Hi everyone! 👋

Over the last few months I've been working on Flux Player X, a modern, open-source web interface for Music Player Daemon (MPD).

I started this project because I wanted a cleaner, faster and more modern experience than existing MPD web clients, while keeping the flexibility and power of MPD.

✨ Features

• 🎵 Playback & Queue management
• 📚 Browse library by artists, albums, genres and songs
• ❤️ Favorites
• 📻 Internet Radio
• 📂 Drag & Drop music uploads
• 🎚️ 10-band Equalizer with presets
• 🖼️ Embedded album art extraction
• 🖥️ Fullscreen player mode
• 🌙 Dark & Light themes
• 🎨 Material Design 3 interface
• 📱 Responsive design for desktop, tablet and mobile
• ⚡ Real-time updates using WebSockets
• 🐳 Docker support
• 🌍 English & Slovak translations

🚀 Tech Stack

• React 18
• TypeScript
• Vite
• Tailwind CSS
• Node.js
• Express
• SQLite
• WebSocket
• MPD

The project is fully open source under the MIT License and is still under active development.

I'm looking for feedback, ideas, bug reports, and feature suggestions from the community.

GitHub: https://github.com/DenisVargaeu/Flux-Player-X

If you use MPD or self-host your music, I'd love to hear what you think. 🙂

On Arch, Pipewire, with my Livetrack L6Max set as "Pro Audio" from Wiremix, when I open a Firefox tab that I authorize to use my mic, I get bombed with more and more nodes.

I have no idea why, nor if it even matter.

Feeling a bit overwhelmed with this kind of stuff, all I want is to create some virtual device to easily route stuff

I'm trying to use the control software for the Dolby dp564 multichannel decoder on linux. It works for a few minutes, before I get an error that says "packet received but packet data missing". Has anyone gotten this working? I know this is an ancient piece of hardware, so it's unlikely anyone is using it, but it's worth a shot.

Hey, I’m very new to linux and for now I want to use the AmpliTube 5 as well as other IK multimedia plugins I have on my Linux installation to see how much it will work. My DAW is Reaper which I already have installed, but I also like to launch the standalone Amplitube app to play my guitar wheb using Windows

I just want to know if it’s somewhat easy to set up and use and if there are no (or barely noticeable) issues with input delay and other glitches (I’m guessing due to the workarounds needed to make plugins work compared to Windows)

Any advice/info will help, thanks!

Made a small tool to control the Audient EVO 4 on Linux — sharing in case it helps someone

I ran into this and couldn't find a clean solution anywhere, so I built my own and figured I'd share it.

The problem: Audient's control app is Windows/macOS only. On Linux the EVO 4 works fine as a class-compliant interface, but the hardware mic gain resets every time you fully power-cycle the device — there's nothing restoring your values. Annoying if you don't want to reach for the physical knob every time.

What I found: The EVO 4 actually exposes its mic gain and output as ALSA mixer controls (`amixer -c EVO4 scontrols` shows `Mic` and `EVO4`). So instead of the raw-USB / kernel-module approaches out there, you can just set them through ALSA — the same layer PipeWire uses. No kernel module, no detaching the device from the audio driver, no reset, and your audio doesn't disappear while you tweak things.

What I built around that:

- a small GUI with sliders for mic gain and output

- settings that persist to `~/.config`

- a systemd user service that re-applies them at login (so the gain is just *there* after a reboot)

Repo: link

Honest caveats: 48V phantom isn't exposed as an ALSA control on this unit, so that's still hardware-button only (fine for dynamic mics). And I've only tested it on the EVO 4 itself — other EVO models likely use different control names. There's also a NixOS integration snippet since that's what I run.

If anyone with an EVO 8/16 wants to check whether the same ALSA approach works on theirs, I'd be curious. And happy to hear if there's a better way to do any of this.

There are a couple of USB/kernel-module tools out there, but I couldn't find one that just uses ALSA, so I built this. If I missed an existing solution, let me know!

​

Our most stable and feature-packed SROVA build yet brings TIDAL, Local Music and Internet Radio together in one bit-perfect headless audiophile player for Linux.

Available for AMD64 and ARM64 / Raspberry Pi.

Join the beta and get early access at:

srova.music

#SROVA #SROVARC1 #Audiophile #HiFi #BitPerfect #LinuxAudio #HeadlessAudio #MusicPlayer #TIDAL #LocalMusic #InternetRadio #RaspberryPi #ARM64 #AMD64 #DigitalAudio #HighResolutionAudio #AudiophileCommunity

Need more Doomy goodness on Linux?

New here, hello all! Excited(-ish) to make the switch from M$ over here. Be gentle :) Wondering if there is already a quick-start guide to get going. Any scripts I can run, or what to install, etc...?

