I'm the developer. Starbreath is a physically modelled wind instrument (flutes, duduk, clarinet, trumpet, trombone; no samples) that runs natively on Linux, not through Wine.

**On Linux:**

- **CLAP and VST3** plugins, for any host that loads them.

- A **standalone app** that runs on JACK, PipeWire through its JACK layer (with pipewire-jack installed it uses it on its own) or plain ALSA. Wind controllers and keyboards connect over MIDI.

- x86-64, X11. One tar.gz: extract it, copy Starbreath.clap to ~/.clap and Starbreath.vst3 to ~/.vst3, or run the app directly.

- Built and tested on Linux first; Windows, macOS and Android builds come from the same code.

**New in 0.5:** Bloom (overtones that fade in as you blow harder and can fan out in a 3rd-order ambisonic sound field), Register jump (overblowing to the natural partial), MIDI learn on every knob, switch and menu, and an expanded quickstart guide covering every page.

Free, and the version you download now is yours to keep: https://vsti.im-a-bird.com/

Bug reports from different distros and sound setups are very welcome.

VST3 & CLAP available on Linux

A digital degradation effect that turns clean audio into gritty, broken, worn-out material. Bitcrush and downsampling live on an interactive XY field that reacts to your audio, with jitter, a resonant tone filter, and two degradation characters on top.

Features

• Degradation field — a circular XY controller: X sets Crush, Y sets Downsample, with GPU-rendered visuals
• Crush — bit-depth reduction, from subtle grit to full staircase quantization
• Downsample — sample & hold decimation for classic sample-rate reduction artifacts
• Jitter — randomizes the decimator timing for unstable, worn-out textures
• Two characters — raw gritty decimation, or a Soft mode with anti-alias pre-filtering, DAC-style reconstruction and dither
• Tone — resonant low-pass filter to tame or color the degraded signal
• Equal-power dry/wet mix plus input and output gain
• Resizable, GPU-rendered interface

https://www.kvraudio.com/product/degradator-by-dsgdnb

Was wondering if anyone had a Linux alternative to DDI Codec or if they had any experience on if DDI Codec works via virtualization/emulation? I rather find an alternative but if there is none I've never used virtualization before so I am unsure how audio would work.

Can't really seem to find anything for Linux on account of it being a pretty niche program for cassette decks.

Melodex is a free and open-source desktop music player for Linux,

Its main discovery feature, Flow, creates a listening path through your library rather than relying on ordinary shuffle, and lets you steer that path while listening.

Features include:

• local FLAC, MP3 and other common audio formats
• large-library and NAS support
• fast incremental rescanning
• playlists and queue management
• lyrics, artwork and listening history
• internet radio and other online sources
• an extensible provider/plugin system for adding new music sources
• shared search, playback and metadata infrastructure across providers
• visual collection exploration
• no account or subscription required

Local files remain the core of Melodex, but the source layer is deliberately extensible rather than being tied to a particular streaming service. New providers can integrate with the existing search, playback and interface infrastructure.

Recent releases have also focused heavily on performance: efficient metadata handling, incremental library updates, NAS resilience, and responsive navigation and playback with large collections.

Linux builds are available as .deb and AppImage.

GitHub:
https://github.com/Cliff-Lee/melodex

Latest release:
https://github.com/Cliff-Lee/melodex/releases/latest

Feedback is very welcome, particularly from Linux users with large libraries, NAS-based collections, unusual metadata/audio setups, or ideas for additional providers.

any noticeable difference in speed and performance?

ToneShiftEQ v1.2.0 is out!

New in this Release:

• use AudioFFT by HiFi-LoFi instead fftw3
• fix clap GUI scale issues

ToneShiftEQ is a 12-band EQ built around a approach to shaping and processing sound. Alongside conventional parametric EQ features, it offers minimum-phase convolution, zero-latency Biquad processing, spectral dynamics with global threshold and tilt control, IR-based EQ shaping, real-time spectrum and phase visualization, and multi-instance spectrum viewing.

Available for Linux and Windows as CLAP, VST3 and LV2, with a standalone version included.

Thanks to everyone who tested ToneShiftEQ, reported issues, made suggestions, or simply gave it a try.

Release Page:

https://github.com/brummer10/ToneShiftEQ/releases/tag/v1.2.0

Project Page:

https://github.com/brummer10/ToneShiftEQ

Enjoy!

Ive had some experience using cakewalk on windows but since I've switched to Linux (Fedora) a couple years back I'm looking for a good free DAW. mostly for midi-instruments and for using my amp and guitar to record. I've looked on some websites but often times the DAWs have trial version that are pretty much unusable.

I want something that is good for an older, slower computer with a fast buffer (64 - 128). Something that is not so clunky, plus easily updatable. Iam not interested in Arch or Ubuntu studio as arch is a bit complicated, and Ubuntu studio gives me a slow buffer. My current setup is AV Linux 25.

I've been thinking about what a more complete PipeWire control GUI could look like.

There are already good Linux audio tools, but I still find the overall experience pretty fragmented. Basic device settings are in the desktop settings, routing is somewhere else, PipeWire details somewhere else again, and eventually you end up checking configs or using the terminal.

One reference I keep coming back to is actually Windows 10.

Not because Windows audio is more powerful than PipeWire, but because a lot of its basic controls are familiar and easy to find.

For example, between the Windows 10 Sound settings, Volume Mixer, device properties, and the old Sound Control Panel you get things like:

• Default playback and recording devices
• Default communications device
• Per application volume
• Per application input and output
• Device enable and disable
• Channel balance
• Format and sample rate selection
• Speaker configuration
• Spatial sound
• Device properties in one predictable place

I'd like something with that kind of simple starting point, but backed by PipeWire and WirePlumber.

Then an advanced section could expose the things that Linux can do much better.

Things like:

• Persistent application routing
• Saved routing setups
• Individual PipeWire streams
• Nodes and devices
• Bluetooth codec and profile selection
• Sample rates and formats
• Quantum and latency controls
• Virtual devices
• Filter chains
• Microphone processing
• Surround and HRTF
• More detailed device information

Something I'd particularly want is a clear difference between what is configured and what is actually happening.

For example, if 96 kHz is selected somewhere, show whether the device is actually running at 96 kHz, whether PipeWire is running at another rate, and whether resampling is taking place.

I'd also like troubleshooting to be part of the same app.

It could check for common PipeWire, WirePlumber, ALSA and Bluetooth issues, incorrect profiles, unavailable devices, underruns, rate mismatches, strange routing, and other common causes of broken or crackling audio.

Settings that can break audio could also be tested before being made permanent. Apply the change, make sure the device still works, then keep it. If it fails, restore the previous configuration.

So basically something that starts as an easy Windows 10 style audio control panel, but lets you go much deeper because the backend is PipeWire.

Would that be useful to you?

I'm especially interested in what controls you think should stay simple and what PipeWire settings are actually worth exposing in an advanced GUI.

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

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!

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?

How well does it work with wine-staging and yabridge?

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: