki.archlinux.org/title/Professional_audio
If you are running Linux, you can do all kinds of filtering with JACK and all of the processors and filters available for it.
It would be best to have a computer dedicated to the audio processing so you don't accidentally send any unwanted audio out over the air. You don't need a powerful computer, even a Raspberry Pi would work if you have a USB audio interface for it.
A few things to add:
ondemandshould not cause DSP dropouts unless the machine is grossly underpowered for the workload, especially with JACK. If they're actually seeing this, it's probably something else, but setting theperformancegovernor won't hurt anything anyway.They should check to see that this problem doesn't happen with ALSA output first, before trying solutions related to Pulse/JACK/cpufreq, which are additional layers of complexity. Don't have to keep using ALSA after, it's just a good idea to check to see if it's driver-related or not.
This URL explains how to run PulseAudio and JACK together if needed. Lots of stuff still needs PulseAudio, unfortunately.
I am using pulseaudio alongside jack. I suppose one can work exclusively with jack but I prefer to use both. You could use this to help you decide.
I know calf has a variety of plugins that includes compressors, equalizers among others, you could refer to this page to check if calf has the plugin(s) you need.
I can't give you any more direction than this, but maybe consider using "Jack" to route the output of an app monitoring your Mic and applying the VSTs into Discord or Zoom or whatever.
I have never tried this, but seems like it'd be a solution. no idea about latency.
You can't just use several usb audio devices together, the audio in any DAW doesn't work like that. Your best bet would be to make an aggregate audio device with something like Jack or Soundflower, but that does require quite some setting up.
https://jackaudio.org/ might do what you need. It's been a while since I've run it, but I think it can provide virtual audio in/out that Zoom would recognize as a device.
Dummy speaker is probably provided by the hypervisor for compatibility for things that require PC BEEP.
Seems like this python module needs an additional installation.
whats this library? is it this? https://pypi.org/project/JACK-Client/
Reading that page it says:
> JACK library: The JACK library must be installed on your system (and CFFI must be able to find it). Again, you should use your package manager to install it. Make sure you install the JACK daemon (called jackd). This will also install the JACK library package. If you don’t have a package manager, you can try one of the binary installers from the JACK download page. If you prefer, you can of course also download the sources and compile everything locally.
Apparently the jack module you installed requires something called the JACK library to be installed as well.
A quick google lead me to this documentation which says you need this install. (not sure if this is the same thing you are working with or not).
They do optimize it a little bit. The goal for TV broadcast will be for the final audio master to be on the periphery of what is acceptable for transmission through mediocre, range-reduced speakers. However, these masters are admittedly calibrated by people who are used to loud noises and often do not have audio processing issues.
What I have done a couple times (usually for streams where the audio is crap) is download JACK and run the stream through JACK, then run JACK into my music software (Reaper) and compress it (amongst other things) in real time. Reaper sends the result out to my speakers.
Alternatively, you could get a hardware compressor and put it between the source and speakers. Here's a list of used compressors from Guitar Center from lowest to highest. The first result is a quad-channel unit, which is more than overkill for this purpose. Would I mix an album with it? No, but for TV or live events, it's more than adequate. It has a peak limiter, which (for the most part) won't allow any loud parts above a certain pre-defined threshold. My only warning is that, since it is pro-audio gear, if you choose to go this route, make sure you match the jacks and nearly-match the impedances of your connections.
It definitely would be nice if they had an extra compressed version of movies available. Or better yet, if more devices had an opt-in basic compressor/limiter built into them; with software, it's trivial to implement.
thats a common view that people have, but you don't actually need a RT kernel to setup a system for RT audio. I'm running RT Jack on a non-RT kernel myself on two machines.
Ah, that sounds like it might do the trick. I found this page which lists netjack1/2 and jack.trip.
Now, to figure out the best way to take inputs and put them over ethernet...
I was hoping you wouldn't ask that because I never did manage the right amount of concentration and spare time to do it myself.
This is the page I often look at for a while before I give up:
> 50 Scarlett 2i2s, linked together as an aggregate USB device in software
Could probably get it working with alsa_in and alsa_out on a Linux system, you'll just need 49 instances of that tool running :-p
Sorry I can't help more because I've never used it, but I can't see why it wouldn't do the job.