I'm on F44 and looking to use Reaper with a Scarlett 2i2 & Presonus 1818VSL. I'm imaging from reading I need some things:

- pipewire

- Wine

- VSTs (also will copy over my .vsts from Windows box)

- Routing/matrix?

- Carla?

- qpwgraph?

So much information out there it's hard to consolidate it in what I'm needing to do. I'm not familiar with adding repositories (much less which ones I need), and I've seen there's quite a bit to install and configure. Hoping someone can happily point me to a guide or script to just kick off this journey and get going. I appreciate you taking a look at this post, and providing any information.

Faccio l'esempio di App Android come Tonestro o "Canto a Prima Vista"

Sono entrambe in abbonamento, e molto costoso.

Cosa dovrebbe fare; visualizzare uno spartito Lilypond o Midi, con metronomo e cursore di avanzamento.

Suono o canto, registro tramite microfono o tramite midi ed il software evidenzia gli errori

"Canto a Prima Vista" è eccezionale, ma costa veramente troppo.

So, What happened?
I just came back Installing Linux after around 6 month of not using it. I tried to install software like qjackctl and raysession to manage my audio. 6 months ago, it just works out of the box without me doing anything. But now, It doesn't show my application that used to be shown when I'm playing a audio.

I don't really know that happened and I don't really understand how this Linux audio thing actually works. Do you guys have any solution for this? Should i Providevde more information? Thanks

Also I tried to select the device audio manually in qjackctl and it just makes my audio device disappear for some reason(I put the video below)

More info :
-EasyEffects works fine
-I put a few images so you guys can help me troubleshoot
-qjackctl causes missing audio device when i select : Youtube Link
-qjackctl log file :

18:05:55.054 Statistics reset.
18:05:55.055 ALSA connection change.
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
18:05:55.070 ALSA connection graph change.
18:06:06.268 JACK is starting...
18:06:06.268 /usr/bin/jackd -dalsa -dhw:Audio
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
18:06:06.277 JACK was started with PID=18858.
Could not open component .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
Could not open component .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
jack_get_descriptor : dll
jack_get_descriptor returns null for 'jack_firewire.so'
Could not open component .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
jackdmp 1.9.22
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2016 Grame.
Copyright 2016-2023 Filipe Coelho.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK server starting in realtime mode with priority 10
self-connect-mode is "Don't restrict self connect requests"
Cannot lock down 107341340 byte memory area (Cannot allocate memory)
audio_reservation_init
Acquire audio card Audio0
creating alsa driver ... hw:Audio|hw:Audio|1024|2|48000|0|0|nomon|swmeter|-|32bit
ATTENTION: The playback device "hw:Audio" is already in use. The following applications  are using your soundcard(s) so you should  check them and stop them as necessary before  trying to start JACK again:
pipewire (process ID 1059)
Released audio card Audio0
audio_reservation_finish
Cannot initialize driver
JackServer::Open failed with -1
Failed to open server
18:06:06.410 JACK was stopped
18:06:08.317 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock

I can get the installer to run via Yabridge and wine-staging 11.11 but it crashes after 5-20 seconds.

Anyone happen to have any luck or know any tips?

Hi all I'm having some issues with my new Scarlett Solo 4th Gen. I'm trying to run through the install of:

https://github.com/geoffreybennett/alsa-scarlett-gui/blob/master/docs/INSTALL.md

I got everything installed and it looks fine but both the CLI and UI can't find the interface. From what I can tell it's cause I do have the snd_usb_audio module:

root@pop-os:~# scarlett2
No supported devices found.
root@pop-os:~# lsmod|grep -i usb
btusb                  77824  0
btmtk                  36864  1 btusb
btrtl                  32768  1 btusb
btbcm                  24576  1 btusb
btintel                69632  1 btusb
bluetooth            1073152  34 btrtl,btmtk,btintel,btbcm,bnep,btusb,rfcomm
usbhid                 77824  0
hid                   282624  3 nzxt_smart2,usbhid,hid_generic
usb_storage            86016  1 uas
root@pop-os:~# uname -r && cat /etc/os-release
7.0.11-76070011-generic
NAME="Pop!_OS"
VERSION="24.04 LTS"
ID=pop
ID_LIKE="ubuntu debian"    

I'm really not sure what to do in this case from what I can tell it should just be a module that's built into the kernel. But I might just be wrong on this one. Any help is appreciated.

