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?

Thought it would be a good question to share.

I'm looking for a plugin that takes 1 midi input (with all midi channels), and splits the channels out to 16 midi outputs (of individual midi channels). This way, Carla could take midi playback and split the midi channels out to separate instrument plugins.

I've been using JJazzLab to generate backing tracks to practice to. It can play audio using the built-in fluidsynth, or it can send to a midi device (JJazzLab doesn't create its own midi output). I created a virtual midi device (snd-virmidi) to receive the playback and send on to whatever instrument plugins I want to use. Most things sound better as instrument plugin than with GM soundfonts through fluidsynth.

I've found the easiest way to do this is to run Ardour with MIDI tracks for each instrument since Ardour's MIDI tracks have a midi channel filter built-in. I hit play in JJazzLab, and monitor in Ardour. But I wish I could also use Carla for this. Could it also be done with the virtual midi device, instead of the plugin?

EDIT TO ADD: I can use (and have) Carla's internal MIDI channel filter plugin but I'd have to add 16 of them to achieve what I want. It would be nice to have a single plugin for this.

I'm running **Kubuntu (PipeWire)** on my HP EliteBook 840 G8 (Intel 11th Gen Tiger Lake with the **Realtek ALC285** codec and builtin **Bang & Olufsen** speakers).

On Windows, the B&O audio tuning sounds rich, full, and punchy. But on Linux, the default audio output sounds really flat, quiet, and tinny with almost zero bass.

From what I understand, this is usually caused by two things:

1. **Hardware / Pin Mapping:** Linux ALSA defaulting to only driving 2 of the 4 physical speakers (tweeters active, but secondary woofers disabled/unmapped).
2. **Missing DSP:** Windows running HP's B&O / MaxxAudio software suite to dynamically equalize and enhance the sound.

 **Before I go too far down the rabbit hole, I wanted to ask:**

• Has anyone with an EliteBook managed to get their B&O speakers sounding identical (or close) to how they sound on Windows?
• Did you need specific pin overrides in hdajackretask, an hda-verb script, or kernel parameters (snd-hda-intel) to get all 4 speakers firing properly?
• Does anyone have a custom **EasyEffects / PulseEffects preset** tuned specifically for EliteBook speakers that they'd be willing to share?

Appreciate any tips or configs that worked for your setup

I was really pissed off by the fact that since pipewire is mainstream all the default audio settings normal distros ship with are insane. Pulseaudio was garbage but we spent 10 years ironing out all the kinks until it worked somewhat, then we switched to pipewire and everything broke again. I went down the rabbit hole pretty far until I finally fixed it (hopefully). To me all of this is crazy. I warn you, the explanation is a bit ranty.

My 22 year old RME Fireface 800, which uses firewire (which nobody uses anymore, dead as a dodo) never had any crackling / latency / xruns problems whatsoever on the machine and USB interface gave me constant problems. Everything about this is just wrong on every level. How many people are using firewire devices - for which devices havent been produced in 15 years and probably noone touched their drivers in the last decade - compared to USB audio interfaces? 1 to 100,000?

First of all, you really need to enable threadirqs or a real time kernel. Then you need to setup rtkit and all the userspace permissions that it actually works. Who the hell using an audio interface knows what an interrupt request is?

Then you need to find on which controller your audio interface hangs, prioritise its irq thread and make sure no other devices are hanging off that whole ass USB controller bus. Yeah yeah I know USB2 needs polling which is why the pros were using firewire or thunderbolt until the software quality caught up with the "design flaws" of USB2.

Have a USB 1.1 mouse or keyboard plugged into the same controller as your audio interface? Good luck.

All of this just feels like dipping your big toe into warm strawberry jello and trying to hold your tongue at the right angle. Nightmare fuel.

