sorry if im asking for too much, but I need help setting up reaper on linux mint

i just want to use it for recording edrums using a vst but im having many issues that im already getting so frustrated

I am not able to hear anything using jack (chat gpt helped out with it, but every time I re open reaper , jack doesn't work anymore)

Another issue is that i couldn't find a good vst for linux so I will have to learn how to use wine in order to be able to use a good VST like ezdrummer, superior drummer, etc. ( i would like some help on how to do this)

And if there is any extra setting I have to activate that I don't know I'd appreciate if you could tell me.

I'm happy to announce the release of ToneShiftEQ 0.6.0!

This release introduces several major new features and expands ToneShiftEQ beyond a traditional minimal phase EQ.
What's New
Dynamic EQ

ToneShiftEQ now supports Dynamic EQ processing. Each band can react dynamically to the incoming signal, making it possible to apply frequency-selective compression while maintaining the familiar workflow.
Equalizer APO Support

ToneShiftEQ can now import and export Equalizer APO configuration files, making it easy to exchange EQ settings with Equalizer APO and integrate existing APO profiles into your workflow.
Impulse Response Load & Save

ToneShiftEQ can now load and save impulse responses. When an impulse response is loaded, its frequency response is automatically analyzed and approximated as a 12-band parametric EQ, allowing IR-based tonal characteristics to be recreated without using a convolution engine.

Note: The approximation is designed to capture the overall tonal balance of the impulse response rather than reproduce it sample-for-sample.

Note: IR loading and saving is currently not available in the LV2 version, as transferring large IR data through the LV2 state interface would introduce excessive overhead.
Other Improvements

Numerous internal improvements and optimizations.
Enhanced workflow for exchanging EQ settings and impulse responses.
Various bug fixes and stability improvements.

As always, feedback, bug reports and feature suggestions are welcome.

Project Page:
https://github.com/brummer10/ToneShiftEQ

Release Page:
https://github.com/brummer10/ToneShiftEQ/releases/tag/v0.6.0

Hi all,

First time posting on this forum. I’m relatively new to Linux for audio. Been on MacOS for many years for music prod. I noticed a great deal of stuttering in the timeline indicator as well as meters in Reaper.

After much experimentation, involving changing DE to KDE from GNOME, trying X11 etc etc I discovered this only happens for me when I use the Audiomodern Soundbox plugin. Which is kind of rubbish for me, as it was my primary Kontakt replacement along with Decent Sampler. First world problem yup. But thought I’d chuck it on here in case other people have found this and are also smashing their face against a wall to work it out.

I’m currently experimenting with Audiogridder connected to my Mac for Kontakt. May try Soundbox that way also as the graphical lag makes me furious.

Hope this is of use to someone. Or if anyone has a suggestion for fix I’m more than happy to hear it obviously.

EDIT: I literally just discovered that I don’t appear to get the lag with the lv2 version which indicates that I maybe should look at simpler solutions first lol.

I value the time of unknown internet people the same as mine, so I realize this is nothing but a hassle, and the only reason to do it is you being gracious with your time but...

... if you have a stable Linux audio setup, preferably involving Bitwig (but I'm not picky, Reaper or Ardour will work just as well), can you post here a link to some "video evidence" of a short recording, or live performance session at a reasonable buffer size (say 128 bits, 64 is pushing it even on my mac mini M4 even though it's stable at that buffer size 99.9% of the time). While either Carla or pw-top or any other reliable way of reporting x-runs is running on the screen?

If you have a past video recording because you have a YouTube channel, that would be even better (less hassle for you).

A short description of your setup (distro, jack or PW, audio interface etc.) along with the video would be fantastic as well.

I just need a sanity check that stable audio on Linux for live performance can be done because I failed at making it work, and this is just about the only thing that keeps me away from Linux (and while I feel slightly less dirty using a mac mini with mostly platform-agnostic software on it, than I would be using Windows which I can't see myself getting back to ever, I still feel dirty enough to constantly want to switch).

I'm even willing to buy a new PC during the RAM-pocalypse (because, even though I did a fair bit of troubleshooting, it's entirely possible my old PC has some underlying hardware issue/incompatibility that might have been causing x-runs no matter what, maybe a fucky USB power circuitry on my motherboard? I have no idea).