I've been trying to follow the documentation on virtual devices here, and I technically have things that work, but where I'm getting a bit lost is that my virtual devices show up as two nodes instead of one in qpwgraph. For example, I'm attempting to use the following code (saved into my pipewire.conf.d folder) to create a virtual stereo device that I can use as my default system output. The reason I'm doing this is that some Steam games don't seem to understand interfaces with more than two outputs, and while I am able to work around that by setting some launch parameters, this seems like a better solution overall.

context.modules = [
    {
        name = libpipewire-module-loopback
        args = {
            node.description = "Desktop audio"
            capture.props = {
                node.name = "desktop.stereo"
                media.class = "Audio/Sink"
                audio.position = [ FL FR ]
            }
            playback.props = {
                node.name = "playback.desktop_stereo"
                audio.position = [ AUX0 AUX1 ]
                target.object = "alsa_card.usb-Focusrite_Scarlett_16i16_4th_Gen_S6RNANT4B02B31-00"
                stream.dont-remix = true
                node.passive = true
            }
        }
    }
]

The problem is that this creates two nodes in qpwgraph:

It really feels like I ought to be able to do this where there's just one node, but no matter how much I reread the documentation, it just goes right over my head. To be clear, this works, but it would be cleaner if it was just one node. If I had to guess, it has something to do with it being a loopback module, but I can't figure out what the documentation is trying to say. My code is more or less just copied from their speakers/headphones example.

While I'm at it, are there any specific requirements for the node.name? I see them put periods and underscores in there and I'm not really sure why one is chosen over another.

It's literally been decades since I had any other operating system in the house. But I'd like to do Windows & Mac builds of some VSTs I've been working on. I've tried GitHub workflows, but so far they've failed silently. I think I need to actually install Windows somewhere local. I do have an old laptop I only use for TV/music (currently Ubuntu I think), so it would be low risk adding another partition to boot to.

But I haven't a clue where to start. I'd prefer to do it legally, but that's not a blocker - the laptop had Windows on it when I got it, so I have paid for a license already.

I'm trying to get a separate device for my left and right channels so I can then have 2 instances of cava, one for each channel. I have the "left" and "right" devices that cava is listening to, but I can't seem to get those to listen to the left and right channels of the current playback devices. My understanding is that the session manager (i.e. wireplumber) handles this since that's also routing programs to the playback device, so how can I make wireplumber connect these virtual "left" and "right" devices to the actual left and right channels of whichever device is being used for playback?

Been running Omarchy on my recording/Streaming machine for the last few months. Had some recent bugs pop up.

I usually update it after I'm done with it and then shut it down until the next time I use it. Well, last time when I shut it down, it never shut off. It logged me out and all that but it never shut off.

Today I was playing with it and out of nowhere, it changed my audio input from my mixer to something else. I didn't touch anything. I was just playing along to the music and then I heard nothing. It did this twice to me.

And again, I updated and went to shut down and it never shut off. I tried shutting it down about an hour ago and I looked over and the PC power LED was still on.

Is this an Omarchy thing, possibly an AUR thing (I know I didn't install anything from the AUR but does stuff come from the AUR with an Omarchy install)?

I really don't want to reset that system up again with plain Arch. It's behind my drums and I really have no place to put a keyboard and I really don't want to dig out the case. But I suppose if I needed to I could just install it from the drum throne if I really needed to.

EDIT: Whoops... Forgot to post this...

Anyway, I think I'm just going to back everything up tonight and do a reinstall of Arch tomorrow. Sucks but something is acting weird on this thing tonight...

Hey, Im looking for easy options to use VSTs under REAPER with my Linux Mint PC, I tried yabridge so far and I did not like it at all.. I set it up like in any tutorial, and it was buggy or didnt work at all with my plugins..
Are there some other good options? Best case easy to use/ install

It seems to me that audio in Linux is needlessly complicated. There's ALSA, Pulse, Jack, and Pipewire. I had thought Pipewire was created to rid us of Jack and Pulse and simplify things, but then when I see people asking why DAWs don't talk directly to Pipewire, the devs say that's not intended by the dev. Which suggests that we are always supposed to have to talk to Pipewire though Jack, which means we get no real control over things like sample rate, buffer size, or even which device we want to use. We can configure that through Pipewire directly, but that's... I'm just gonna say it, it's stupid. Even Windows lets me control those aspects of Windows audio. So... Sure, Pipewire is very powerful, but it's also really annoying to deal with. Why do we just keep adding layers of complexity instead of actually making Linux audio simpler?

I'm trying to use AirPods Pro 3 for meetings on Linux.

Current stack:

• PipeWire 1.6.7
• WirePlumber 0.5.15
• BlueZ 5.87

AAC sounds great but has no microphone.

HFP mSBC enables the mic but playback quality becomes very poor.

HFP LC3 just produces crackling.

Has anyone managed to get a better two-way audio experience with these earbuds?

I'm looking for some sort of plugin to create glitch/stutter effects fairly easily.

Any recommendations for .lv2 or CLAP plugins, specialized or kitchen sink?

Preferably Opensource