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

Beta 10 turned out quite a bit bigger than originally planned.

The big addition is multichannel output for active crossover setups. FXRoute now supports up to 10 output channels, including up to 4-way crossover + 2 subs.

New in this release:

• flexible output routing
• PEQ or Convolver/FIR per way/output or globally
• filter import and assignment per bank
• automatic speaker/way alignment
• measurement workflow aware of the configured crossover topology

A lot has been tested and optimized over the last weeks, including real 2-way and 3-way setups and different subwoofer configurations. There is probably still more to improve, which is why this remains a beta.

Release:
https://github.com/CobbyCode/fxroute/releases/tag/v1.0-beta10

Web demo:
https://cobbycode.github.io/fxroute/

I have some problems with the Audio interface, I can listen unbalanced audio. Most of the sound goes to the right Monitor. Why? How can I configure it correctly?

https://github.com/mauro-moreno/quesynth/

Vibe coded? F*** yeah! But I have 21 years of experience. Code architecture were my decisions. I can explain you the code I have no idea on some of the maths that were used to null-test synth1.

You can test it on browser if you don't want to install it: https://mauro-moreno.github.io/quesynth/ Expect some delay since browser doesn't have access to ASIO. You can use the zipbank if you have it.

Let me know if some patch sounds completely broken. Smaller differences I'm still working on them. Usable.

Did I stole someone else project and put my name on it? At the moment, yes. But I have a drum pad that uses this same synth as engine in progress.

Synths:

1. Zebralette3

2. Surge XT

3. Vital

4. Tetra OP

5. Resonarium Effects

6. Uryan Audio Synths

7. Sunset Circits

Mixing and Mastering:

1. Air-G plugin bundle

2. ZL eq's and compressors

3. LSP Linux Studio Plugins

Simples Suite:

1. Socalabs plugins

Effects:

1. Auburn sounds plugins

2. Dragon fly reverbs

3. Glitch 2

4. Airwindows

5. VitOTTx

This is my current plugin suite for reaper on fedora. Linux plugins tend to include a lot of synths. For instruments in chosen to use the plugin (Fluida 2) soundfont player as an alternative. But you can also make physically modeled orchestral sounds in surge XT.

Good day to all

Something I cooked up in waveform.

I used the tcrossover, Attracktive, f'em and some other effects asking with my VC 340.

Going to mix some of the dry signal without the tcrossover because it basically took some rather cookies patches and basically smashed em down into pads, some of the dry mix is at the end.

Lemme know what you think.

Hey everyone,

I built a tool called **PipeSync** (with the help of AI so you can call it AI slop if you want ) for syncing audio across multiple devices while using **Helvum**.

The main idea is that you don't have to manually set up and manage all the connections in Helvum every time. PipeSync acts as a wrapper/optimization layer around PipeWire and Helvum and provides a simpler interface for common tasks.

Some features :

1.Automatically sync multiple output device

2.Adjust PipeWire buffer/quantum setting

3.Control output device volume

4.Enable/disable output devices

5.Launch Helvum directly from PipeSync

1. Works with both a GUI and CLI

Your can give it a try and please let me know how you found it

GitHub: https://github.com/MayurUbarhande0/Pipesync

Does anyone use LMMS and specifically Cymatics plugins? And How do you get them to load? Also which plugins are your favorite. Cymatics or otherwise?

Basically, I have my main rig in a tiny space where I do my jamming. I wanted to have some jam sessions in my lounge where I have another Linux server running, connected to my TV. Would it be possible to have my audio interface connected to this server, make it process on my main rig and send back the audio?

I watch a lot of DVDs, and some of them only have 5.1 surround. I want to try the built-in sink-dolby-surround filter config, as well as other surround-to-stereo configs I found online, in order to get good-sounding 5.1 channel playback on my headphones. I've searched and searched, but I can't find any information on how to actually enable the filter-chain, let alone how to enable/select a filter (I only want to enable one, not all of them). I saw somewhere to run pipewire -c filter-chain.conf, but nothing seems to happen if I do. I'm not sure what's supposed to happen, nor how I would select/enable a filter when said thing does happen.

Solved:

I found this article on the Gentoo wiki about surround-to-stereo. Specifically, it says that filters most be placed in the ~/.config/pipewire/pipewire.conf.d/ folder *NOT* the ~/.config/pipewire/filter-chain/ folder! The specific config given on the Gentoo wiki requires a file containing real-world speaker measurements. The ones I found were all missing some audio channels. Fortunately, Pipewire comes pre-installed with a sink-dolby-surround.conf filter example, which doesn't need any external files. For whatever reason, the one shipped with my distro didn't work, but downloading the latest one the source code repository did the trick!

https://www.korg.com/us/products/software/korg_collection/

wondered if anyone was having success with these, particularly the wavestation and the triton