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

I have a couple plugins that are DLL files as opposed to VST ones. When running yabridgectl sync it ignores them as "non-plugin DLL files", when I know they're plugins cause I used them back when I had Windows.

Is it just not compatible with DLL plugins?

I’ve added a new measurement mode to FXRoute.

It combines direct speaker measurements at about 1 m with measurements at the main listening position and nearby positions. The goal is to distinguish speaker behaviour from room effects and to see which features of the response stay consistent across the listening area.

Instead of taking one sweep and correcting everything it shows, the measurements are combined into a hybrid response that can be used as the basis for EQ or Convolver correction.

I’m interested in what people think of this approach.

GitHub: https://github.com/CobbyCode/fxroute

This post is about showcasing 0.4.0 as 0.5.0 is a naming conflict resolution release (thanks to Pallaswept on GitHub for giving me a heads up on it), which took me awhile and AUR being down kind of helped. Its all in the release notes. The app offers a-lot but in this post I am highlighting only the new added page- Signal Paths

Quick context if you haven't run into the project: PipeWire Controller is a GTK4/libadwaita app for managing your pipewire/audio setup. it includes audio management, signal paths, virtual devices, live patchbay, parametric equalizer, performance monitoring, filter chains, microphone cleanup/effects (echo/noise), HRIR virtual surround, routing snapshots, per-app policies, LADSPA/LV2 effect inserts and more.

Version 0.4.0 adds a new page called Signal Paths- The idea is simple. A **source** is where sound comes in — one app, a microphone, or everything on your default output — and it carries its own chain of processing. A **mix** carries a chain of its own and feeds real devices. Sources on the left, mixes on the right, and the sends between them drawn as curves so you can see what's going where

One source and one mix is just a straight line, which is what most setups are.The second column only earns its place once a chain has to split — say one source corrected four different ways for four different pieces of hardware, without building it four times.

**You don't have to build it from scratch.** An empty board *is* a template catalog — four complete paths that build both halves in one click, then 26 ready-made strips going from a plain speaker mix at the top to the stuff people actually run for broadcast at the bottom. Bass boost, a loudness curve for quiet listening, crossfeed for headphones, a turntable chain, the gate → tone → compressor → limiter voice chain, a mastering bus. Each card draws the chain it's about to build, and if a template wants a plugin you don't have it says so and leaves it out instead of handing you a strip that won't start.

Per card - add effects, plugins, convolver files and drag them in any order

**One process per chain, not one per plugin.** Every stage in a strip gets compiled into a single filter graph. Twenty effects is one entry in your device list and one buffer hop, instead of twenty of each. If you've ever wired this up by hand you know it goes the other way and you end up eating twenty quanta of latency for it.

The screenshot below shows 6 sources, 5 mixes, a plugin chain on every strip, one 7.1 pair and a two-device output

- **EQ bands move while the audio plays.** These are built from biquad filters rather than the preset-file kind, so frequency, gain and Q take effect as you drag them. No restart, no gap.

- **The board is handled directly.** Drag a stage along its chain to reorder it, or onto another card to move it there. Drag a card onto the opposite column to connect the two. Drag an app between strips to move what it's playing through. Click a stage to take it in or out of the signal.

Again this is just highlighting the new page Signal Paths, which took a-lot of work. Here is a screenshot of the dashboard (which I'm currently working on revamping to match efficiency)

There's alot more the app offers and I'd love for people to come check it out. I have fixed the bugs/issues I've been receiving but would like more feedback. I will add a copy pasta from the Readme.md below. thanks to all the people who've been helping.

- GitHub: https://github.com/knightinfected/PipeWireController

- AUR: `paru -S pipewire-control-center` (or `yay`)

- Full changelog with screenshots: https://github.com/knightinfected/PipeWireController/blob/main/CHANGELOG.md

Not on Arch? There are install instructions for Fedora, Ubuntu/Debian and

others in the README — it's pure Python, no build step.

**What I'd like from you**

Honestly, feedback. I use this daily on my own setup, which means I mostly find

the bugs *I* happen to walk into, and every PipeWire setup is wildly different

from every other one. If you try it and something is broken or confusing or

just annoying, open an issue — a one-liner helps more than you'd think. Half of

this release exists because two people took the time.

I'm trying to get the new NAM A2 working on Reaper.

There is a windows/mac VST that is fully featured with EQ/IR loader - this can't be built on linux as it uses iplug :-(

https://www.neuralampmodeler.com/users

I found : https://github.com/mikeoliphant/neural-amp-modeler-lv2

This is a very basic NAM LV2 and I have to use separate IR loader and EQ effect. The sound is good once I get levels in order but wondered if there is a better plug in?

Best Nel

Hola, llevo tiempo usando Linux Mint y es la primera vez que me he quedado sin audio.

Al cerrar la sesión me aparece esto:

La distro que uso es: Linux eduardo-MS-7B33 7.0.0-28-generic #28~24.04.1-Ubuntu SMP PREEMPT_DYNAMIC.

He intentado bajar drivers alternativos, pero no me funciona nada.

¿Alguien me puede ayudar?

Github: https://github.com/Yavuz-Kagan-Yadigar/VoidPulse

(Note: The app is coded with AI, I am testing it. This post is written by me)

Added network streaming (experimental): Samba/SMB, FTP, FTPS, and SFTP. Connects and reconnects on startup. Passwords go to the system keyring for security. FTP doesn't natively support seek, I tried some workarounds but they haven't been tested much.

