I recently decided to try upgrading from Focusrite to RME and uh... even though people say RME works well on Linux, it is nowhere near as easy to get set up. There are a few utilities out there to help, but they are mostly abandoned, missing features, or just don't quite work right. Fortunately, alsamixer has most of what you need. It's not going to get you the full TotalMix setup, but here's how to make some sense of the 8 billion options. Some of them are still a mystery to me, so maybe someone can chime in so we can make a more complete guide.

First, run alsamixer from your terminal. Then you'll hit F6 to select the device and then select your Babyface. We're actually going to start with the capture section because there's a lot less to see. The main things to look at here are the "Main-Out ADATx" and "Main-Out ASx" options. These control the incoming levels of your ADAT inputs, with 50 being the unity value (as far as I can tell). I assume it has something to do with doubling as S/PDIF, but AS1 and AS2 are ADAT1 and ADAT2, with the rest being more obviously named. I'm not really sure what the other things on this page are for, and I have not so far found them to be necessary, but please chime in if you know.

Now the playback section, which is massive. But it all follows the same basic pattern, which seems to be Source-Destination. So, for example, "PCM-ADAT3-ADAT3" is trying to say that this is how much of software ADAT 3 should go to the physical ADAT 3 output. Every output has its own mix, so when you select ADAT 3 as your output in your DAW, it's more of a virtual output that can actually go to any physical output you want. So, for example, if I just want it to work "normally," which is to say that when I send something to ADAT 3 in my DAW it should just go to ADAT 3, then I would set "PCM-ADAT3-ADAT3" to 50. If I wanted to route ADAT 3 in my DAW to the physical ADAT 7 instead, I would set "PCM-ADAT3-ADAT7" to 50. Make sense? So here, "PCM" is just saying that this is a "virtual" output.

Here we will also see the "AS" channels as before, but also "AN" and "PH." "AN" is for "analog" and "PH" is for "phones," or analog outputs 3/4 (hence why they are PH3 and PH4 instead of PH1 and PH2). This is good to know because mixing here is much more flexible than just using the output copying feature on the device. So you can have the same thing going to your main outs and your headphones, but not also going to ADAT or S/PDIF. Setting both "PCM-AN1-AN1" and "PCM-AN1-PH3" to 50 will get you the same signal to the left channel of both your monitors and your headphones, for example.

Next there are a bunch of "Line-" channels, which control how much you will be monitoring the incoming signals on each of the outputs. For example, "Line-ADAT8-AN2" would control how much of your incoming ADAT 8 signal you will hear coming out of your physical output 2 (typically the right channel of your monitors).

These are followed by the gain and sensitivity settings for your two line inputs. I'm not really sure why gain is here when it is also in the capture section? Or sensitivity, for that matter, but that isn't in the capture section. Same for mic gain, which comes later on. But hey, what do I know?

Then a bunch of "Mic-" channels that follow the same format as the "Line-" channels. After that, you can enable/disable the pad and 48v for your mic inputs, in this case by pressing "M" on your keyboard or just clicking on the button instead of using your arrow keys. There's also mic gain again, but neither this one nor the one in capture have any correlation to what you set on the device, so that's confusing. Then you get a few S/PDIF options, most notably turning it on or off (to use ADAT instead).

Finally, you a "Main-Out" for each of your outputs. These are the master fader for each of the mixes/outputs, so you will want to set all of these to 50 (unity) except for AN1/2 and PH3/4. If you don't have these turned up, then it doesn't matter what you did with the rest of the channels. None of the GUI options I've seen give you control over these, so I just couldn't use my ADAT until I figured this out... AN1/2 and PH3/4 you'll want to leave at 0; these seem to act more like a boost, so you'll be in for a rude surprise next time you reboot your PC if you turn them up. The last option is to toggle the clock between internal and autosync which is something I couldn't even do on the Babyface in CC mode (the instructions in the manual didn't work), so it's nice to have it here.

Until someone decides to make a proper TotalMix clone (or RME ports it over, yeah right), this does at least give you access to the ability to mix however you want. Hope this helps! If anyone has corrections, please do let me know. And a pro tip, you can use page up/down to increase or decrease the faders by 5, or you can just press a number to set it to that value (5 sets it to 50, for example).

Hear where tracks compete. Make room for both.

Kick and bass fill the same low end. A vocal gets buried when the guitars come in. A synth turns sharp as the filter opens. Ressup-EQ helps you shape the tone, control the moments that jump out, and see how two tracks fit together.

