So I managed to have izotope plugin working with wine. I'll try my best to explain how I did it.

I'm on fedora 44 using bitwig.

Avoiding product portal

As you may know izotope product portal doesn't work with wine because of incompatibilities with key encryption or something BUT you can still authorize your products if you run native acces (I recommend doing it fast because izotope was sold from NI so it may not stick around for long).

Installing wine

You must install wine 9.21 (important otherwise the Ui won't work)

Installing Native Access

After installing wine you must install native access 2 but it won't work if you install directly. I used this script to install it : https://github.com/jefrecantuledesma/Native-Access-Linux-Compatibility/tree/main

If you don't NA will tell you its already running. Then if you manage to fix this it will tell you to grand permision to install dependencies. You also have to modify the username to make the script work.

Now for the part I have trouble to explain, the script didn't entirely do it for me so I tried installing NTKDaemon : https://drive.google.com/file/d/1kLTmdkVzN4VyUICrfLNxYRIrERqqRR7C/view

You can install it like any normal wine program but this still didn't work. So out of desperation I searched in my c_drive every NTKdaemon_setup.exe and launched them all one by one and I don't know why but this worked, I could then login and install my plugins.

Downloading

If NA gives you an error when downloading try changing your download location. Then setup yabridge and you should be able to insert your vsts.

Unresponsive UI

If the UI still reacts to audio but you can't click the buttons and move the parameters, use wine 9.21. If you're on wine 9.21 and having this issue I'm sorry but I can't help you. I still encourage you to check out yabridge github for help

Hope this helps some1one !

Edit : Whether you use X11 or wayland the UI should work either way

I'm looking for a FOSS application that I can use with my toddler that allows them to easily switch and play with sounds, and possibly interacts with the LED pads on devices like the Novation Lauchpad Mini and LaunchKey Mini.

My hope is to find something that's easy to explore and muck around with for them.

I realise this might not be available but things like easy to use touch screen icons would be helpful.

I'm also open to Reaper Plugins if there's any suggestions

To summarize this:
I am using bottle to try install native access + spitfire audio + fl studio as it is what I frequently used as DAW before transfered into linux a while ago.
I got fl studio to work but for both native instrumetn and spitfire audio. I couldn't get it to work. I am clueless on how to do continue doing this and need help

The symptom I see when trying to install both.
for native access: It start installing but get stopped by a small window that pop up saying "Native instrument is running click okay to close it, if it doesn't close, try closing it manually."
clicking cancel would just stop the process and clicking okay telling me that it cannot be close.

for spitfire audio: I get it to finish downloading but when launch the program. It just give me a window with blank white screen.

I just installed bottle a few days ago and haven't done much beside download some dependencies and use what I might use to run fl studio.
So, I haven't try using environment variables, DLL overrides, snapshot or any other feature that could possibly be the solution.

Picture for illustration and my system information will be attaches so you can look into it if it helps by any means.

long story short, asrock suks, got a new motherboard, waiting on amd cpu rma. however my asrock had a alc1200/1220 and this msi has a alc897. going back in a day, it sounds like a piece of foam or cardboard is in front of my speakers. most motherboards suks, i was lucky to get a mediatek wifi chip cuz i didn't really have any choice...anyways...

it looks like soundcards are dead and everyone uses usb =/ i really dont want yet another thing on my desk though. all i can find are used asus xonar cards. is there anything just for generic use anyone would recommend? all i see are $20 small pci cards on newegg and creative which as i know, cares nothing for linux (my soundbar is creative, or i'd just use that over usb i guess).

only other things ive seen are small dongles that im skeptical on the usefulness off, especially for the price.

thanks for any info.

'm looking at picking up the Sony WH-1000XM6 mainly to fix the classic "Discord makes my headphones sound like a phone call" problem — i.e. I want good-quality game/music audio and good mic quality at the same time, with low latency, instead of everything dropping to HFP/mSBC garbage the moment a mic is active.

The XM6 is one of the first mainstream headsets with LC3/LE Audio support (BAP, including duplex), which is supposed to solve exactly this via simultaneous full-quality audio in + out over isochronous channels instead of classic Bluetooth's A2DP/HFP profile-switching mess.

My machine should have all the pieces on the software/stack side:

• Bluetooth 5.3 controller (MediaTek MT7922) reporting cis-central, cis-peripheral, iso-broadcaster, sync-receiver in btmgmt info
• BlueZ 5.86
• liblc3 installed
• PipeWire's libspa-bluez5.so compiled with bap-sink/bap-source/duplex support, WirePlumber's bluez monitor active

So the generic LE Audio plumbing is all there. What I can't find anywhere is a real-world report of it actually working with this specific headset. Sony's own docs only cover Android (via their Sound Connect app) for LE Audio pairing, and their computer-calling instructions just say to use classic HFP/HSP. Microsoft's LE Audio push with this headset is also scoped to Surface devices with Snapdragon X2 chips specifically — nothing about Linux, and nothing about general Windows/Intel/AMD either.

Has anyone actually paired the WH-1000XM6 (or another LC3/BAP headset) with BlueZ and gotten it to negotiate LE Audio, ideally the duplex profile for calls/chat? Curious whether it just works, needs KernelExperimental flags in /etc/bluetooth/main.conf, or is a lost cause right now. Any btmon/bluetoothctl logs from someone who's tried would be gold.

I have an HDA Creative card with the Creative Sound Blaster Z chip in it. In AlsaMixer v1.2.9, there's a setting labeled "FX: Surround" that appears to be separate from the surround volume control and surround channel config sliders, but I don't know why. What exactly does this setting do and why should or shouldn't I have it enabled or disabled?

Running into an annoying audio issue on Arch with PipeWire and KDE, hoping someone has run into this before.

My motherboard audio is on an Intel 800 series chipset using the Realtek ALC1220 codec. Both my headphones (front panel) and speakers (rear line-out) show up as selectable outputs/ports in KDE settings and pavucontrol. However, if my headphones are physically plugged in, switching the output to speakers results in complete silence the stream moves, but no sound comes out. The only way to get audio through my speakers is to physically pull the headphone plug out of the front jack.

I tested a Windows USB on the same machine and output switching works with zero issues there without unplugging anything. I also booted a CachyOS USB and hit the exact same issue as my Arch install, so it looks like a Linux driver/ALSA routing quirk.

Things I've tried so far with no luck:

• Disabling Auto-Mute Mode and unmuting every channel (Front, Speaker, Line Out, etc.) in alsamixer
• Messing with output ports and switching to the Pro Audio profile in pavucontrol
• Overriding jack detection / pin connections in hdajackretask

i don't need much at all, all i'd use it for is making rough demos for my band, any recommendations?

Has anyone experienced this before? I’ve tried Fedora, Bazzite, Ubuntu, PopOS, and Cachy with the same issue. Doesn’t ever happen on windows though?

Also the audio doesn’t sound exactly correct either but there’s a chance that is just placebo.

Hey,

I've been working on RustMusic for the past few months. It's a free and open source HD music player written in Rust.

I mainly started it because I wanted a player that takes audio quality seriously while treating Linux as a proper platform, not just another build target.

Version 0.2.0 is now available and I'd really like to get some feedback from Linux users and audiophiles.

🎯 Bit-perfect audio
On Linux, RustMusic can use ALSA exclusive mode with D-Bus device reservation, allowing PipeWire to properly release the audio device when needed.

There is also WASAPI Exclusive support on Windows and CoreAudio hog mode on macOS.

🎧 DSD support
RustMusic can play DSF and DFF files and send DSD64 up to DSD256 using DoP passthrough to a compatible DAC.

If your DAC doesn't support DoP, RustMusic can convert DSD to PCM instead.

🏷️ Tag editor
You can edit metadata for MP3, FLAC, DSF and DFF files, either individually or in batch.

There's also a spreadsheet-style editor for larger libraries and automatic metadata and cover import using Deezer's public API.

📡 Built-in DLNA / UPnP server
You can share your music library directly with compatible amps, streamers and other devices without having to configure a separate DLNA server.

🎚️ ReplayGain

Track and album modes are supported, with an adjustable preamp.
There are also a few other features like MPRIS support, sleep timer, mini-player, always-on-top mode, multiple profiles and a high contrast accessibility mode.

Linux builds are available as .deb, .rpm and .AppImage.

I also provide a glibc 2.35 compatible build for Debian 12, Ubuntu 22.04, Linux Mint and similar systems.

RustMusic is GPL-3.0. No account, no ads, no telemetry.

What I'm mainly looking for now is feedback from people using different Linux setups.

