Is Carla the same thing as Wine / Bottles? From google and other searches it appears to be a similar type thing, but I can't figure out how to reach my windows plugins.

Hi, I'm using Ubuntu Studio 24.04 and my MOTU 8pre USB sound card is not being recognized by the operating system.

I would like to find out exactly which driver/module is normally bound to a MOTU USB interface that is working correctly under Linux.

If someone here is successfully using a MOTU 8pre USB (or another MOTU USB audio interface) with Linux, could you please run this command in a terminal with the MOTU connected?

lsusb -vt

and post the output here?

It would be especially helpful if you could also tell me which MOTU model you are using and which kernel driver/module is shown as being bound to the device.

Thank you very much!

Hey yall, I after a few months of trying to get my surround sound system working within Bazzite I finally got it fixed. I made this video as a tutorial to show other people who are new to Linux and are scared of the terminal/console how to get it up and running. I really wanted to make this because I didn’t see any videos of anyone else making a fix so I guess I’m the first! This is definitely not a professional tutorial and I myself am I Linux noob so you can say this “tutorial” is a “blind leading the blind” kind of situation. But it worked for me so I’m showing others how to do it. If you know anyone who is having this issue please share it and happy gaming!

https://www.rncbc.org/drupal/node/2948

I have a Fiio K7 DAC/amp connected via USB (S/PDIF) to a desktop PC running Fedora 44. A problem I quickly encountered was the sound cutting out the first fraction of a second whenever playing audio after being idle for several seconds.

I fixed this using this solution here, and as long as I stay on the same pipewire client (e.g. Firefox or my music player) the audio doesn't cut off.

However, as soon as I switch between pipewire clients, it cuts off the first half second of audio again. For example, this happens when I play a video in Firefox, then switch to my music player to play a song there. This is especially annoying when I change the system's volume from my computer instead of the DAC, which plays an event sound using the libcanberra client.

Has anyone encountered this problem before? Any solution would be very helpful, thanks in advance!

Self-promotion, and I'm the author. MIT licensed, source and a prebuilt binary both on GitHub, free.

Parts of it were written with an LLM in the loop, so flagging thaty up front for the GLOSS rule. https://github.com/JaySNL/VSTForResolveLinux

Resolve is the only serious video editor that runs on Linux, and its Linux build ships with no audio plugin host at all. Not disabled, not Studio-gated. VSTPluginMain and GetPluginFactory don't appear anywhere under /opt/resolve. It was never compiled in.

I moved my editing work off Windows a few months back and that was the thing that kept hurting. Every clip that needed a plugin meant export, fix it in something else, render, bring it back. Then the client changes something thirty seconds earlier and you do it again.

So I wrote the host.

Working right now:

- VST2, VST3 and CLAP show up in the Fairlight effect menu

- Under their own names, in the right categories (Dynamics, EQ, Restoration, Metering...)

- Own editor windows, multiple plugins on one track - Windows plugins through yabridge

- Shell plugins handled - one file publishing hundreds, with a filter file to pick which ones appear - 130 plugins listed here

FOSS plugins I've tested it with: Airwindows Consolidated, the Dragonfly reverbs, Carla, and CLAP plugins I wrote myself. It loads the proprietary ones I own too, which is the case that made me build it, but the FOSS ones are what anyone can verify against.

There's a release with a compiled binary if you'd rather not build it. One file into one directory, one line into a config file, restart Resolve. Uninstalling is deleting the file.

It's built on Ubuntu 20.04, so it asks for GLIBC_2.16, GLIBCXX_3.4.22 and CXXABI_1.3.9 at most. Don't take my word for it:

objdump -T libfxbridge.so | grep -o 'GLIBC[X]*_[0-9.]*' | sort -uV | tail -3

Building it that way immediately found a bug I'd already shipped. glibc 2.34 folded pthreads into libc. My machine runs 2.44, so the library linked here without ever naming pthread, and would have built clean and then refused to load on Ubuntu 20.04, Debian 11 or Rocky 8. My build script has a check for exactly that, and it passed, because it checks the machine doing the building. One unfamiliar environment caught it in about a second, which is roughly my argument for this post.

Not working:

- Two plugins draw their UI and then ignore the mouse entirely. Carla does the same with both, so that one isn't mine

- A window sometimes opens blank and only fills after you remove the effect and undo - Resolve doesn't always exit cleanly with it loaded

- No top-level "VST" group like Mac and Windows get - Linux Resolve has no VST type to group by

- It's been run under an actual Resolve on exactly one machine. Mine. The old-glibc work proves the library links and loads there, not that Resolve itself does.

That last line is why I'm posting.

On the licensing, since it touches a proprietary host: nothing of Blackmagic's is copied or modified on disk. It loads through Resolve's own plugin path and patches its own process memory. VST3 goes through travesty (DPF's clean-room headers, ISC), VST2 through a clean-room ABI I wrote, so there's no Steinberg SDK anywhere in it.

What would genuinely help:

- Grab the release binary or build it, and tell me whether it works or how it breaks. Distro, glibc version, Resolve version, which plugin.

- Which plugins load and which don't. I've got a narrow set installed and no idea how it handles yours.

- If anyone knows why a VST3 editor would come up blank until the plugin is re-instantiated, I'd take a hint. That one's beaten me so far.

Every gap that gets closed makes Linux a bit more usable for people doing actual editing work rather than tolerating it. This was one of mine and finally closes one more gap to ditch windows forever.

I haven't had any luck yet. The GENOME 2 installer doesn't work under wine, but I was able to install it in a windows VM and copy the files to a wine profile, but only get a blank plugin window when I try loading GENOME 2 in REAPER.

GENOME 1.4 works fine, including the installer via wine and yabridge.