You might want a startup script set up so it starts the shennanigans when booting and a watchdog to reboot the pi if it crashes.
Solved, I followed [this](https://jackaudio.org/faq/linux_rt_config.html) procedure and then rebooted. Now everything works as expected. I don not if the issue has been solved by the linked procedure or the reboot.
Thanks for your answer.
fuser -v /dev/snd/* yields
/dev/snd/controlC0: <user> 12864 F..... pulseaudio /dev/snd/seq: <user> 289254 F..... ArdourGUI
I have pasuspender -- jackd in the “Server path” field in Qjackctl (as indicated in https://jackaudio.org/faq/pulseaudio_and_jack.html). It doesn't appear to stop pulseaudio. So what is the best way to stop pulseaudio? I did not change anything else in the config. I do have a file ~/.jackdrc, apparently autogenerated. It has
/app/bin/jackd -t 200 -p 2048 -R -T -d alsa -n 2 -r 44100 -p 1024 -d hw:USB,0 -X raw
Depending on what you want to do you may not need JACK. I find JACK a pain to use. Pipewire will hopefully replace it soon.
JACK basically a mixer that lets you route input and output sources for audio. It has a very bad relationship with PulseAudio and wants to murder it, and vice versa. They simply don't get along.
You can find out more about JACK at https://jackaudio.org/
If I understand correctly you need to use JACK. It’s a very powerful audio server that let you do all kind of complex stuff.
I actually started switching to linux my self. I was recommended to use gui to configure Jack : Cadence (very nice), qjackctl. You will end up graph view of all your input/output device, daws, standalone plugins,…
I use Ardour as DAW and plugin host (vst,lv2).
Look it up and feel free to ask me questions if that’s what is you looking for.
En un principio usted solo puede conectar Jack a una sola tarjeta de sonido:
https://jackaudio.org/faq/pulseaudio_and_jack.html
En Qjackctl se hace atrevés de SETUP->Configuraciones-> Avanzado.
You should check the JACK log (in QJackCtl or Cadence) to find the cause of xruns: whether it is the hardware, the JACK daemon, or a JACK application. If it is not the hardware than maybe the daemon or applications were unable to acquire real time privileges. This might happen if your user is not in the audio group, or if the audio group has not raised rtprio PAM limit: see https://jackaudio.org/faq/linux_rt_config.html
Voicemeeter Banana And Jack will give you considerably more control over your audio, but Jack can be a real bitch to set up properly. It has issues with my usb headphones for some reason.
I've had success using JACK Audio and Remote Desktop. I have a NUC running linux in my kitchen and sometimes I need to remote into my studio workstation for some Ableton work while I'm eating breakfast.
Basic steps (assuming your desktop is Windows):
- Install JACK on your desktop and set your audio interface in the JACK settings. Then start the JACK server thing. *Do this all before opening Ableton
- Once the JACK server is running, open Ableton and set the audio interface to JACK
- On your laptop you should be able to hear your Ableton audio
I've done these steps both locally and in a remote session and it's worked. You can see the jackaudio.org site for more info.
Over a wired connection the latency isn't too bad. It's been decent enough to do quick edits in large projects (60+ tracks). YMMV with wifi but in general but ethernet will always have lower latency.
It's a little quirky in other ways, for example mouse wheel scrolling seems to be magnified so that can be annoying. It's possible this is just a linux issue... Anyway the audio side works. Hope this helps.
Unfortunately there isnt one yet. Most projects on Github to make a Virtual Audio Driver are very old and not-compiled. There is a current project to make a virtual audio driver solution via python afaik, but that one currently only works on Linux, so Windows Python wont run it. But in general, virtual audio drivers are very simple made and its normal that you cant really control its volume, since its virtual and it has no real sound output for you to hear.
But if you actually enhance your experience with Virtual Audio, you could add Jackaudio. If you invest osme time into it, you will learn how it works real quick. With that, you can split a sound to a virtual audio driver and to an actual audio device and set up both volumes to the same level to get an impression of how loud/silent it actually is.
If you actually need more virtual audio drivers, there is Virtual Audio Cable. It allows you to create alot more virtual audio drivers.