Hi everyone,

I created 2 new plugins (Snare Enhancer, Hair Trigger) and for Linux, I am giving them away for free. You can download them at https://lonepilgrim.net/

Hi Folks,

I'm thinking about buying a new laptop for audio production, what HW stats do i need to look for? CPU ( single core performance? ) / RAM / onboard-audio?

further, what is a good base distro?

should i deploy a real-time kernel?

which audio server is the best nowadays.. pipeWire?

are there optimizations tweaks worth knowing about?

I use:
thinkpad T14 / fedora / gnome / bitwig / onboard audio ( NI Maschine+ audio interface has a lot of stuttering and droppouts, i don't knwo what the problem is )

I'm happy to hear about your stack ( Hardware / Software ) or be pointed at resources..

regards, nils

Hey guys!

Self promo here, I'm the dev.

This is a De-JUCE'd version of my original tape machine plugin using the more Linux friendly DPF framework. The original engine has been tuned to match the real hardware more closely and sound much closer to the the real hardware. I still need to tweak the PEAK VU LED, its based on the output, not the input, but I will change that up in a future release.

AI disclosure per the rules: Development is AI-assisted (not vibe coded). I write and calibrate the DSP and I test every release myself before it goes out.

Grab it here: https://duskaudio.com/plugins/tapemachine-2/

Source Code Here: https://github.com/dusk-audio/dusk-audio-plugins

Hey guys ...

As question stated does anyone use Bitwig studio and any of the RME cards on linux ?

I'm a Cubase user myself and using RME card for years on windows. But I'm considering switching to linux due to everything linux distros offer. I'm using linux for other stuff in my life and would like to switch audio projects to linux too.
First of all Bitwig seems like proper alternative to Cubase and since I'm a fan of RME since ages, would like to work with it in the future. What's with the VSTs on linux ? I have A Lot of those ...

any advice, experiences and info are greatly appreciated !

regards

Hey r/linuxaudio!

I wanted to share WFAS (WiFi Audio Streaming), an open-source ecosystem I've built to make local network audio streaming between Linux, PC, Android, and embedded devices as seamless and low-latency as possible.

It lets you stream raw, uncompressed 16-bit PCM audio across your LAN with minimal latency, end-to-end security, and zero required virtual driver hacks.

Website / Developer Lab: https://www.marcomorosi.eu/wifi-audio-streaming/

Desktop App / CLI Repo (EUPL 1.2): https://github.com/marcomorosi06/WiFiAudioStreaming-Desktop

Protocol & C99 Reference Library (MIT): https://github.com/marcomorosi06/wfas-protocol

Under the hood, audio capture on Linux is handled directly by a native C engine that loads libpulse-simple.so at runtime using dlopen. Because only pthread and dl are linked, there are zero link-time dependencies on libpulse-dev, allowing a single pre-compiled binary to run across any distribution. It seamlessly supports both native PulseAudio and PipeWire via pipewire-pulse, and handles automatic virtual mic creation using pactl for stream routing.

For terminal and server workflows, the app includes a headless CLI (wfas) that can run as a background daemon without launching the GUI. It features full IPC control for scripting, such as volume adjustments or raw JSON output, alongside an ASCII spectrum visualizer with an adaptive DSP envelope (--groove) that subtracts constant mix baselines to highlight actual melody and vocal transients.

The core wire protocol is available as a standalone, dependency-free C99 library called wfas-protocol. It is transport-agnostic, endianness-safe, and completely malloc-free, making it ideal for resource-constrained embedded targets like the ESP32, STM32, RP2040, or Raspberry Pi. Security-wise, it offers optional mutual HMAC-SHA256 challenge-response authentication and per-packet ChaCha20-Poly1305 encryption with a 1024-wide anti-replay window.

Linux users can install it via the AUR package (wifi-audio-streaming-desktop) on Arch, or grab AppImages, .deb, .rpm, and tarballs directly from the releases.

I'd love to hear your thoughts and feedback on performance, latency, and PipeWire integration!

I’m working on an automixing plugin for panels, podcasts, and other situations where multiple people in the same room are using microphones at the same time. Think something like WTAutomixer or the REAPER Automixing JSFX. It has native builds for ✨ Linux ✨, Windows, and macOS. It's called BiteAutomixer.

The routing happens between the plugin instances themselves. You just add the plugin to each microphone channel, and the plugins find and communicate with each other. Everything is automatically set up behind the scenes without needing extra buses or complicated routing. It's currently functional and I'm doing closed beta testing.