Use EQ, Dynamics and Res Sup across up to 24 bands. Bring another track into the analyzer, see where their frequencies overlap, and adjust either instance's EQ from the same window while listening to the mix.

Try latest version 1.26.2 · macOS Apple Silicon: VST3 / AU · Windows x64: VST3 · Linux x86-64: CLAP

100% FREE no limitation, no renewal, totally FREE Github link -> Download Ressup-EQ

Enjoy the freedom of shaping sound

Hello r/linuxaudio !

I've been working on GlorpDrums, an open-source drum instrument as a standalone application and VST3/LV2 plugin. The goal is to build a free, open source alternative to commercial drum software like EZDrummer 3 (i know i'm not on that level yet lol) particularly for people who want to perform or record alongside an electronic drumkit through MIDI.

Project: https://github.com/asuka-glorp/GlorpDrums

I'm sharing here because I'd love to get feedback from people with real experience developing or using Linux audio software.

I would love to get DAW users who can test it in their actual setups and report compatibility issues, crashes, audio glitches, or other problems.

And also electronic drummers who can connect a MIDI drum kit and test how well the instrument responds to real playing, including velocity sensitivity, pad mapping, and hi-hat behavior where supported.

A bit of personal context: I had to sell my own electronic drum kit earlier this year because of health complications. Drumming is still a personal passion, so I decided to go ahead developing this project. I can't personally test it against physical drum hardware right now, which is one of the biggest gaps I'd like to address with the community.

It's still very much early days for this project so i'm not pretending it's perfect, but i think it's solid enough (hopefully) to start seeing the light of day.

I'm not deep enough into audio in order to be able to fully build effects like these myself out of the plugins that exist in EasyEffects. However, no other option seems to work nowadays, and most people, as far as I can tell, use EasyEffects for much more productive purposes like tweaking their audio quality to their preferences or producing music, rather than silly things like voice changing for role playing games and such.

I tried clownfish and the blasted thing doesn't want to pipe correctly, and has too few options for my needs, lyrebird has been abandoned, and there seem to be few other options for voice modulation that aren't just "Do it yourself with Pipewire coward".

Hi everybody, as DAWs created using AI spring up like mushrooms, here's another one.

It runs on Linux, macOS and Windows but is Linux first. It's free and will always be free.
You can look at the README here for its features and other screenshots:

https://github.com/BurningTreeC/faderframe/blob/main/README.md

You can download a release from here, you don't have to install it, it's also portable:

https://github.com/BurningTreeC/faderframe/releases

At first startup it starts with a demo session loaded to showcase some features. After creating a new project, it starts up with the last project open.

I spent a lot of time to get things right and planned this carefully, but some Users reported issues on systems without explicit realtime privileges. See the Troubleshooting section of the README (at the very bottom) for details.

I hope somebody likes this, I do and I'm using it successfully.

So, cheers from Italy,
-BTC

Out of the box the left channel is 100% and right channel is 20% -- so annoying.

With the AI knowledge the solution was to open alsamixer, select the audio interface. There speaker front and speaker rear were 46, 46 for both. The fix was to set it 100, 100 for speaker front. For speaker rear have it 0, 0.

I don't know why it is like that. Writing it here in case anyone has the same problem. I was already thinking of sending the unit back.

And in the end store the setting with the command:
sudo alsactl store

perhaps i can give a little help for the sound engineers who are a Linux user. i can port some of the software i made from windows.

DISCLAIMER: No personal or financial affiliation with EMR MUSIC GROUP.

I have been using Tape 16 (virtual 16 track reel to reel tape recorder) for a while now. It works on Linux (as well as Windows, Mac and iOS). It seems the passion project of an Aussie guy who was sick of the over produced, under ”performed” music that tends to happen with the infinite safety net of modern DAW-based music. So he built this. And now, he has released Tape 24 (you guessed it, 24 tracks…) which is immediately available for Linux the same time it was released on other platforms. It now includes customisable channel strips, some onboard fx, including of course, tape-based fx along with the ability to use your own plugins. Because us Linux users are often starved for choice, audio wise, I just feel this guy should get some worthy recognition on this platform. Recommended.