But that aide, I couldnt find a good, maintained and updated opensource alternative either, yet.
>solution found!
>
>if you start jack without realtime with pulseaudio bridge (in manjaro it's "pulseaudio-jack") with cadence, navigate to the "jack bridges" "pulseaudio" tab, and then stop and start pulseaudio, it works!
>
>I also used this guide: https://jackaudio.org/faq/linux\_rt\_config.html
solution found!
if you start jack without realtime with pulseaudio bridge (in manjaro it's "pulseaudio-jack") with cadence, navigate to the "jack bridges" "pulseaudio" tab, and then stop and start pulseaudio, it works!
I also used this guide: https://jackaudio.org/faq/linux_rt_config.html
The obvious start is Jack so you have time control, sync, and audio patch between applications. I've never gotten Jack to work successfully on my own system, but since I don't really need it, I've just assumed that it requires some kind of animal sacrifice and left it alone.
From there, https://jackaudio.org/applications/ is a pretty good resource.
If I understand the issue, I think you'll have to pipe your PC audio out into the scarlett line in. You can try using asio4all and JACK to do this virtually, but every time I've tried that I ended up using a splitter and cable.
Alright, I'll give you a brief overview to start with. Here goes.
Linux is an operating system, much like macOS or Windows. (An operating system is the underlying piece of software that controls almost everything on a computer, from high-level stuff like running programs to low-level stuff like handling the literal ones and zeros on your hard drive or SSD.) Except that isn't really true, because Linux by itself is just a kernel, which is the very core of an operating system that interacts directly with your hardware/firmware (there's actually a whole copypasta about this). One of the biggest boons of Linux is this modularity. When you, for example, change out your taskbar for another taskbar on Windows using Classic Shell, you're not really completely swapping it out. The old taskbar is still there, and now Classic Shell is sitting on top of it as "bloat". On Linux, if you want to completely change out your taskbar, you can do that. You can actually completely change out your entire desktop environment (which does everything from window management to light/dark theming) fairly easily.
Another thing that is useful on Linux is that there are many tools out there that are actually only available on Linux. An example might be JACK, which is a framework you can use to do something similar to that of a full-on patchboard COMPLETELY IN SOFTWARE. You can even do things like applying effects to your voice or music and playing it back to yourself in real time. Another example could be QEMU/KVM, which is a virtualization technology that is unrivaled by ones available on other operating systems. (there's a reason most VM software for servers is Linux-based).
These are the steps I took for audio on my manjaro kde.
Install cadence and jack2. Follow these steps: https://jackaudio.org/faq/linux_rt_config.html Using cadence configure to jack (mostly just choosing the right device and sample rate and I use pulseaudio bridges as well).
And you could do this, but I've had no issues with my cpu governors, powersave. Check your available cpu govenors with: "cpupower frequency-info" and then you should probably "sudo cpupower set-frequency -g performance". But maybe substitute performance with another governors of your likings.
My interface is a Behringer UMC22 (which works out of the box using linux) but my laptop sound card works as well, I've had some x-run issues with HDMI though. Furthermore I just use the main-line kernel. I've heard using the RT-kernel makes little to no difference.
Always happy to help!
LMMS is Linux multi-media studio , no? If you're on linux then try SooperLooper instead. If works with the Jack audio system and it should be easier to set up. A really easy way to set it up is through Ubuntu Studio OS or their installer for linux OS's (I've used their installer to add jack to Kubuntu, for example.)
I don't think you need to buy any special hardware, you just need to dive deeper into using a virtual patch-cage for your applications.
A patch-cage will operate as the hub where you view/edit your virtual connections. Think of it like having a bunch of guitar pedals plugged into each other.
You can easily connect your mic into your looper, send you keyboard into a synth program then send that into the looper. The output can easily be sent both into headphones and into a recording program.
Jack is available on windows but I have no idea if there are better programs on windows for this.
Haven't use it but I suspect it works like other audio server software by allowing programs to send and receive audio data to/from each other.
The issue you are facing currently is that the mic is an input and the zoom audio is an output and you can't record both simultaneously with your current set up. By rerouting your mic audio input-->output you can make the mic also an output allowing you to capture it along with the zoom audio.
*edit. I've used Jack audio in the past to do this. I just realised it is available for Mac.
