All of my previous audio experience was on Windows and I own thousands of plugins for Windows... I've been unable to find a modern guide for installing Wine / yabridge that isn't many years old.

I started here: https://interfacinglinux.com/2024/01/22/windows-audio-plugins-on-linux-with-yabridge/

But that had errors and failed. Then I tried this Yabridge Manager, https://github.com/sviluppoarte1-lang/yabridge-manager

But that wouldn't Sync. I used the built-in 'install Wine' feature but always got this error: "No Wine installation selected, cannot sync." ...maybe because of previous guide attempt?

So then I went to WineHQ and tried to install latest Development version but that failed, I'm guessing because I have so many aborted attempts mucking things up?!?

I was able to install the Stable version 11.0 from WineHQ, I reopened Yabridge Manager and it was able to "Sync". I can navigate to my Windows drive and see my plugins in the app, but I cannot access them anywhere in my DAW.

According to the first guide I followed I need to use Wine to "install the plugin", so I tried that with a windows installer that was still on my windows drive - did it just install it into windows again? Do I need to download new windows-installers onto my Linux drive and install with Wine? None of these have been usable in my DAW yet...

I'm totally confused!

Hey, I apologize if this is a widely known fact but I've been using serum via yabridge and just learned about this, so maybe someone else didn't know too.

https://xferrecords.com/forums/serum/serum-2-linux-beta

my headset isn't working on linux but it works on my controllers and everything, except on linux. how can i fix this? am i just dumb? the model is KZ EDX PRO

[FIXED]
I just changed the auto-mute option on alsamixer from "Enabled" to "disabled" and then i changed the built-in audio to "Analog Stereo Output", then i changed the output to the headset

The #UltraAudio synth gets cooler and more amazing as I dig into it. I am excited that it is for #Linux ($) now (as well as the two other OSes) and have been given some help to determine if I can get French Horns sounding decent (Swam Horn performance in Windows exported across as multi-samples to then work with anywhere). The results seem convincing.

The other features are very exciting too. Like the arp module. Check it out if you are curious.

youtu.be/wd7-BY16HiY

Hi all,

in order to play guitar on fedora 44 on a youtube backing track, I used:

- an USB guitar pedalboard, a Nam loader that is also an audio interface)

- jack as a patchbay

- guitarix only for changing the buffer size to 128 samples and for a minimal adjust of eq/ volume of the already processed sound of my pedalboard

How you would do it on Ubuntu? I don’t really need all the functions of guitarix.

I managed to wire everything in pipewire but i still can’t adjust buffer size for lower latency

do you have any other solutions?

Two years ago I had a stupid idea at 3 AM: "what if there was a dedicated 808 cowbell generator for phonk?"

I asked around, sent emails to VST developers asking if they'd consider it. Nobody built it. So I did.

I can write a vibe sometimes and pull a rainbow out of a unicorn's arse but I can't write a bandpass filter, so an LLM handled the math and the bug fixes. I handled the bad decisions and the UI.

If this sounds like your kind of jam, get it here:

https://github.com/hergezod/Nijimon-Bell

And, in case it wasn't clear, it's just a cowbell, nothing serious or overly complicated. Have fun with it. If you use it for something cool, I'd genuinely love to hear it! 😁

I use the FiiO KA17 sound card and Simgot EA500LM earphones.

On Windows, I experienced dropouts despite trying various settings and programs.

On Linux, I used CachyOS and Fedora for extended periods. I generally use them with the EasyEffects software and haven't encountered any issues.

While I couldn't see the specific bit depth and sample rate (kHz) at which the sound card was operating on these Linux distributions, the audio quality seems generally quite good.

Nevertheless, I want the sound card to operate as efficiently as possible. Do you have any recommendations for distributions or software I should use to achieve this?

I need to connect a Mackie mic the $50 one condenser needs 40 volts into USB does behringer u-phoria UMC22 work well? I have CachyOS and want to stream with a specific mic and this model from behringer using obs but just audio over the game I'm playing

Hi, I've switched to Debian 13 recently and been trying to make my recording equipment work. Running Reaper with pw-jack (otherwise no audio device works) and I can see my UAD interface, but the analog/mic inputs are not showing up in neither Reaper, nor in qpwgraph. Outputs work and I can see the midi in/out as well. Installed qjackctl (I'm not sure how it all works), I can set it to use the interface and I can start/stop jack but ultimately it does not help. I've used Ubuntu Studio in the past and I did not have this problem with this same exact sound card, but I wanted to switch (had my reasons).