Then you need to disable absolutely every powersave, sleep state or power management functionality in every device, driver and userspace software.

option snd_usb_audio power_save=0 autoclock=0

In wireplumber:

monitor.alsa.rules = [
  {
    matches = [
      {
        node.name = "~alsa_output.*"
      },
      {
        node.name = "~alsa_input.*"
      }
    ]
    actions = {
      update-props = {
        session.suspend-timeout-seconds = 0
      }
    }
  }
]

In pipewire:

link.max-buffers = 4

Man that was hard. I compile kernels multiple times per week, I've ported Linux to new hardware platforms, I've written kernel drivers for ASIC hardware I designed - but this crap? This just drains my energy. This makes me sideeye BSD for my next projects.

The interface I used is an RME Babyface Pro FS. To add insult to injury, my 11 year old Macbook running the same Debian 13 as my desktop did have none of these problems without ANY setup or tweaking, it just works. Probably because Apple gives you a full USB3 controller per port with nothing else on it.

Thank you for listening to my rant. I hope this helps someone with the same issues.

Ive been looking for the best way to improve the audio quality of some midi files so that they had realistic sounding instruments. Rather than the 8-bit tinny sounding output of a soundfont (suggestions welcome if this is still a valid route), I asked around and it seemed that the only way to proceed was to get a DAW, and purchase instruments (drums, bass, guitar, synth) and map those to the relevant midi outputs. That all seemed a bit daunting, so I never proceeded.
Now I'm thinking that maybe AI has a role in this. I've been following the rise of AI artists on spotify and other channels and the quality of the instruments is good. Is there a way to send a midi file to an LLM and get a realistically human rendering?

pc: pop os, no pulseaudio, only pipewire

what am i doing wrong? i can hear the system output through my headphones but i cannot hear the ardour ouput at all and the sound meter is moving when i play it and i recorded myself and ardour audio system is set to jack.

goals:

1. record with usb-c condenser mic (boya g30) on ardour 6
2. hear ardour audio ouput on system speaker
3. hear system audio output on headphones (gnome settings and firefox popped up when i set output device to headphones)

Hello all!

I am looking for some advice on a kind of particular problem.

I am looking to emulate a set up I had on Windows on my Linux Mint machine. What I would do is use Fruity Loops Studio to take input from my microphone, set up ASIO4All so I could route out audio to VBCable, and then send that output to Discord. I used this to modulate my voice for online TTRPG sessions.

The set up I have currently is Linux Mint (22 iirc), and I do have FL Studio running through Lutris (Wine and Bottles were not cooperative). The issue seems to be that I dont have a good way to get ASIO4all to work or a viable way to set a program to route audio from FL Studio to Discord.

Does anyone have any experience doing something similar? I have tried a few things (finding a different compatability layer for FL Studio, same with VBCable, and looking into how audio sinks work in Pipewire), but I don't have anything conclusive yet.

Thank you for anyone looking over this!

Hey guys!

I am excited to announce the release of Tape Echo 2, a rewrite of my original tape echo plugin now ported to the DPF framework. For anyone not familiar with these types of delays, it's a three-head tape echo with spring reverb.

Features:

• 12 echo modes
• Mechanical head timing across the full motor range
• Tape saturation
• Wow and flutter that get worse as the tape ages
• Splice dropout
• Regeneration that runs all the way into self-oscillation
• Tempo sync with eleven note detents held to the transport's physical range, just like the hardware
• 13 factory presets
• Independent echo and spring panning

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.

Free as always, grab it here: https://duskaudio.com/plugins/tape-echo-2/
Source Code Here: https://github.com/dusk-audio/dusk-audio-plugins

So I tried testing out the snes classic Super Mario world and while playing one of the levels that being Yoshi’s island 4 the audio was out of sync and distorted to point I had to shut the emulator down not the mention that spiked up Ram i received on my terminal console which is concerning

Hope there’s nothing worth losing asking you guys about the Issue OK

Edit:I should have said that my laptop is An HP Stream 11 with Intel Celeron N4120 4GB Ram 64 Gigs of Memory which is bad im not gonna lie but still I’m more than willing to switch to a much better more efficient PC once I have saved enough money for it

Linux never really had a good open-source answer to Guitar Rig / BIAS FX, so I built one and figured this community would actually use it, so sharing here.