Not to my knowledge - you'd have to leverage a virtual audio driver and incur a little latency. I personally use Virtual Audio Cable. I hear Jack can work as well.
You probably set the mic as jack sound card. Now jack wants to send audio to the mic. That ofc doesn't work.
For ubuntu there's the excellent Studio Controls which basically does the following. 1) set your ”normal” soundcard as main device. 2) create extra jack inputs for your usb mic: you can either use alsa_in or the zita bridge. alsa_in comes preinstalled with jack.
As a bonus studio control also bridges pulseaudio to jack so you got your regular desktop audio while working with jack studio.
It can be done, but it will be a hassle amd might not work as wanted.. for example this might solve your problem: https://jackaudio.org/
But I can't recommend enough buying an interface you can get decent used stuff starting from 50-70€. It changed my life..
I found links in your comment that were not hyperlinked:
I did the honors for you.
^delete ^| ^information ^| ^<3
not an expert here, but one of the top suggestions is usually: do you have a real-time type kernel installed and have you given jack real-time priority?
You will find old documentation on an real-time kernel which I think is no longer being developed while there is a low latency one which is and is in many distribution repositories. The latter (low-latency) is what you want I think these days. I did not read all the history but I think it was first second to best and now much of the original real-time kernel code has been moved into it.
Again, not systems engineer, so I am about to maybe be waving my hands around and not say anything actually correct - but anyway - the perception you are having may be because pulseaudio does not have the same demands for real-time. I believe the pulseaudio server can add lantency if the cpu demands are high because the usages of pulseaudio are typically higher (you wouldn't notice if latency become 1-2 ms longer from time to time) and pulseaudio does not have the same requirements for clock-perfect synchronization of audio. As I understand it, jack on the other hand is far far less forgiving, when the system says "I need the next sample of audio" - there MUST be the next sample of audio otherwise there will be problems. That makes it harder to have a low-latency system but also gives it advantages / reliability for audio engineers etc.
Again, the above comes with a disclaimer of: that might be bull-shit but I think it might explain the apparent paradox you are experiencing.
I'm mixing! At the bottom, to the right I have Mixxx, a free and cross-platform DJ mixing software. To the left is Carla, an "audio plugin host" which is something that I use as part of my Linux audio stack (it's a graphical program that allows to run equalizers, noise filters and other plugins on top of JACK). Hope you enjoy the set!
I tested it. My speakers are inbuilt to the Monitor and connected via HDMI. It is not working for me either. So yeha, maybe that's the reason.
Alternatively you could experiment with Jack Audio if you have time and the will to learn it. It is not the easierst stuff.
How are your studio monitors connected to your computer? Directly to the 3.5mm jack, or through and audio interface?
Edit: to elaborate, if you have an aduio interface that your studio monitors are connected to, you could record through that device and monitor that way as well.
Another option is to use the ASIO4ALL driver (on Windows) as your ASIO driver, and then use the control panel to set your input and output to different devices. That's one big bonus of using ASIO4ALL.
A third option would be to use Jack to route the audio in and out. https://jackaudio.org/ Jack isn't all that easy to use, so if you don't have a separate audio interface, ASIO4ALL is likely your best bet.
The scarlet focusrite interface costs like 100 EUR, that's cheaper than most amp for a better result.
All the rest can be achieved with free (as in free speech) software. Jack would be the interace between all the software you use. My typical chain is Interface ---> Guitarix --> Speaker but I can send guitarix to ardour or add some effect/balance before if I want
Maybe have a look at jack. With this, you can route from one software to another. I'm not sure if you can capture system sound output on Windows (maybe just one software at a time) but I don't see why not
You make it sound easy but I have spent hours trying to figure out how to use JACK when I switched from Windows to Linux a few years back. The my statements are based on what I was able to figure out from tinkering with it and searching online. JACK is simply not plug and play and difficult to set up for anyone who is not very experienced with Linux. All of the instructions for using JACK that I have seen told me that I have to stop Pulseaudio if I want to run JACK. Here is a good example: https://jackaudio.org/faq/pulseaudio_and_jack.html
I for one want to see more people discover the great things about Linux start using it but when it comes to Audio Production, Jack is a huge roadblock for those who don't know all the ins and outs and are new to Linux.