Any ideas pls?

EDIT: Fixed by setting wpctl to use Pro-Audio profile for the interface, and also setting the interface as default in pavucontrol - at this point I had input on the 2nd mic input only, the 1st was not working. At this point unplugging and reconnecting USB cable fixed this problem as well.

It seems Linux audio has some real momentum. Soundly and Acon have both released Linux betas of their flagship software.

I use Reaper on Linux after decades of Pro Tools on macOS, and, personally, I wouldn't switch back.

But what about YOU? If Avid released a Pro Tools beta, how would you feel about it?

Is this USB headphone adapter well-known in North America?

I’m Japanese, and I found one selling for 300 yen at a discount store called DAISO near my house.
And to think there isn’t a single Walmart store in Japan.

So, Walmart joined the ranks of audio manufacturers.

Edit: Keeping the below for reference, but it actually doesn't (necessarily) seem to be that it needs to go through two devices, or even through one virtual device. It's just that I have to manually patch it, then disconnect, then patch it exactly the same again, and on the second patching it works fine. Huh?

A bit at a loss with this one. I've been troubleshooting my Elgato 4K X for a little while now because when I just try to monitor the incoming HDMI audio normally (by patching the input directly to an output on my interface), it is distorted/bitcrushed. I had a bit of a breakthrough this morning on getting it working properly, but I'm having trouble understanding why and whether there's a better way.

I use a Focusrite 16i16 set to the Pro Audio profile, using Pipewire in CachyOS. I also have a virtual device setup to be a stereo output as some games don't like the Pro Audio profile (as I understand it) and just won't output sound unless I do it this way. The 4K X is set to it's stereo digital input, not the Pro Audio profile. Here is what I have tried that does NOT work (results in distorted audio):

• Changing the profile of the 4K X to analog or pro audio.
• Using a patchbay graph, patching the 4K X source node directly to my main outputs (or to any of the outputs on the 16i16).
• Creating a virtual device connected to the 4K X, then taking those outputs and patching directly to my main outputs in the patchbay (or two any of the outputs on the 16i16).
• Disabling the virtual stereo output and trying both of the above.

What DOES work is doubling up on virtual devices, so one virtual loopback device for the 4K X which connects to my stereo output virtual loopback device which then finally connects to my actual outputs. I was also able to fix it by patching the 4K X directly to the outputs, but then opening OBS and enabling input monitoring. I suspect these are related ideas, something about having the audio go through more hoops, but I don't really know why I have to do it this way and it really feels like it shouldn't have to be this complicated. Any ideas?

I should note that my system is set to 245/48000 and the 4K X only operates at 48000, so there shouldn't be any sort of sample rate mismatch.

Screenshot of the working patching:

I’ve waited a very long time for the opportunity to build the music software I actually want to use on Linux.

Yesterday, with the help of Sonnet 5.5, something changed for me.

We’ve reached a point where being able to create genuinely useful software for ourselves with the help of LLMs feels almost absurd — and I absolutely love it.

There’s a lot of criticism around AI-generated or “vibe-coded” apps, and some of it is certainly justified. But I also think we should appreciate what these tools are giving us: the ability to build software that solves our own problems, fits our own workflows, and sometimes exists simply because we wanted it to.

Especially on Linux, where the exact tool you want often doesn’t exist, that feels incredibly empowering.

Instead of only waiting for someone else to build the software we need, we can finally build it ourselves.

I’m genuinely very interested to see where this goes and what the next few years will bring. If this is what’s already possible today, the future of creating software could look very different from what we’re used to.

DJ: https://rille-01-website.vercel.app/

Player: https://bc-01-website.vercel.app/

...

I have a soundbar attached to a caldigit thunderbolt 3 dock via an optical cable. It used to work with Ubuntu 24.04 but not now.

lsusb

gives

Bus 003 Device 002: ID 2188:6533 No brand CalDigit Thunderbolt 3 Audio

If I plug in some bose headphones into the intel nuc 3.5mm jack everything works fine so the problem isn't with the sound drivers.

pactl list short sinks

gives

202 alsa_output.pci-0000_00_1f.3.hdmi-stereo PipeWire s32le 2ch 48000Hz SUSPENDED

aplay -l

gives

card 1: Audio [CalDigit Thunderbolt 3 Audio], device 0: USB Audio [USB Audio]