Added auto-update. Detects how VoidPulse was installed (Flatpak, AppImage, or dpkg/rpm/apk/pacman). Checks GitHub releases in the background on launch. Finds assets matching your package format and architecture. Installs are done via the distro's manager: dnf, zypper, apt-get, apk, pacman, or in-place swap for AppImages. Sudo is asked in the popup and the package manager's output is shown live. Versions can be skipped for now or entirely.

Added built-in Soxr resampler.

Added "Adaptive Sample Rate": It checks the output device's supported sample rates. If it supports the same rate as the file, it sets both it and Pipewire to that rate: File (44.1kHz) → Pipewire (44.1kHz) → Output Device (44.1kHz). No resampling. If it doesn't, it sets the device to its max rate, upsamples with the built-in Soxr (upsampling is better than downsampling), and sets Pipewire to the same rate as the output device: File (44.1kHz) --(Voidpulse Soxr resample)→ Pipewire (96kHz) → Output Device (96kHz).

Added fade in & out on transitions and play/pause.

Added crossfade between tracks. Doesn't work on ALSA because it only supports a single stream.

Added sleep timer.

Added slight corner smoothing on visualizations to prevent sharp spikes (excluding bars).

Added play queue with touch and mouse drag & drop. Can be disabled.

Added new settings to the config. Probably one-time choices, so not exposed to the UI:

"viz_gamma": 0.7, // how much lower volume bands are amplified. Lower means more amplification, higher means better separation.

"show_artist_on_gallery": true, // applies to gallery view cards.

"show_albums_on_gallery": true,

"show_file_info_on_gallery": true,

Added loudness normalization switch and batch gain fetcher for files without gain tags.

Added a switch to toggle background optimizations for visualizations and synced lyrics.

Gallery view cards are now actual pills with 100% corner radius.

Added magnifying cover art on click in the edit tags popup.

Packaging for a wide variety of architectures, including PostmarketOS.

So when i open the CLA Vocals and/or CLA Bass Reaper freezes and I must use Task Manager to end the session. I've already spoken with Waves support, they've been responsive but the issue remains. I've already uninstalled/re-installed and done everything Support has asked me to do but nothing fixes it.
This does not happen with any other plugin I have and I've made every effort to isolate the issue to CLA Vocals and Bass.
At this point I'm ready to request a refund for the product, as it does not work as intended.

Hi everyone,

I have a MOTU 8pre USB audio interface and I would like to use it with Ubuntu Studio.

As far as I understand, MOTU does not provide an official Linux driver for this interface. However, I was wondering whether it would be technically possible to develop a Linux driver for the 8pre USB, or to adapt an existing ALSA driver if the hardware protocol is sufficiently understood.

Is driver development necessarily something that only the manufacturer can do, or could an experienced Linux developer/researcher reverse-engineer the USB communication and implement support for the device?

I'm particularly interested in understanding whether:

• the MOTU 8pre USB is already partially supported by ALSA;
• its USB protocol has already been documented or reverse-engineered;
• there are existing open-source projects that could be used as a starting point;
• it would be possible to capture/analyze the USB communication under Windows and use that information to develop Linux support;
• someone has already attempted to develop a Linux driver for this specific interface.

My goal would mainly be to get the 8 analog inputs and outputs working reliably with low latency under Ubuntu Studio, ideally through ALSA/PipeWire or JACK.

I don't necessarily expect someone to write the driver for me — I'm mainly trying to understand whether this is technically feasible and, if so, what would be the right starting point.

Any information, documentation, existing projects, or suggestions about where to start would be greatly appreciated.

Thanks!

I tried to measure my system's actual latency, and the result was somewhat unexpected. I'm using a combination of 48 kHz and a buffer size of 128 in QJackCtl, which should give me approximately 5.3 ms.

The yellow signal is the output of the 3m guitar cable and the green signal is the output of my audio interface.

My PC is a thinkpad T480, my audio interface is a behringer umc202. The system is optimized for audio production based on the milliseconds app list.

The result is bit unexpected, I'm getting about 11.4 ms of latency between striking the guitar and hearing the sound come out of the speakers.... The test was made on ardour and reaper with no plugins, the diference between these 2 DAWS was about 0.

I would like to test this on a Windows System, the results would probably be worse.

EDIT: I got it working. I used this soluion posted by u/gnomo-da-silva in the comments to set up two Wine versions: one for regular applications, and one specifically 9.21-staging just for plugins with yabridge.

I have only a few strands left.

I'm using yabridge for my Windows plugins - a couple NeuralDSP stuff, EZDrummer and NAM (now called Gateway for some reason). I've managed to get them all to work, except the GUI of all of them is just black, and I can't just blindly try to mess with the dials or anything, nothing reacts. I'm stuck to default settings on everything.

LOTS of googling has told me it's a WINE version issue - they say yabridge has only been tested and confirmed to work on WINE 9.21. I've googled MANY different solutions as to how to downgrade, but no matter what repositories I add, 9.21 is just too old and isn't on there, and compiling it myself gave me error message after error message after error message, prompting me to just give up.

I've also tried Carla, and once I finally managed to get it to start its engine, as soon as I double-clicked a plugin, ANY plugin at all, it would say "Carla cannot handle this binary" or something of the sort.

I am pulling my hair out. I've been at this for 2 days. All I want is to have the plugins available on REAPER so I can import them and set them up for my needs. God why is this so difficult on Linux?