.profile
+export QT_AUTO_SCREEN_SCALE_FACTOR=0 # Not sure if this does anything
.bashrc
+#Disable nnn auto-open on right or l key +export DISABLE_FILE_OPEN_ON_NAV=1; + +export LC_ALL=en_US.UTF-8 +export LANG=en_US.UTF-8 +export LANGUAGE=en_US.UTF-8
.bash_aliases
+alias g="w3m https://google.ca" +alias t="tree --filelimit 500 -C | less -R" +alias ll="ls --color=always | less -R" +alias jack_docs="w3m https://jackaudio.org/api/jack_8h.html" +alias cldoc="w3m http://www.lispworks.com/documentation/HyperSpec/Front/X_AllSym.htm"
If you would want a audio patch system specifically designed for low-latency, only Jack comes to my mind, it also exists for Windows. Here's some reading material about its latency (there's probably much more around).
On Solus I originally had a bit of an issue with getting it to work. I'm quite certain it was my fault, but since it's a while ago I can not tell exactly what I did I'm afraid...
I remember getting help from this Forum post though and following the official setup according to the Jack site.
For me I did not have to touch anything related to Alsa. Main point seems to be editing the config-file correctly and being in the audio group.
Sorry for a bit unclear answer. Hope it helps somewhat.
My understanding is that Jack (free) and Voicemeeter are the same things but for PCs. Not sure if just renaming the audio device is a viable strategy on PC though.
You could use Jack for that. That is a framework designed for real-time audio routing and processing, especially for music production and other demanding audio applications. Setting it up is not as easy as for Pulse Audio (which 'just works' on most Linux system), but when done right it will allow you do do everything you want and more.
You would have to set up Jack to run together with PulseAudio. Usually Jack handles the audio hardware in such case and connects ro PulseAudio is connected with jack sources and sinks. Applications that support Jack directly connect to Jack server, everything else goes through PulseAudio. With tools like QJackCtl or Carla you can then connect the applications and hardware in whatever way you want. Inside Carla you can also load audio processing plugins (LV2, VST or LADSPA) for even more flexibility.
You can try Jack OS X - https://jackaudio.org/
I have used IShowUStudio on OS X before to record a mic, the output from Pro Tools, and screen record. Not free, but I think it's about $15, and I ended up using it after messing with OBS for a couple of hours trying to record both system sound and an external input, which turned out to be, as far as I could tell, impossible for OBS to do.
The trick with IShowU is it sets up an aggregate audio device that takes external and system sounds as inputs, then uses its internal mixer to give you the output of that without creating a feedback loop.
You will need to start JACK from qjackctl before any application will be able to use it.
You will also need to route audio from musescore to an output.
There are some great tutorials online about how to do this. Have a read through this: https://jackaudio.org/faq/jack_on_windows.html
I've finally found the time to set up the pedals with the synth. Cardboard pedal board because the only place left was a corner of the spare room, but I quite like the cushion on the floor setup, very meditative :)
It goes Palmer Daccapo (line to instrument level, ground lift) - Fuzzrocious Bongripper (gain, overdrive, distortion)- The Zero Fret Serpent (octave, distortion à la Life Pedal)- Zoom multistomp (delayyyyyy).
I'm using Linux (Ubuntu) with jack and Ardour. The 2i2 just works out of the box. You have to select it as input/output in jack's advanced settings tab. Then configure jack for real time scheduling: https://jackaudio.org/faq/linux_rt_config.html
In Ardour, you can select the 2i2 as input, and done. Figuring out Ardour is a harder task.
So far, I've noticed: the ground lift is really nice to cancel hum, and the effect of the distortion pedals is way more subtle than on guitar, but maybe that's because I'm monitoring on headphones and I keep the volume low? (because I've already got tinnitus and I'm scared to death it'll get worse)
I think you can find something useful on the sections of the applications tab from JackAudio
Also, have you tried changing the Media Player?
Good alternatives are also on the same site above
Edit: I've said about the media player because sometimes changing the equalizer on them gives better result than system wide settings
There's a support thread with some recent entries on the BUTT SourceForge page here. The suggestion there is to run the following command in Terminal:
xattr -cr /Applications/butt.app
Then, try opening the app via LaunchPad. Let me know if it works for you.
