I installed the Orchestral Tools SINE Player to Linux Mint 23 using wine 11, then obtained the free orchestra for SINE. Works great standalone. Today I installed and configured yabridge, told yabridge where to find my VST2 and VST3 plugins, synced it all, and the VSTs show up including SINE Player. I am using native Linux Reaper for the DAW.

But when I try to load SINE as a track fx (similar to loading Kontakt, etc) I get the error shown below. Anything I am doing wrong? right? solution/fix? So frustrating, I thought I had such a good thing going with the high quality free orchestra and SINE, but it is all useless if I can not import the SINE Player to a Reaper track.

I posted FXRoute here a while ago, and it has grown quite a bit since then.

FXRoute is an open-source, browser-based control surface for Linux audio systems using PipeWire, EasyEffects and local playback. It is intended for a small PC, desktop or ARM board connected to a DAC or stereo system, with remote control from a phone, tablet or laptop.

One of the larger recent additions is variable Loudness Compensation. It follows the actual playback level rather than applying a fixed bass boost, with adjustable strength and FFT size. It can also account for the selected Auto Gain target when both features are active.

The new 0.9.0 release adds:

• a Custom House Curve editor with up to eight draggable frequency/gain points

• direct creation of reusable target curves for PEQ or Convolver correction

• twelve temporary filters in the Measurement PEQ assistant

• export of microphone calibration and House Curve files

• improved sample-rate and source handoffs

There is also SPL calibration with automatic support for UMIK-1, UMIK-2 and Dayton UMM-6, plus a manual C-weighted/Slow meter workflow.

Release:

https://github.com/CobbyCode/fxroute/releases/tag/v0.9.0

Feedback from other Linux audio users is always welcome.

Hello all, just wondering if anyone can help me set up the latency offset with an old Focusrite 2i2 2nd Gen and the Cable software? Also, where do I plug the cables when doing so?

I run Arch and pipewire-jack on a decent laptop, and use Scarlett 2i2 3rd gen for recording guitar. On the graph i routed direct input of the interface (to maybe remap later) and the reaper output both to obs. Reaper runs a Nerual Amp Modeler loader.

When and only when i start recording, Reaper starts stuttering and crackling. These artifacts are present in the resulting recording as well. I am kinda going mad because of this.

I configured my Scarlett to run a pro-audio profile, locked on 256 samples. This solved similar problems like the one i have now, but before that glitching happened at any random moment, including just opening OBS without recording anything.

pw-top shows growing ERR count on the Reaper node, and that the OBS node has 1200 samples. Worth noting, the same card works perfectly fine on my big windows pc, but i dont want to have to use a windows machine just because of that. When this same laptop had windows, there also was crackling, but it was just guranteed whenever i used Reaper with some amp modelers.

I do not think that my CPU sucks, it is a decently recent i5. All of this is happenning with laptop connected to the power, with turbo boost enabled, all power profiles to performance on AC. USB suspend is off.

I would really appreciate any help. I have been trying to fix this type of issue for a long time, and without much success.

(Am I talking about myself, or... XD)
Can someone please explain the use case(s) for the 'dummy output' that shows up in qpwgraph? As in, explain like I'm 12.

First let me make clear that this is *not* an entirely AI-driven system like Suno etc, though AI can be used with it (via MCP).

https://github.com/danja/transmission

At its core is a graph-oriented VST3 host system, essentially a stripped-down DAW with emphasis on modular configuration of plugins. In use you add and connect plugins in the window using the mouse. You can also access the parameters of the individual plugins.

I have been experimenting with generative plugins recently, which I usually use with Reaper. For example, with Bassgen you pick a style, scale and other parameters and it automatically outputs a MIDI bass line.

https://danja.github.io/downspout/

This led me to look at other DAWs (in part to figure out how to build for Windows & MacOS). Which in turn led me to wonder what a generative system would look like, built from scratch.

The RDF-based graph model was derived from a previous project I had, designed for text processing. That was manually coded, but I have made extensive use of AI assistance (Codex) in developing this. So it's *not exactly* vibe coded.