Clara on AUR
Github Releases page: https://github.com/acropolis914/Clara/releases

Just posting again cause the last release was broken. Sorry haha
and oh, next release is themability haha. this is in nord cause why not

Seeing all that AI generated software and plugins here in this sub, does it even make sense to still develop professionally plugins to sell them?

(obviously since the tools are now here, part of it will be vibe-coded, or AI assisted, though in a responsible way, I do not mean just posting the plugin online, not having it tested and asking for "beta testers") I fear there will be many cheap knockoffs idk?

To clarify I am someone that develops them though I wonder whether it makes sense to continue

Hi everybody.

I play the guitar and I want to record some guitar videos for Instagram.

My actual setup for recording is this: guitar --> focusrite 4i4 --> PC (Fedora) --> MODdesktop --> OBS Studio.

Then I edit the video with Kdenlive where I simply add a limiter of 0,84 (for the peaks) and a Normalizer (2 steps) to -16 LUFS.

Thats it, no compressor or other things.

The problem is that when I watch the video from the computer sounds great, but when I transfer the video to my smartphone and in Instagram the volume is very low if I compare it with the normal videos of other guitarists.

And if I try to increase the volume with an app (like VN-Video Editor) it raise distortion.

Can you suggest something? What can I do?

Thank you very much!

Gio.

Disclosure: this was written with an AI coding assistant (Claude, via Claude Code). I described the problem, made the design decisions, and tested it on my own hardware; the AI wrote most of the code and docs. Please review it like any other script before running it; it's small, plain bash + Python, and needs no root.

PipeWire runs most outputs at 48 kHz by default, and settings like pw-metadata clock.force-rate are lost on every restart. I wanted my USB DAC (Fosi SK02) to keep its rate across reboots and replugs, per device, without root.

dac-rate writes a WirePlumber 0.5 rule (audio.allowed-rates) for the chosen card: - Lock: one allowed rate, so everything is resampled once by PipeWire - Auto: all rates the card supports, and the global allowed-rates list is merged for you, so the device follows the content - Rules match the whole card by default, so they survive profile changes; per-USB-port matching for identical DACs, or per-output - Optional resample.quality per device - Reads supported rates from the USB descriptor, or asks the driver via aplay - status warns about other configs or a runtime force-rate that override it

There's a PySide6 window + tray icon (live hardware rate from /proc/asound), and a CLI with --json. Everything lives in ~/.config.

How it was tested: on my machine (Arch + Plasma 6, PipeWire 1.6, WirePlumber 0.5.18, Fosi SK02), including real reboots and power-cycling the DAC, plus ShellCheck and automated app tests in CI.

Limits, honestly: it's not bit-perfect when locked (PipeWire resamples), Auto is only bit-perfect with a single stream at 100% volume, and Bluetooth isn't covered. Only one DAC and one distro tested so far.

https://github.com/vcostin/dac-rate

Bug reports from other distros and DACs are very welcome.

If you didn't catch it in the news, a coder has used AI tools to easily have AI create almost identical software for Linux, Mac, Windows, for all the Adobe products as well as MS Word, MS Excel, Powerpoint, etc. The products are clustered under a platform called ArtCraft; free, no registration required, open source; in alpha currently, maybe 50-70% there but updating daily squashing bugs, allegedly will be in beta within a few months.

I was wondering, could the same process be used to have AI create composing tools and instruments for Linux? Maybe not sampled instruments, but perhaps sophisticated synthesizers, or possibly synth-based emulations of sampled instrument for at least rough composing? Just a thought.

my roblox mic (sober) for some reason does not detect the audio input

Enlivener listens to the guitar and provides a live accompaniment for it with a synthetic bass, keys, and optionally a user-provided drum track.

The app can be heard in this performance: The Two Step Circle as well as most other videos on that channel.

Enlivener uses pulse audio and runs in Linux. It is a time-domain harmonic tracker and synthesizer. It doesn't need to be online and it works with whatever overdrive or wah or reverb or whatever else you care to put through it. All the videos on that channel above were recorded with Enlivener on this machine:

CPU: dual core Intel Core i7-6600U (-MT MCP-) speed/min/max: 509/400/3400 MHz

Kernel: 6.8.0-138-generic x86_64 Up: 7d 21h 6m Mem: 4555.9/7797.3 MiB (58.4%)

The repo is here: https://github.com/CynthiaChristie/Enlivener

Oh yeah, FOSS of course, CC BY-NC-SA to be specific.

