Hello,
I'm new to audio production in Linux, so apologies in advance if I will
use wrong terms and bad jargon. I'd like to record my guitars and bass,
but I've found a problem which I think it can be JACK-related.
I bought 2 USB audio interfaces, a 2-channel Behringer which I returned
and a Focusrite Solo 3rd gen, which I'm testing right now.
The issue that I have is with the programs that use JACK, like Guitarix
and Reaper. It seems I can only get distorted sounds out of them and
both have JACK as input source.
Instead, Audacity, which lets me choose the hardware audio interface
directly, lets me hear the sound of my instrument just fine.
What I've tried so far:
- my everyday Ubuntu 18.04 with low-latency patch: Guitarix and Reaper
distorted audio and Audacity just fine
- live AVlinux: Guitarix distorted audio and Audacity just fine
- live Ubuntu Studio: Guitarix distorted audio and Audacity just fine
Can someone please help? What else can I try?
I'd like to stick with Ubuntu 18.04, as it is my main everyday desktop.
Thank you.
Roberto
Hello I am using Jack on Windows machine. I am using the Jack to process audio in real time with few very CPU hungry VST plugins. I have a computer with 4 cores CPU but it is old machine. The playing software did not utilise the cores as it suposse to be. I have got an idea to use VST hosts, connect them and with Process Lasso to force the right core utilisation. And I have succeded the goal. All cores are utilised almost at the same level. My card is an USB interface with max buffer 4092 and ASIO driver. I have configured the Jack with maximum possible frames/buffers. With full set of VST plugins I am getting DSP load 100% what is total disater and in the same time CPU core utilisation in around 30%. Is there any way to solve that problem. Or it is just a kind of bottelneck connected with my USB interface or Jack? Regards
Hello jack community,
I am trying to run two separate Jack instances, one for low latency monitoring using Carla as a plugin host and one for recording using Ardour with a higher buffersize for performance. The two jack servers are connected by zita-njbridge.
Its working quite well so far, the issue is that I cannot compensate for the latency of the zita-bridges, 2*10ms. To my knowledge, jack provides latency information to Ardour, so i need to inform jack about the latency happening. I tried the following setups:
Main Jack, alsa backend using USB Interface (Presonus Audiobox USB96)
Ardour Jack, dummy backend
and
Main Jack, alsa backend using USB Interface
Ardour Jack, alsa backend using onboard audio (Intel 8 Series/C220 if it matters)
Signal way in both cases for recording is:
System in Main Jack - Zita to Ardour-Jack - Ardour - Zita to Main Jack - System out Main Jack
So I never use the onboard physical connections.
The first setup is problematic because the dummy backend doesn't have extra latency options, the latter seems to be ignoring the -I -O numbers in the Ardour-Jack, at least jack_iodelay tells me the same frame count no matter what I enter, which is not the case in the Main-Jack, where I can get it to 0 with the correct settings.
So I get late recordings by 20ms from zita + the latency of the Main-Jack, that the Ardour-Jack doesn't know about.
Is there any way to make this work? And why is the Ardour-Jack ignoring my extra latency settings?
Even though I don't think it matters too much, I'm on Arch Linux, and using jack2 1.9.16-1 and Ardour 6.3-3.
Looking forward to your replies, stay safe and healthy!
Best regards,
Robin
Hello, all
I'd like to introduce you to a new midi tracker that I wrote. It's very
minimalistic, a glorified repeater really. The main idea of the program was
to have something to put between Carla and an external MIDI controller.
selling points:
- it only knows Jack
- fast workflow
- designed for live performance
- irregular looping of tracks (have to see to believe)
- funk mode (nice for triplets)
- midi triggers
- freewheel rendering
Of course it is released under GPL3, otherwise I wouldn't be bothering you
kind people. It compiles and runs on latest Ubuntu and Fedora (developed
under Mint). Gnome app written in Python. The sequencing engine is in good
ol' C (with jack being the only dependency).
You can find the project at:
https://github.com/rdybka/vht
You can also check out:
https://sourceforge.net/projects/vht
for a current tarball and a deb for Ubuntu
Hope someone besides me will find it useful
Regards, Rem
Hi, Jack people !
I am using the new Windows build of Jack Audio. I am connecting a Roland
DJ-505 controller that has lots of inputs and outputs. I may be missing
this somewhere but is there a way to make Jack (and the new session
manager) use the names of the ports that are provided by the DJ-505 ASIO
driver ? If not then I'd like to request that this feature is added to
Jack Audio as it must be relatively trivial to copy the names from the
ASIO driver.
As to the Windows build. Great ! Many thanks to the developers :) ...
I'd almost given up on a new Windows build turning up. I use a lot of
Windows software such as FL Studio and Traktor.
Michael
I have two usb audio input devices that I'd like to mix and connect to
Jamulus, an application that uses jack (not pulseaudio). I'm trying to
find out if this is possible.
I'm running jack2 on a debian 10 system. There's one audio card (chip,
actually) that's listed as HDA Intel.
FYI The usb input devices are a Shure MV5 microphone and a Scarlett Solo
audio interface.
I can connect either of these devices to jack by specify one from
qjackctl setup: Settings -> Advanced -> Input device.
I don't see how to capture and mix the input from both devices. I
haven't found any documentation that explains how to do this or if it's
possible.
Thanks
Hey, new subscriber here. I've been using JACK for several years but
know just enough about it to get it to work (usually). I've hit a couple
of issues with a new system build that have me stumped.
Specifics of the system: Ryzen 3700X; Asus Prime X470 Pro motherboard;
using (or at least attempting to use) the on-board audio, which is an
ALC1220. OS is Ubuntu 20.04 LTS; JACK is configured to run with realtime
priority.
Issue #1: Can't get any output from the digital (S/PDIF) port. I've made
sure S/PDIF is unmuted in alsamixer, and the S/PDIF output works fine
with PulseAudio. I've tried outputting to all of the system ports that
show in the JACK connection manager, and none of them produce any output
on the S/PDIF port.
Issue #2: If I run with the default number of output channels (6), JACK
stutters horribly with constant XRUNs. Already tried changing the
--period option; if I increase it JACK fails to start, and if I decrease
it the stuttering gets worse. Also tried changing --nperiods; increasing
it prevents JACK from starting. If I disable the last 2 channels (i.e.
run in 4 channel mode) the stuttering goes away.
I can live with #2 (don't really need the last 2 channels), but if I
can't get #1 resolved I'll be looking at a 3rd party soundcard (internal
or external) for this system.
Any thoughts/insights would be welcome.
Thanks!
Hi,
Can you please put back link to downloads for windows and macos? Now
people get the wrong impression (at least one person have expressed
this opinion to me) that the downloads are gone because jack is
unstable on those two platforms, which they are not. (qjackctl is
though, but not jack itself).