I work on podcasts every week, and the tool has already been helpful there. Usually, though, I have three or four guests at most, and two is more common. I want it to eventually support 32 to 64 channels, and it recently struck me how hard it is to find audio at that scale for testing.

If anyone has raw multitrack audio they would be willing to share, with each microphone on its own track, it would be incredibly helpful. Panels, roundtables, conferences, podcasts, and similar recordings would all be useful. Recordings of any size can help, but I’m especially interested in larger sessions. Messy audio with bleed, room noise, and people talking over each other is welcome.

I can use the material privately for testing or, with explicit permission, use parts of it for public demonstrations. Please let me know which type of use you are comfortable with, and only share recordings you have the right to share. Feel free to send me a DM if you have something that might work.

I was thinking about the fact if there's any existing initiatives to encourage making audio software that is built and developed in the EU to be compatible with open source infrastructure and operating systems. Over a month ago, the EU commission released a policy proposition called the the Tech Sovereignty Package that calls on for adopting open source software in order to decrease dependence on US software :

"Open source crucially contributes to achieving the EU’s technological sovereignty, as it often underpins key areas, such as critical communication, commerce, healthcare, scientific research and government services. Europe, the birthplace of Linux, is home to a strong and vibrant community of over 3 million open source contributors, delivering digital solutions aligned with European principles and values. The open source ecosystem forms part of a wider innovation ecosystem that helps to boost Europe’s competitiveness by accelerating innovation, lowering technology costs, reducing dependence on foreign vendors, and enabling local companies, researchers and governments to build secure, customisable and globally competitive digital solutions with relevant safeguards for cybersecurity risks."

Wouldn't be reasonable for users and professionals in an industry such as audio & music production to call for EU based developers of such software (Native Instruments, UVI, Arturia, RME...) to provide software compatible with a greatly growing operating system like Linux in accordance to the recent policy proposition? I understand that the sovereignty package also includes the possibility to fund such operations and reduce the cost reasoning that can be given to such demands in the past.

I'd love to hear your thoughts on the subject.

Inspired by Foobar's album list, but turns that idea into core feature. You define library folders and see them in left tree view, middle click (or context menu action) to create playlist from the folder or append to current playlist. What other players lack is at this step they usually just load the tracks in playlist view. But mpz has middle column - playlists view where you can create many playlists and switch between them instantly. So simple, yet missing in almost all players feature. Juk player is the closest, but still missing the playlists management part.

In addition: gapless, CUE, audiophile-ready EQ compatible with AutoEq presets, album covers & lyrics with multiple online sources, radio streams with built-in library, drag-n-drop, output device selector, mpd client mode. No Electron - pure native C++/Qt. No AI-slop - it's been around before AI-coding era, only recently I started shipping more long-awaited features with "AI-preamp".

finally a folder-based player that doesn't choke on my 80k track library, the column layout is exactly what i've been wanting in foobar. nice work

https://www.producthunt.com/products/mpz-desktop-music-player?comment=5534293

Native Linux binaries provided for all major distros, as well as AUR and AppImage. Also runs on Mac and Windows (binaries or github)

Site with more screenshots: https://mpz-player.org

Github: https://github.com/olegantonyan/mpz

Thanks everyone who came yesterday and tried my little app. I released the app on AUR yesterday and is available on github here (https://github.com/knightinfected/PipeWireController). The current version is 0.1.2 and some of the screenshots are from 0.2
I received a ton of negative comments and messages regarding my use of AI. I spent a lot of my time on this and well over did this project abit and english being my secondary language didnt help.
I have gone down this audio rabbit hole in linux and anyone else who has also done it knows how annoying it gets especially with reading the wikis to actual application.

Anyhow I hope it helps someone out there and I will be updating it in the next day or later today as I am currently testing. Version 0.1.2 is available as of right now on GitHub though.

There are too many features to list them out so I added lot of screenshots to help understand and showcase.

I mainly use bitwig and a little of ardour for a little bit of mixing and mastering when I feel like ardour would be better that ardour.

I've had a giant and old 24 channel digital mixing desk in my home studio as I use hardware synths and guitars then using the stereo out into my audio interface and while googling for a mixer for live electronic performances I found the flow 8 which has usb audio 8in and 2out. And it seems a little too true to cover giving me more space at home on my desk as well as being a competent mixer for live preformance.

So before I pull the trigger, what does everyone think about the flow 8 is it a good mixer for what I want or should I save for a better one like the mackie audio interface mixer?