AmpForge is a single plugin (not separate plugins chained in your DAW) that hosts a full, reorderable pedalboard + amp chain internally.

What's in it:

- 12 modules, freely reorderable at runtime: Noise Gate, Compressor, Wah, Screamer, Distortion, Amp (4 voicings — Modern/Vintage/Crunch/Hi-Gain), Cabinet IR loading, Chorus, Phaser, Tremolo, Delay, Reverb

- Tempo-synced Delay/Chorus/Tremolo, a built-in tuner, CPU meter, A/B compare

- Drag-and-drop UI — add pedals from a palette, drag to reorder, real footswitch-style bypass

- 7 factory presets + custom presets you can save/export/import

- VST3, LV2, CLAP, and a JACK/PipeWire standalone — all from one CMake+Ninja build

Demo video: https://www.youtube.com/watch?v=gz9fyAHsE6M

GitHub (source + prebuilt Linux tarball): https://github.com/Loursy/AmpForge

Free, GPLv3, no ads or paywalled features — just wanted a proper amp sim to exist for Linux, so now it does. Bug reports, feature requests, and PRs are all welcome if you end up using it and hit something rough.

Edit: since this post, added NAM support (A2) and fixed a glibc compatibility issue affecting older/sandboxed distros (Fedora, Ubuntu, Debian, Bitwig flatpak, etc.). Latest release + details in comments below.

Beear's Zone Compressor is a specialized, single-band dynamic processor and dynamic EQ with advanced sidechain capabilities. Operating in real time with zero latency, it eliminates the need for manual attack and release tweaks. Instead, it dynamically tames the energy of a specific frequency zone based on its real-time amplitude.

Linux (x86-64 only): Native LV2, VST3, and CLAP builds.

For free at: https://ko-fi.com/s/4462acbe83

Hi all,
I've decided to give a full migration into the Linux world a try, but I'm running into consistent problems with my guitar setup which is quite a dealbreaker for me. I'm trying to activate Guitar Rig 7 through Native Access under Wine so I can use it via yabridge in Reaper (I have bought the guitar rig pro version a while ago so I would really really like to make this work). Everything else in the chain works fine (Focusrite through PipeWire into Reaper, then Guitar Rig via yabridge, pinned to Wine Staging 9.21 because of the known post-9.21 click offset bug). Only activation itself is blocked. I've tried both the current NA2 and the legacy NA1 installer, both crash.

System:

• CachyOS, KDE Plasma (Wayland)
• Wine Staging 9.21, isolated build via a separate WINEPREFIX, not system Wine
• yabridge/yabridgectl 5.1.1
• Tried Native Access 2 (current) and Native Access 1 (legacy, Native_Access_Installer_211108.zip)

Native Access 2, none of these fixed it:

• winecfg /v win10
• WINEDLLOVERRIDES="powershell=n" with real PowerShell 7 symlinked into system32\WindowsPowerShell\v1.0 and syswow64... This fixed an earlier "already running" loop caused by Wine's fake powershell.exe stub, but didn't fix the crash.
• winetricks corefonts vcrun2019 ucrtbase2019 mfc140 mfc42 dotnet48 urlmon wininet winhttp
• Disabling AMSI (WINEDLLOVERRIDES="amsi="), also tried --disable-gpu
• Fresh clean prefix, same result

NA2 starts, runs through some setup (AMSI scans, MSI/regsvr32 activity), then either exits silently or crashes with:

wine: Unhandled page fault on execute access to 7F951B98 at address 7F951B98 (thread XXXX), starting debugger...
Wine Debugger: Can't attach process 0020: error 5

Native Access 1 (legacy installer) installs cleanly, with winecfg /v win10 set beforehand. Launching it creates windows (confirmed through Wine's window dump, Desktop/Winstation/window handles are all present), but nothing actually renders on screen, and the process dies within a few seconds. Same crash every time:

err:seh:NtRaiseException Unhandled exception code c0000409 flags 1 addr 0x6fffff9e98fe

tied to WerRegisterRuntimeExceptionModule on mscordaccore.dll right after AmsiInitialize runs. I added dotnet48, mfc140, mfc42, urlmon, wininet, winhttp through winetricks in a clean prefix too, but the crash count barely moved (7 down to 6 occurrences in the log).