​
And FYI, once you get BUTT working you might also need JackPilot to capture the audio stream from your hardware. I'm no expert on it, and configuration will vary depending on your setup, but if you need help I might be able to point you in the right direction.
I’m on a Mac, and I’m sorry that I don’t know Windows very well anymore.
Solution number 1 is to manually patch an output from your computer back into an input with actual audio cables. This isn’t easy at all if all you have is a 2-in/2-out interface, but if you have access to more ins and outs, this is the way to go.
If that’s not possible, there are software solutions to do essentially the same thing; route audio from your computer’s output or from individual apps back into another app. On a Mac we’d use JACK Audio, Soundflower or Audio Hijack. JACK has a Windows version but the others don’t. Google also suggests that Total Recorder might be a good alternative on Windows.
"Most users do not need an RT kernel in order to use JACK, and most will be happy using settings that are effective without an RT kernel." https://jackaudio.org/faq/realtime_vs_realtime_kernel.html
I think in the past rt kernel was a must though (circa 2005)...
I followed https://jackaudio.org/faq/linux_rt_config.html. However I got the same error upon starting jack (via cadence). My solution was (as root):
[1] add a file: /etc/security/limits.d/jackuser.conf with the following two lines:
@jackuser - rtprio 95
@jackuser - memlock unlimited
Permissions -rw------- were sufficient in my case.
[2] add a new group jackuser
[3] add myself to group jackuser
This was sufficient to silence the error. However I haven't investigated the quality of scheduling yet...
Making it clear.
by the post you made, I guess you're using pulseaudio or alsa for redirecting output which won't work in any case.
You need this, GUI helper for JACK is cadence
You need JACK Audio Connection Kit! :) It's supported on all platforms, incl. Linux, it does what you need and even more.
Probably not the only software that exists to do this kind of stuff, but it's the one that pop up into my mind.
Edit: I'm not a Mac expert, but I think MacOS have its own way to manage audio and midi routing (Sunflower maybe, after a quick google search?). There is also Virtual Audio Cable for Audio on Windows.
Rewire is old and has some janky limitations. FL as a vst is a new alternative for rewire client mode, but there's no real alternative to rewire host mode. It may be possible to rig something up with the Mac equivalent of virtual audio cable, but you might be out of luck. (look into the JACK program maybe?)
Good luck.
I think you could do what you want with Jack. However, you may get some sync issues. Take a look on this article: https://jackaudio.org/faq/multiple_devices.html
That being said; you should consider some of the options the others here are suggesting. You can get a really nice, compact and easy to use multi track setup that is both relatively cheap and more stable than with separate USB mikes.
I prefer using the line level input on the 2i2. Usually from a board or rack unit. The 2i2 isn’t terrible, it’s just not the best suited for hi-Z input. It’s microphone preamps are pretty decent quality, but those are as mentioned microphone preamps. And here lies it’s strength.
There are two versions of jack. Read here https://jackaudio.org/faq/
I use jack2. Jack(1) is using the jackd command. And jack2 is using a series of scripts/commands based on jack_control, but should be backwards compatible and still have the jackd commands usable. If you use jack2, the command to check if it’s started or not would be jack_control status. If you use jack(1), just do a `ps -A |grep jack’ to see if it’s running. And use kill to nuke it. Scroll through this and you’ll find other useful commands. https://wiki.archlinux.org/index.php/JACK_Audio_Connection_Kit
My use case is "run JACK and programs using JACK" which is essentially what every single professional audio users on Linux do. JACK has a guide here and they claim that you don't need a realtime kernel. I beg to differ. If you're in the studio recording a band, you do not want even the possibility of xruns, or even worse, if you're on stage performing for a crowd. Again, right now I'm not using a preemptive kernel anyway, but I am not chancing anything if I were performing or recording.
https://jackaudio.org/faq/realtime_vs_realtime_kernel.html
Again it's ALL in the configuration of process priorities.
THAT WILL GET YOUR LATENCY IN CHECK.
Installing a RT kernel patch MAY help. But not for the reasons you might think. And it MAY again because of the configuration required, may actually INCREASE your latency depending on your workflow if no configuration is done.