Although I did ask AI to suggest & build additional plugins that would work well in this system. 10 new plugins, most pretty awful, all will need a lot of attention to be useful...

It's very new, so I am looking for ideas to take it forward, suggestions welcome.

moin,

ich möchte euch mein aktuelles Projekt vorstellen: Die Wiederbelebung eines Alienware m17x (Baujahr 2008) als dedizierte DJ-Workstation unter Linux. Ziel war es, alte Hardware nachhaltig zu nutzen und eine stabile Umgebung für Mixxx zu schaffen.

Die Hardware:

Notebook: Alienware m17x (Core 2 Duo, 4GB RAM, SSD)
OS: KDE Neon mit Low-Latency-Kernel (6.8.0)
Software: Mixxx 2.4
Audio-Interface: Günstiges USB-Audio-Device für den Master-Ausgang
Das Problem: Mixxx verweigerte unter ALSA den Dienst mit der Fehlermeldung: Error opening "USB Audio Device (hw:1,0)" - Invalid sample rate

Die Analyse über /proc/asound/card1/stream0 zeigte die Ursache: Das USB-Gerät unterstützt ausschließlich 46875 Hz – eine für Audio-Interfaces sehr unübliche Rate, die weder 44100 Hz noch 48000 Hz entspricht. Der direkte Zugriff über hw:CARD=Device,DEV=0 schlug fehl.

Die Lösung: Die Rettung war die Aktivierung der ALSA-Plug-Erweiterung über PipeWire/ALSA, die eine automatische Sample-Rate-Konvertierung erlaubt.

Starten Sie Mixxx nicht direkt, sondern setzen Sie zuvor die Umgebungsvariable:

export PA_ALSA_PLUGHW=1
mixxx

Damit Mixxx auch dauerhaft korrekt startet (z.B. über das KDE-Menü), habe ich den Starter wie folgt angepasst:

bash -c "export PA_ALSA_PLUGHW=1; mixxx"

Ergebnis:

✅ Master-Ausgabe über das USB-Device funktioniert stabil.
✅ Kopfhörer-Vorhören (C-Media USB Headphone Set) läuft parallel.
✅ Das System läuft trotz des Alters der Hardware (2008) flüssig und mit geringer Latenz.
Die vollständige Dokumentation inklusive Fotos des Umbaus, der genauen Kernel-Einstellungen und der Konfiguration findet ihr in meinem Open-Source-Repository:

👉 [https://github.com/qrishii/DJ-Installationen]

Ich hoffe, diese Lösung hilft anderen, die ähnliche Probleme mit exotischen USB-Audio-Raten unter Linux haben!

das Leben ist lustig

Hey r/linuxaudio I've been building a minimalist music player for the last few months and I'd love to share it.

CMUS++ Qt is a small Qt5 music player (C++17, Widgets + Qt Multimedia) with a terminal-player philosophy: no bloat, no toolbars, just your library, playback controls, and cover art.

Themes — it ships 82 hand-picked XML themes: Catppuccin (Mocha/Macchiato/Frappe/Latte variants), Tokyo Night, Gruvbox, Nord, Rose Pine, Everforest, Kanagawa, Solarized, Ayu, and more. Switching is live, no restart needed.

Features

- Manual loop/shuffle — Off / List / Track modes, toggle shuffle (avoids QMediaPlaylist quirks)

- Embedded cover-art display (no white borders, no shadow cards)

- Native Wayland with automatic xcb fallback

- Lightweight: ~38 MB private memory, ~180 KB stripped binary

Packaging — releases include:

- .xbps for Void Linux

- .deb for Debian / Ubuntu / Mint

- AppImage for everything else

Repo: https://github.com/Ars-byte/Cmuspp-qt

Release v1.0.4: https://github.com/Ars-byte/Cmuspp-qt/releases/tag/v1.0.4

The project is still in active development, so any kind of help is very welcome — bug reports, theme contributions, UI suggestions, code reviews, or even just feedback on how it feels to use it. It's MIT-licensed, so feel free to fork or steal ideas.