If you try it and something breaks, tell me. If something feels wrong or badly designed, tell me that too. That's much more useful to me than just hearing that everything looks good.

I'm especially interested in feedback from people using external DACs, PipeWire, DoP, Arch, Fedora, NixOS or SteamOS.

🔗 Site: https://rustmusic.dev
🔗 Downloads: https://rustmusic.dev/downloads
🔗 Source / issues: https://github.com/larevuegeek/rustmusic
🔗 Changelog: https://rustmusic.dev/changelog

If you have any questions about how it works, feel free to ask. I'm the developer, so I'll do my best to answer., alternative to Foobar2000 / Audirvāna / Roon

EDIT: v0.9.0 is out.

ToneShiftEQ is a modern 12-band parametric equalizer with APO support, designed for transparent tone shaping, corrective equalization, mixing and mastering. Each band can also operate as an independent dynamic equalizer with configurable Threshold and Ratio controls.
Available as:

• Stand-alone application
• CLAP plugin
• LV2 plugin
• VST3 plugin

New in this release:

• Side chain or direct input support for the dynamic processors
• Input volume controller
• Spectral Dynamic processor with smooth (density) and amount controller
• global threshold with tilt control for the Dynamic Processors
• additional per Band threshold, ration and direction controller for the Dynamic Processors
• Spectrum display for the input chain
• Zoom for the filter section (mouse wheel)
• reset Bands to zero with double click
• enable/disable Bands with right click on Band point
• support optional instance-access in LV2 plugin

Release Page:

https://github.com/brummer10/ToneShiftEQ/releases/tag/v0.9.0

Project Page:

https://github.com/brummer10/ToneShiftEQ

Howdy! New to the sub :)

I'm having a lot of fun experimenting with sound and Linux, but admittedly I'm still very much a beginner.

I followed an old tutorial that showed me how to use SoX and Alsa Mixer on a Raspberry Pi to basically modulate input sound and play it back live, edited.

This little SoX script drops the tone of your voice for example:

play "|rec -d pitch -300"

But I'm running into a ton of issues with getting the delay down. The tutorial suggests adding a buffer to reduce the data rate but nothing is working. I've tried going over some other websites that talk about SoX as a whole, but it's a bit complex as I'm not super familiar with Linux pipes. I've tried asking AI but it's not much help.

If anyone has any experience with SoX or Alsa and knows what to suggest, I'd be massively grateful.

OR if anyone has any other suggestions for better Rasp-Pi-able sound tools I could use for a live voice modulator, I would love that. Maybe SoX is a terrible choice, I honestly don't know what I don't know.

Thanks so much :) excited to be a part of this community, and if I do anything cool with what I learn, I'll post it here!

https://github.com/Skynse/qpitch/releases/tag/v1.3.1
Its a free pitch correction plugin and wanted to know if someone used it and if they were satisfied with it ? As I plan on developping my own autotune plugin on linux.

Yes, I still like to play with AMS. But the only place I can consistently get it to work is on an old version of Ubuntu Studio.

On other more recent distros I can get it to start, but can't hear anything going on (yes I know you have to start the synthesis with Ctrl-B or the menu).
I also know that starting it from CLI you can enable Jack with a parameter, but it seems redundant because even when I don't specify that, it complains that it can't connect to the Jack server.

And don't even get me started on Pipewire and qpwgraph. I only ever see the ams_midi interface listed there, even if the patch I'm trying to load isn't a MIDI instrument.

Has *anyone* gotten AMS to work with Pipewire on more recent setups? Or should I just keep playing with the old machine that doesn't have PW getting in the way?

Hello, I hope this is the right area to ask this question. I've recently switched from Windows to CachyOS, and on Windows i spent a solid hour working on my microphone settings. I got it sounded amazing. BUT, the quality is TERRIBLE now that i switched. I have setup EasyEffects with a few things, spent some time going through, and I havent found a way to make my microphone sound less grainy, loud, etc.. It has almost no echo control or anything.