Backstory

Working with mixing and mastering and ambient music for a couple of years now. Always used Windows but always wanted to use Linux without it taking up too much time from actually making music, hence why I felt it was time, with help of AI as well.

In one day I, with the help of Claude, moved my entire music studio from Windows to Linux, paid plugins and licenses included, and it runs like it did on Windows if not better. If you've been waiting for "Linux is ready for music," this is my report: for me, it is.

What I run now

- CachyOS (GNOME) on an external Samsung T7 SSD that boots on any PC basically (have to deactivate and activate plugin licenses tho, and make sure Secure Boot is off on the PC I boot from)

- Reaper natively, with all my settings, templates, shortcuts and scripts moved over from Windows

- Behringer UMC1820 through PipeWire, my MIDI keyboard over ALSA

- About 100 Windows plugins working inside Reaper

What works, licenses included

- Slate VSX 7 with HD-Linear, licensed via iLok machine activation: no dongle

- Soundtoys 5 and Eventide Blackhole, also iLok

- iZotope Ozone 12, RX 12, Neutron 5, Nectar 4 through Native Instruments' Native Access 2

- FabFilter, Voxengo SPAN, TBProAudio ISOL8, Mastering the Mix LEVELS and REFERENCE

- Waves: Waves Central installs everything and moved my licenses over

- Speedrum 2, has a native Linux version

My biggest surprise: iLok, the copy protection most people name as the reason they can't switch, worked.

How

The core is Wine, which runs Windows programs on Linux (it translates, it doesn't emulate), plus yabridge, which lets a Linux DAW load Windows plugins. Plugin audio runs at native speed, and I can't tell the difference in my sessions.

The trick that made everything click was three separate Wine setups, each on the Wine version its plugins like best:

1. iLok plugins (VSX, Soundtoys, Eventide) on Wine 9.21
2. Native Instruments, iZotope, FabFilter and the rest on Wine 11 Staging
3. Waves on its own, so Waves Central can't interfere with anything else

A tiny wrapper script tells Reaper which Wine to use for which plugin, so it all just loads.

The things I couldn't find documented anywhere, in case they save you a day:

- Slate VSX: Slate Audio Center registers VSX but installs nothing under Wine, because its install script fails and then deletes itself. I grabbed the unpacked files mid-install and copied them into place by hand. After that, VSX worked perfectly.

- iLok: the official installer fails, but a manual install (github.com/shibco/ableton-linux/issues/235) works. Start Wine as "wine", not "wine64", or the license service never starts.

- Mastering the Mix: LEVELS and REFERENCE need two DLLs in Wine's system32: TurboActivate.dll (from MTM's installer) and libiomp5md.dll (Intel's free OpenMP package).

- Soundtoys: presets saved on Windows refuse to load, with a "Version field error". Presets saved on Linux are fine, so I swapped my templates over to fresh ones bulk wise via script.

Native Access 2 runs thanks to the github.com/selimbucher/native-instruments project, and Waves Central needed real PowerShell installed in its Wine setup.

The extras that made it feel like home

- VSX on everything: a PipeWire virtual output routes Spotify, YouTube and games through VSX hosted in Carla, and it mutes itself automatically while Reaper plays, so I never hear VSX twice.

- Quiet fans: my Windows FanControl curves rebuilt in CoolerControl, with case fans following whichever of CPU or GPU is hotter. GPU fans stay off at idle.

- My projects cleaned up: a script converted all 2,825 Kilohearts Gain plugins across 111 projects to Reaper's own JSFX, each at its exact original setting.

- Notes and files: Obsidian syncs to my NAS with Syncthing over Tailscale, and my music library with Nextcloud.

- Backups: a daily config backup, plus a full image of the whole system that restores onto any new drive, so I'd only have to reactivate licenses.

What didn't work

Being honest: the Kilohearts installer needs a login that failed under Wine, and XLN's and Mastering the Mix's own installer apps never show a window. REFERENCE 3 is missing one DLL. And Wallpaper Engine's scene wallpapers can't be the GNOME background (KDE Plasma can show them). Still got LEVELS and REFERENCE working though since I had the apps downloaded. Kilohearts did not work as it needs something else from the installer, very anti-linux friendly sadly, intentionally or non-intentionally.

Was it worth it?

Yes. No Windows updates restarting my PC mid-session, no Defender scanning my sample folders, and a system that does exactly what I set it up to do. The setup took a day, done together with an AI assistant (Claude Code) that did the digging and scripting while I handled the logins and licenses myself. Super simple, and I wanna be transparent about the AI use.