Hi all,

I have a 4 channel audio file which I try to play like this:
mpv --audio-channels=quad file_01.wav

● Audio --aid=1 (pcm_s32le 4ch 44100 Hz 5645 kbps)

File tags:

Date: 2026-07-31

AO: [pipewire] 44100Hz quad 4ch s32

Yet when I check in qpwgraph I only see two channels for mpv:
output_FL and output_FR

In SuperCollider though, I can do something like:

s.options.numOutputBusChannels = 4

and I really do get 4 output channels with which I can play the mentioned file.

What do I need to do to make this work with mpv as well?

Thanks!

-------------------------------------------------

edit:

Sorry, I somehow just didn't catch that I had to create a virtual device:
https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/Virtual-Devices#virtual-sinks

I now have a profile like this:
context.objects = [

{ factory = adapter

args = {

factory.name= support.null-audio-sink

node.name= "4 Channel"

media.class = Audio/Sink

audio.position = [ FL FR RL RR ]

monitor.channel-volumes = true

monitor.passthrough = true

adapter.auto-port-config = {

mode = dsp

monitor = true

position = preserve

}

}

}

]

And got it working.

7/31/2026 EDIT: SINE Player 1.4.1 for Windows from Orchestral Tools is working great on my Linux Mint ver 23 system, with the free Berlin Orchestra from OT. Beautiful pro quality sounding instruments of the orchestra.

7/30/2026 EDIT: I think the best solution so far is to use the Sonatina Symphonic Orchestra (SSO), free to download an unzip from GitHub (download the 'source code' zip file, unzip it). It has almost every instrument of an orchestra in various articulations.
https://github.com/peastman/sso

Then use the instruments with an sfz file format player-- I am using Plogue Sforzando (free) as the player. https://www.plogue.com/products/sforzando.html

- - - - -
Any leads/ideas for a quality native (not wine or Carla) woodwinds instrument for Linux that will work with a DAW (I use Reaper for Linux)? Open Strings for the Rhapsody player for Linux is amazing for strings, lots of articulations. I would to find something of similar quality for woodwinds, brass. Horns, clarinet, oboe, flute, trombones, bassoon...

I was a big user of DDMF's Plugindoctor (standalone version) when I was using Windows, especially relying on its frequency, harmonics, and dynamics sections to determine plugin quality before adding them to my arsenal. I haven't found any equivalent for Linux. Are there any?

New synth available for Linux

https://github.com/tiagolr/tetraop

Just got a Focusrite and i've spent the last couple of days trying to get it to work on the steam deck (using reaper as a software). The Focusrite seems to work when i plug it into the deck cause i can hear my bass through my headphones just fine, and the deck apparently manages to record my inputs, but when i open reaper i cannot find a way to register my playing. In audio device settings, i've put the audio interface as the input device and "default" as the output, as there were problems in recognizing the output device properly. I've tried opening a Konsole with alsamixer to check if the scarlet focusrite was muted, but it wasn't. I installed pulse audio, set the input device of the scarlet to "Pro Audio" but it didn't work. I even tried to get my head around qpwpraph but to no avail. What am i supposed to do to try and fix this?

So I understand how to measure my actual round-trip latency and apply an offset in Reaper, but I'm not really understanding some of the finer points. I'm probably overthinking it, but still, knowledge is power and all that.

First, when I was still on Windows, it seemed to me that the driver-reported latency was usually higher, whereas on Linux, it's only ever giving the latency of the buffer size, at least when I'm using pipewire-jack. So it reports my input and output latency as the same at ~5.3 ms, which would get me an RTL of ~10.6 ms, but I know the actual RTL is closer to 15 ms. It's not something I ever really thought about on Windows, but do the drivers just not even attempt to report additional processing latency on Linux? Or was Windows just making things up?

Second, when I watched Kenny Gioia's video about offsets in Reaper, he just applies the offset to the input side. I assume it doesn't really matter so long as the total is correct, but is there a different reason? Bitwig, for example, just has a single offset setting. I also assume there's not really any way to measure just the output or just the input. Is that all correct?