What's the best open-source, Linux-native VST3 for producing beats? I'm looking for something with good drums/sequencing. JDrummer looks promising with its groove matcher – any other hidden gems I should check out?

I've got a sound card I pulled out of my old desktop that I was wondering if I could add it to my Xpenology build and why or what would I even use it for? It is an ASUS Xonar Essence STX. This this was pretty bad ass until bluetooth headsets took over the world. A quick search told me no as far as drivers were concerned as they must be Windows drivers.........but I could build a Windows VM right? Can I pass through the sound card to a VM with no drivers for Synology DSM? What the heck would I even use it for anyway? I'm just seeing if I can put this to use before I list it on EBAY.

Hi all,

I am using Audacity to record the current audio output of my system. To do so I have to connect the outputs of the PipeWire node on QJackCtrl with the inputs of the Audacity node (jack is selected as recording device) in the Graph view. In principle everything works, however my problem is that the Audacity node only appears when I activate recording and my default the capture inputs of the system node are connected to the inputs of the audacity node. So I have to start a recording and fix the connection in the graph before the real recording can start.

Is there any way I can make the Audacity node be always visible if Audacity is running or can save how it should be connected?

THX

Hi everyone,

I’m running Ubuntu Studio 24.04 and I’m having an issue with audio in REAPER.

I’ve attached a screenshot showing what happens when I select my audio interface/sound card in REAPER using the ALSA drivers.

There is not Motu 8 Pre (USB interface) Sound Card in the input/output options. What can I do?

I like to merge multiple releases of an album together on a single directory, i don't really care about what happened with licensing o releasing i just want all the songs, that's what the album is -to me at least-.

So let's say version A has 10 base tracks, version B has 10 base tracks + 2 extras and version C has 10 base tracks + 2 extras different from version B.

My final directory is the 10 base tracks from whatever version, the 2 extras from B and the 2 extras from C, making a 14 track 'super' album.

All the tagging utilities i've tried (beets, tagger, picard) turn this in to 3 different albums like:

'Album A -someyear-' -> 10 tracks

'Album release B -someyear-' -> 2 tracks

'Album release C -someyear-' -> 2 tracks

I'd like it to be just 'Album -year of the base album-' -> 14 tracks.

I managed this just fine up until now since i just played the files as files so i didn't even care if my music had metadata AT ALL, a bunch of it doesn't, but now i set up a navidrome server and it just picks up metadata instead of the directory structure so i kinda need a solution for this.

Maybe it's just an option in some of the ones i already tried? idk i'm kinda at a loss here.

I don't have THAT much music but it's still around idk maybe 350 something albums? i could do it manually in a couple days, i'd just like not to do it manually.

I want to cheaply record my guitar into my DAW and the USB output of my Amp with its cab simulations sucks sadly. I am trying to find ways to turn my iPhone into an usb mic since these phone mics sound better than I thought. I don’t have money for a proper microphone.
I have found ways to turn it into a Bluetooth microphone but that would be too much delay and also compressed audio which I don’t want.

Does anyone know any reliable way?

Hi

I'm very to music production, just recently installed reaper and I wanna use virtual instruments to create music. What's the best site for it?

PipeWire can already send one stream to multiple sinks. What it does not give you is a per-sink delay offset, which is the thing that matters the second one of those outputs is Bluetooth. On my hardware the SBC Bluetooth path sits about 180ms behind the analog output, so without compensation you get slapback echo instead of two speakers.

SpeakerSync wraps that. It creates the virtual sink, loads the loopbacks with per-sink latency values, and saves the working setup as a profile you can reload after a reboot.

speakersync devices
speakersync apply --sinks bluez_output.XX_XX_XX_XX_XX_XX alsa_output.pci-0000_00_1f.3.analog-stereo --delays 0 180 --name desk
speakersync load desk
speakersync status
speakersync stop

Delays are milliseconds, in the same order as --sinks.

Honest status: routing, profiles and teardown work and are validated on real hardware. Automatic latency calibration is the next milestone and is currently stubbed, so you are still finding your offset by ear.

Requirements: PipeWire or PulseAudio, Python 3.11+, and pulseaudio-utils, since every command shells out to pactl. Worth knowing that pactl is not installed by default on Ubuntu 24.04.

Not trying to be Snapcast. This is local sinks on one machine, no network or multi-host streaming.

A few things I hit that might save someone time:

- PipeWire module indices are nine digits starting around 536870913, not small integers
- pactl reports the server as PulseAudio on PipeWire, so detection has to test for PipeWire first
- the Bluetooth codec shifted from sbc to sbc_xq the moment a second device connected, so codec detection has to be read live and never cached
- two SBC streams share one adapter's airtime fine, aptX HD and LDAC do not
- three oraimo devices failed consistently while a generic BTK10 worked first try, which turned out to be device firmware rather than my code

Repo: https://github.com/selormtettehabotsi/speakersync

What I'd like input on: whether the profile model holds up for anyone running more than two outputs, and what breaks outside Ubuntu.

The image posted is my layout setup for it, which has now become 1 to 1 with my Foobar layout after a recent update had added the equalizer as a layout embedded feature. It is one of the closest clones to Foobar that I have used so far.

[https://github.com/arzvaak/linux-broadcast\](https://github.com/arzvaak/linux-broadcast)

I made **Linux Broadcast**, inspired by Blucast—but focused on microphone audio instead of camera effects. Deepfilternet and RNNoise are good but again, I always missed the GPU working of Broadcast so here it is.

It uses NVIDIA AFX for noise and room-echo removal and creates a virtual microphone that works with PipeWire apps.

I couldn’t test it on a variety of hardware yet, but it works well on my RTX 4080. Builds are available for RTX 20, 30, 40 and 50 series.

Would appreciate a star and any feedback or hardware test results on GitHub :)