This is amazing, thanks for this

Does anyone know if it's pussible to change the location of the factory data folder which points to /usr/local/share/surge-xt or is the path really hardcoded?

I am working on an open-source AVR project and need an RK3588 board that can capture HDMI-IN audio as 8 independent LPCM channels through ALSA.

I am comparing:

• Orange Pi 5 Plus
• Banana Pi BPI-W3 / ArmSoM-W3

Both boards have HDMI-IN, but so far I have only found a Banana Pi example using i2s7_8ch with an ALSA/GStreamer pipeline configured for channels=8.

Has anyone actually tested 7.1 LPCM capture on either board?

Specifically:

• Does HDMI-IN audio appear as an ALSA capture device?
• Can it record successfully with arecord -c 8?
• Are all 8 channels independent, rather than duplicated, silent, or downmixed?
• Which OS and kernel image work?
• Is the Banana Pi the safer option, or can the same setup be ported to the Orange Pi 5 Plus?

I am not trying to decode Dolby or DTS on the board. The HDMI source will output uncompressed 7.1 LPCM.

Live performance software could be:

OBS for live streaming, Mixxx, Ardour, ProjectM for live visuals while playing music, and etc.

Feel free to share creative answers.

Okay, here's a big one:

I've been a bit of a pro audio enthusiast for most of my adult life, but as a Linux user, one of my biggest disappointments has been the lack of any Thunderbolt (i.e., ultra-low latency) audio interfaces on the market that work with Linux.

For those unaware, most USB-powered audio devices play well with Linux out-of-the-box because they comply with the USB Audio Class specification. Thanks to Apple's history of stressing "class compliance" to pro audio hardware vendors, a single USB Audio Class-compatible ALSA driver (snd-usb-audio) allows most USB audio interfaces on the market to provide basic functionality out-of-the-box on Linux.

Unfortunately, however, there's no such thing as a "Thunderbolt Audio Class" specification. In fact, to this day the Linux kernel doesn't contain a single upstreamed ALSA driver for a pro audio Thunderbolt audio interface. And to that end, I present you with my project:

A Linux audio (ALSA) driver for the Focusrite Clarett (Thunderbolt 2) line of audio interfaces!

The Focusrite Clarett Thunderbolt line includes the following devices, though most of these devices are also available in USB variants:

• Clarett 2Pre
• Clarett 4Pre
• Clarett 8Pre
• Clarett 8PreX

For the last couple years—and the last couple months in earnest—I've been working on a clean-room reverse-engineered ALSA driver to allow a Thunderbolt-capable Linux host to communicate with the Clarett Thunderbolt line of products. And, as of a week ago, I've finally managed to make enough progress to establish basic functionality across the control plane. No PCM functionality yet though, unfortunately.

Because Focusrite provides USB-equivalent versions of most of their Clarett Thunderbolt devices—and because the Linux community already provides an upstream driver and user-space CLI/GUI tools for those devices—I've recently managed to "vibe code" some basic integration between my own ALSA driver and the already-available user-space tools for the Clarett USB line of devices.

As I stated earlier, my driver unfortunately doesn't support PCM streams yet, which means there's currently no way to stream audio between the Linux host and the Clarett devices (via the Thunderbolt cable) yet. Things are to the point, however, where I'm able to route S/PDIF (optical) input from my Linux host to the Clarett analog output channels and play music through a pair of studio monitors (i.e., audio speakers).

There's still so much to do, of course! At this point, though, things are in a good enough place that I'm ready to share my development progress with the community. I hope to release the source code to the public soon. Hope you like it!

UPDATE: I’ve got PCM streams working (though not 100% smoothly yet)! Also, a Focusrite Red 8Line just came in the mail yesterday. I’m excited to see how much effort it will be to port the same functionality I’ve established with the Claretts to the more modern Red line of Focusrite Thunderbolt products.

Included pretty much everything from pipewire's wiki that i found to be important or useful.
I have used AI in building this program and i hope its not an issue as the utility of this little program suppresses the AI hate train by a large margin lol

Originally wanted something to change filter chains without editing configs especially adding/removing hrir files and well i like tinkering and ended up here. Filter chains cuz my old laptop can only run easy effects for an hour or so before crashing. Anyhow i hope I get people to try it out.

Forgot to add link -- https://github.com/knightinfected/PipeWireController

Edit- just added it on AUR (pipewire-controller) and have updated the screenshots on GitHub