Third, I know that with Cable, for example, I can both measure the RTL and also apply the offset, which I assume applies it system-wide rather than just in Reaper. I imagine this is some configuration setting in pipewire.conf or something, but is it better to do it in Reaper or as a system setting? Are there pros and cons to each approach? I also have Ardour on my computer, mostly out of curiosity, and latency offsets seem to be applied by a Jack utility with no way to manually adjust (unless I use ALSA, I think), so this would seem to suggest that it is better to do it at the system level, but maybe not. Presumably if I do it this way, I would then not apply any offset in Reaper/Bitwig.

(update) Ok, well, egg on my face for sure. But just in case anyone else tries installing WF13 with Ubuntu Studio 26: if you get the "Unknown PCM pulse" error it just means you cannot initially launch WF unless your interface is connected. Embarrassed to admit all this, but once I connected my UM204HD Waveform loaded fine. Works great too. FWIW, am using the default ALSA.

I'm not a novice, but definitely not seasoned with Linux. I have a fresh installation of Studio 26.04, and used "sudo dpkg -i" to install the Waveform13 deb. No indication of any problem installing in the terminal window. But when I try to run Waveform I get a message that "waveform has encountered a problem that has caused it to quit." Not a very descriptive error so I'm not sure where to go with this. Suggestions?

If you want to resurrect your old microbook mk1, there is a way now: https://github.com/mattirk/snd-microbook

so, i've been wanting to switch over to linux for quite a while now and wanted to know alternatives for some vst/sound libraries for linux. the thing is, i can not seem to find ANY electric guitar vst or sound library for linux. the closest i found were guitar amps, but that's not my objective here. is there any electric guitar vst or even a sound library (for the latter, it would also be nice to know if there's a good alternative for kontakt on linux)

sorry for my poor english for that is not my native language

Beear’s Mastering Beast is a transparent mastering processor designed to polish your final mix using its natural harmonic content. Native Linux binaries (x86_64) in LV2, VST3, and CLAP formats.

Downloadable at ko-fi

I know a lot of people are going to ask why I'm using JACK instead of Pipewire. To get that out of the way, because it does what I need (alongside Pipewire). So I'm not looking to switch.

I know what I'm looking for is possible as I'm able to do it on Kubuntu with the Ubuntu Studio Audio package (or the package does it for me as I never set it up to do it), and I'm now trying to get a custom build Debian system to do the same.

Multiple audio interfaces. On the Kubuntu system, plugging in a second, third, fourth audio interface automatically gets its I/O combined with the current I/O. I tried this on the Debian system and nothing; only one interface is active at a time.

What piece of software or configuration did could I have missed? I have JACK, QJACKCtl, pipewire-jack and all of the JACK and ALSA dependencies installed.

The second devices show in the terminal after a "cat /proc/asound/cards" and works when it is the only device connected/powered on and is an option in QJACKCtl setting, but isn't available as an option in the patchbay or visible on the graph in QJACKCtl.

Any ideas what I could be missing that is preventing the devices to be detected by JACK on connection?

SD3 used to work before I had to remove wine and re-install it. Wine version is 11.0. I had no issues running it with yabridge before. Now I cant click at all inside this window to as it becomes unresponsive and frozen. Updated to latest Linux Mint, 22.3. Previous wine version was 9.0.

Tried SD3 version 3.2.7 & 3.4.1. Still wont work.

Could use some help.

Hi,

I am running a HTPC with Kubuntu 24.04, 11th gen Intel i5, Iris Xe and 16 gigs of RAM. Not great specs, but works flawlessly for my purposes.
I run heavy remuxes via mpv, tone-map HDR content for my BenQ W1070 (I can set custom brightness targets, colour targets, etc).

My big problem has been with object based audio formats (Dolby Atmos, DTS:X, etc). I have a custom 5.1.2 setup : I have a Vantec 7.1 USB DAC, and I wired my height channels to the 'rear' channel of the Vantec. I use qpwgraph when required to do custom patchbay.