TL;DR

Linux is a valid replacement even using professional Windows n Mac plugins under iLok now.

If you're on the fence: back up, try Linux on an external SSD like I did, and keep Windows as a fallback. You might not need it.

Just thought I'd share the day's findings since it would have gone even faster had I had an easy way to confirm it worked. I had a day over and some tokens to burn.

Probably takes more time than worth it doing it all by hand, but yeah, I'm not a dev either and I just started learning the terminal, although I've set up a Raspberry Pi NAS as well and some stuff, so I'm familiar with the terminal. Tried Ubuntu Studio before but I absolutely hated it. CachyOS feels nice. Will try KDE Plasma tmrw cuz I like Wallpaper Engine.

Note: I just let AI summarize most of this message as it's very late now, so some of it isn't fully relevant to audio, just other programs like Obsidian that I use for notes, calendar and field recordings via Syncthing, so it's all synced between my phone and PCs and my NAS for backup. But most of this I wrote myself.

Hope this is helpful for someone. If not, well, at least I tried to provide value to someone. A lot of this was news to me.

Happy to answer questions.

Other than just unplug it, I mean. I posted last week asking how people were able to run such low buffers and went through a bunch of troubleshooting to see if there was something more I could do, but today I decided to see if it was a conflict with another USB device and... it is. Even though my interface is not on the same bus as my capture card (Elgato 4K X) or my webcam, at least according to usb-devices, simply disconnecting the capture card allows me to go all the way down to 64 samples system-wide without issue, or at least without any immediately detectable issues. With the capture card plugged in, I can't go lower than 256 or I just get constant xruns.

For now, I've just unplugged the capture card to keep testing at 128, but how can I fix this while still being able to use the capture card? It has to be connected to the one USB port as it needs a USB 3.2 gen 2 port and back when I was on Windows, the USB-C to C cable it came with was the one that worked the most reliably. But that port isn't even on the same bus and the bus the interface is connected to has all kinds of things connected to it: keyboards, a mouse, a 2.4 GHZ receiver, whatever btusb is (bluetooth controller? It's built into the motherboard, though), Stream Deck... For reference, it's this Gigabyte B660I board. Strangely, those two ports at the top aren't on the same bus; the USB-C is what I am using for the capture card and the interface was connected to the other port next to it, but the issues are the same if I connect it to the blue ones below that or even to the bank farther down. I know that the capture card also has issues where its own audio is garbled unless I specifically turn on the console, connect its outputs to another output to listen to it, disconnect those, then reconnect them again, so maybe there's some sort of sync issue? I'm a bit baffled.

Edit: I decided to actually test that last point and yup, that fixes it, too, but it's a big hassle to have to do it every time. So basically, audio coming from the capture card (taken over HDMI from the connected console) is garbled, which I'm now assuming is because of xruns, regardless of my choice of buffer size until I perform specific steps. Additionally, I get continuous xruns from my interface if I run the buffer at 128 or lower while the capture card is connected unless I perform those same steps. I had that posted as a separate issue in this post, but basically, I have to very specifically first turn on the connected console, then connect the capture card's audio to an output pair on my interface, then disconnect those, then reconnect them again (I use a pw-link script I made to simplify that). Because the console has to be on first for that to work, I can't just run the script at boot. This feels like a clocking/sync issue, but I don't know how to fix it, I've tried everything.

Edit 2: Not a fix, but as a workaround, I found that I can just disable the audio device for the capture card entirely using a Wireplumber config and then I don't have issues anymore. Since I was already having issues, I'm running the audio in through the interface anyway, so this works fine for me.

I could not get myself to throw away a NIO 2|4. Today, a combination of Claude and Gemini prompting and some Wireshark sniffing has resulted in a driver for ALSA. It works on input and output for a 6.18 kernel. I hope there are some more people who still have these interfaces to test the implementation with. In the future, other kernel versions would also need to be supported.

The Makefile is setup for NixOS so maybe you will need to change it for different distros.
https://gist.github.com/Zurga/9fb7354d5e1d878c4d4a87fef618d78e

SoundCraft is a free, open-source DAW built in Rust with Pro Tools-level features, native CLAP/VST3 support, and 9.1.6 surround sound—no subscription https://guitarbomb.com/blog/free-rust-daw-disrupting-music-production/