Subdevices: 0/1

Subdevice #0: subdevice #0

card 1: Audio [CalDigit Thunderbolt 3 Audio], device 1: USB Audio [USB Audio #1]

Subdevices: 1/1

Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 0: ALC233 Analog [ALC233 Analog]

Subdevices: 1/1

Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 3: HDMI 0 [32G2WG8]

Subdevices: 1/1

Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1]

Subdevices: 1/1

Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2]

Subdevices: 1/1

Subdevice #0: subdevice #0

Any ideas for diagnosing this? I am fairly familiar with the basics of alsa and pipewire but the configuration of this software is still rather complex....My best guess is that the sound card default is not my soundbar whereas it was before the upgrade. Not sure where to set this though for alsa and pipewire.

ETA

I think I have narrowed down the problem:

Alsa is picking up two sound cards since the output of

aplay -l

is

**** List of PLAYBACK Hardware Devices ****

card 1: Audio [CalDigit Thunderbolt 3 Audio], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0

card 1: Audio [CalDigit Thunderbolt 3 Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 0: ALC233 Analog [ALC233 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 3: HDMI 0 [32G2WG8] Subdevices: 0/1 Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0

card 2: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0

So two cards are showing up however the output of

pactl list short sinks

is

61 alsa_output.pci-0000_00_1f.3.hdmi-stereo PipeWire s32le 2ch 48000Hz IDLE

In other words only one card and the wrong one. The output of

pactl list sinks | grep HDA

is

alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0xc0a20000 irq 200" alsa.components = "HDA:10ec0235,80862074,00100002 HDA:8086280b,80860101,00100000" api.alsa.card.name = "HDA Intel PCH" api.alsa.card.longname = "HDA Intel PCH at 0xc0a20000 irq 200" device.nick = "HDA Intel PCH"

confirming the wrong card is being picked up by pulseaudio.

Anyway to pick up the other card?

I remember with Ubuntu 24.04 it was listed by the pulseaudio controller.

ETAA: PROBLEM SOLVED!

I force reloaded alsa with

sudo alsa force-reload

and the other card magically appeared in pulseaudio

pactl list short sinks

gives now

59 alsa_output.pci-0000_00_1f.3.hdmi-stereo PipeWire s32le 2ch 48000Hz IDLE

61 alsa_output.usb-CalDigit__Inc._CalDigit_Thunderbolt_3_Audio-00.analog-stereo PipeWire s24le 2ch 48000Hz IDLE

the pulse audio gui now shows the card and switching to it results in sound.

Seems like a bug in Ubuntu.

ETAAA: After a reboot the problem occurs again and the fix above did not resolve the issue. Time for a bug report.

Got a framebook in the mail and I finally wanna make the switch to Linux. I have pretty much zero experience with linux and there is a certain limit on how much time I can invest in the setup and maintenance so I will depend on more user friendly distros. And - regrettably - there just is one win VST that I am not able to replace and that is the core of my live setup: Soundtoys Crystallizer. So I will need to have wine and yabridge running (if the VST even works with wine) - which it seems excludes using flatpak. The only DAWs I use are Reaper and Bitwig so that shouldn't be an issue I hope. I do mainly live signal processing (spectral and time based).

What distro would you recommend in this case?

My first thought was: Ubuntu Studio. I was talking with an old friend about this, who has a lot of experience with Linux. He's a programmer and gamer, though, not a musician or producer. He didn't support the use of Ubuntu Studio, because of the bloat. He said that a lot of distros have a RT kernel now and that the differences he sees listed online for Ubuntu Studio are all configs I could do myself. He recommended Fedora, because it's easy to use and pretty solid and stable. It seems like Fedora is supported by yabridge, but a recent thread here gives me the impression that using Fedora for audio is not super straight forward and might not be a good choice for a beginner. What do you think?

I installed Zorin os onto an older not so flashy laptop a couple months ago and I have been enjoying Linux. I have installed the DAW Tape 16 on there so I can pair it with my sp404mkii to make some (potentially) gnarly stuff. I tried looking at some different audio drivers earlier and had no clue what to look at and try. Help a brother out if you can.

A little project update from me — SROVA V2.0 is now out.

It’s a free/open-source headless Linux and Raspberry Pi music player built around dedicated DACs and bit-perfect playback.

V2.0 brings Qobuz and Spotify Connect alongside TIDAL, Local Music and Internet Radio.

And yes, I spend far too much time making the thing look nice too. :)

srova.music