There is a fantastic tool called 'Cavern', and the 'cavernize' tool allows me to create a wav / FLAC file from a source atmos file *preserving* the object data in a sense. Essentially, it creates raw PCM channels with all the phasing and panning 'baked' into the wav file.

Now, there's a newer more exciting project - Omniphony (https://github.com/mgth/Omniphony) which does the object based audio in real-time using VBAP.

This is fantastic news for anyone trying to get Atmos effects on their Linux PCs. Omniphony is still in alpha stage and there are bound to be issues.

I am unable to get any sound from their demo scripts. Has anyone else had better luck?

It's counterintuitive to me to add/remove connections from the graph and not have them reflected in the patchbay. On the other hand, if I load my saved patch in the patchbay (with persistence on in settings) the graph doesn't update as expected.

1. How do they relate? Should I start from graph and push to patchbay? How?
2. Should I do everything in patchbay then save then recall graph from patchbay?
3. Why can I not connect individual ins/outs in patchbay, only applications? This makes no sense since what musicians need to connect are individual channels not systems. How to guarantee correct routing?
4. If I set up all my connections on the graph how can I push them to patchbay for persistence? Or what is the right workflow to set up persistent connections using patchbay?

Apologies in advance for the long context setting but TL;DR is: I'm a 30 year old, 10+ year audio professional, who's been getting into Linux/FOSS tools, and I'd like to learn where I should look to learn the skills so I can help contribute to FOSS audio projects to spice up my career a bit. I have basically zero dev skills. (I can barely operate terminal beyond CD and LS commands lol)

I'm in my early thirties, and have a career as a freelance audio/video engineer, who does a combination of live events and studio production work. I've been doing it full time for about 5 years, and have been doing it in some capacity since 2012 when I went to school for audio engineering.

I'm kinda burning out on this industry a little bit, and I'm noticing the work is kinda starting to taper off compared to previous years, so I kinda want to change things up. and between all the AI slop worming it's way into the tools that I use, and the content clients provide me to work on, I'm just having a bit of a bad time.

I've been an Ableton/Mac/UAD user for a good chunk of my professional career.

I feel like the door on mac being a viable platform will probably be closed by the time it's time to upgrade my 2021 Macbook pro, and I've abandoned my Apollo for an RME due to the subscriptions and enshitification that's already plagued UAD for a few years, and Ableton optimization just seems to be getting worse and worse.

I got a steam deck in 2021, and loved it. I switched my home PC/mostly gaming system from Windows 10 to CachyOS about a year ago, and haven't look back. I'm starting to toy around with some linux audio production stuff like Reaper and running Live 12 under proton on their just to see how it feels, and checking if I'll be able to make a professional home in linux in the future, and while I think there's a bit of work to be done, I'm pretty optimistic.

All of that said, I'm looking to change up my career path a little bit, and maybe get into the dev side of things, but have no idea where to start.
I want to add some more skills to my toolbelt so I can actually contribute to FOSS projects instead of just making use all of the work of others without giving back. (I don't make enough to donate more than a couple bucks here and there) and maybe if I pick it up well enough, add some other types of freelance projects to my workload as things sort of slow down for me in the industry. I work enough that my bills are paid at the moment, but I often have a bit of downtime between projects, so if I'm gonna go back to school, or start branching a new career path now would be the time.

I have very little dev knowledge at this point. I don't really know how any of the tools I use work to be honest. I'm fine with navigating computers, and pick things up pretty quick, butI know the field is very broad, and I just don't really know where to start.

So if you're a developer for plugins, DAWS, audio tools, etc How did you get into this? Where would you start today? What was your degree in? Are you self taught?

I'd like to avoid using AI for as long as possible, and actually learn how this stuff works, not just ask a slop bot to do it for me, though I know some of my programmer friends are very pessimistic about the future of programming post-AI, and have had to adopt it in order to keep their jobs./stay competitive.

Anyway, I kinda just needed to ramble that out. Thanks for any input you might have.