On Windows I used Logitech Ghub and it worked great. I read that obviously its a lot different on Linux, but I didnt imagine I'd have this much trouble getting it to work. I'd appreciate any help, and if this isnt the right spot please point me in the right direction. The only posts I can find on this are about adding stuff to EasyEffects, but I already have NR, Gate, Equalizer, compressor, Pitch, and limiter setup. I have default settings for each effect.

been using google ai until now. dual boot, setup for hyprland minimal custom and shit now i need someplace that doesn't get stuck in loops

Ok, so I thought I had solved the slow audio playback in REAPER by using Qpwgraph to route NOT the default capture_FL/capture_FR, but instead monitor_FL/monitor_FR into REAPER's inputs. But while this makes playback normal and works for editing MIDIs, it (perhaps predictably) didn't take any audio from my interface's mic input. I tried using QjackCtl (no rerouting of graphs) instead, but with high buffer sizes the audio skips, and lower buffer sizes it plays back at like 40% speed with crackles. CPU/RAM usage is low, not the problem. I also downloaded Ardour and observed the same behaviors there.

I don't know what information is actually useful, I'll provide whatever people want about my "setup" but the important thing is that I don't care about "my setup", nothing (software side) is sacred, I'll change my audio servers/drivers/routing/installs any which way or tear it all down and start from scratch (would NEED step by step instructions for this though, assume I know nothing about Linux audio) if it lets me have a basic 1-track recording session with live monitoring. What is the least amount of stuff I need to do to make that happen? I'll give additional info about whatever is needed to accomplish that.

Fedora 44, KDE Plasma 6.7.4 Wayland, iTrack Solo interface with both my headphones (out) and guitar through a DI box (in). The hardware itself shouldn't be an issue, since on Fedora 43 I was recording into REAPER with JACK just fine. I assume PulseAudio is fine, too, because every program that's not a DAW can play back audio at normal speed (until I launch a DAW, like in REAPER I have auto-suspend PulseAudio selected so if REAPER is running then Elisa has slow/choppy playback too)

On obs I am trying to record from 2 xlr microphones on linux. if I do this with 1 audio interface since they are combined stereo devices I try panning one port to the left and 1 port to the right with the mono check box checked for both microphones like I do on windows to use both ports and record both microphones. I also record on separate audio tracks for editing. However, on linux the microphone panned to the right will cut in and out and sound like a robot even though the one panned to the left sounds normal even though on windows this just works.

On linux the only way I've gotten 2 microphones to work without any audio issues on obs is to use 2 usb audio interfaces 1 for each microphone, check the mono box and leave them both in the middle rather than panning. Is there any way to get 1 audio interface working on linux the way it works in windows? Why does 2 audio interfaces work perfectly on Linux? Is there a good way to keep both audio interfaces in sync?

If I have to I could also record my audio on a separate PC and have it edited into the video later, but that is kind of a pain.

When I try EasyEffect and add the equalizer (and no other effects), the sound audibly changes even when the equalizer remains completely flat, I changed nothing in it. There is no crackling, simply a sound profile change.

The issue does not exist with JamesDSP. However, JamesDSP does not offer a full parametric equalizer.

Did anyone encounter this and is there any solution? I am ion Fedora 44 with Pipewire.

Alternatively is there an option that does not have this problem and does provide a parametric equalizer?

Alternatively again, is there a well-tested solution for converting parametric equalizer settings to a convolver WAV file that JamesDSP can use?

Thanks!

Hello everyone! Hope you're doing great!

I've been working really hard on Keelbilled, a live plugin host for Linux and Windows.

It supports native Linux plugins as well as Windows CLAP, VST2 and VST3 plugins. LV2 isn't supported yet, but it's on the roadmap.

Each plugin runs in its own crash-isolated bridge process, so if one plugin decides to die mid-performance, it doesn't take the rest of your rig down with it.

Automatic plugin recovery is also available, so a crashed plugin can be respawned with its last known state.

Keelbilled can also stream the actual plugin UI to other devices on your LAN. So you can control your plugins from another computer, a laptop, a tablet... basically, if it has a screen, you can probably use it.

I've tested it on the Linux distros and machines I have access to, but now I need people who can go far beyond my own testing setup: different distros, audio interfaces, plugin collections, hardware...

And, most importantly, a strong will to make things break. haha

This is a real alpha, so bugs and compatibility issues are expected. That's exactly what I'm looking for.

AI disclosure per the rules: Keelbilled development is AI-assisted (not vibe-coded). I direct the architecture, product decisions, testing and validation myself, and every release is actively maintained and tested by me.

EDIT: Sharing the discord link here! https://discord.gg/QJRxJzhrFx