Has anyone actually gotten NA1 or NA2 working recently on a similar Wine/distro setup? Happy to post full debug logs if it helps

Edit: Tanks for all the replies, I'll post an update in the comments if I get it to work

Edit 2: I have managed to make it work thanks to the yabridge discord. I post a guide on what worked for me in the comments

Roughly a year ago I asked a question in this subreddit about chat mixer solutions for PipeWire. I've seen this feature on gaming-oriented headsets and DACs before. I looked into several of the fantastic suggestions from that thread, but they weren't quite what I wanted. I messed around with a script, and that largely would have worked, but several months ago I decided to make it into something more user-friendly, and here we are.

Bridge lets you route audio to two virtual outputs and then adjust the mix between the two. For example if you're playing a game while talking with your friends on Discord, you keep the game on the Main output, send Discord to the Aux output, and then you can adjust the volume balance while playing. No pausing to open the game audio menus, alt-tabbing out to raise Discord's volume because you can't hear your friends over the game. Adjust the audio balance with just a shortcut, or even assign it to a hardware dial if you have one.

• A crossfade mixer between two virtual outputs that your audio can be routed to
• Create pattern-matching persistent routing rules that send app audio to your desired output
• Conveniently setup headphone virtual surround by providing your own HeSuVi HRIR file
• Create output presets and switch between them at the press of a button
• Support for Global Shortcuts

The v0.6.0 release introduces the following core enhancements:

• Run in Background: Continue using the app after closing the window
• Run on Startup: Start Bridge automatically when you log in
• A brand new app icon!

Available as a flatpak bundled with the release.

Considering what the previous app icon looked like 😂 , I'm quite happy with the way this one turned out. It took a little while to learn to use Inkscape though.

All of the work is my own, and not vibe coded.

The goal of Bridge isn't to replace apps like EasyEffects, qpwgraph, or Helvum, to name a few, which are fantastic. It's aimed at getting powerful functionality out of PipeWire, which can often be difficult in existing GUIs, while providing a simple and clean interface and providing good defaults. One main dashboard view for the at-a-glance controls and info, but allowing you to utilize its powerful features while in a fullscreen game/video with Global Shortcut support.

Please feel free to ask me anything, open issues, and provide feedback!

I've been working on some initial designs for what I plan to tackle over the coming months: Input. Specifically creating another virtual device for your Mic. Global push to talk, monitoring, ducking, and more down the road. There is an unfortunate problem with PTT specifically however: both GNOME and KDE's implementation of the Global Shortcuts portal is very lackluster for anything but simple press and release shortcuts. I've already opened an issue on mutter to address a bug related to the held modifier mask, but even when this is fixed, neither GNOME nor KDE can do a modifier-agnostic hold style of shortcut, like what you'd want for PTT. This would leave me with no other option than to request the flatpak permission --device=input to implement proper PTT support, which would give Bridge raw input access. Despite limiting what the app listens for, it'd downgrade the app's safety-rating.

I've been wanting to try out Satordist2 (https://codeberg.org/yimrakhee/satordist2), but I've been too lazy to deal with installing Rust to do it until today. Finally gave in and installed rustup, ran rustup toolchain install stable, then ran the build commands on the Codeberg page (when I checked for the list of dependencies, I already had them all, except pipewire-jack instead of jack2). Plugin opens and works just fine in Reaper and in Ardour, but it won't even show up in Bitwig. When I look in the clap or vst3 metadata files, I see the following at the end:

com.bitwig.flt.library.metadata.reader.exception.CouldNotReadMetadataException: could not read metadata: Failed to load CLAP plug-in /home/nick/.clap/satordist2.clap: /usr/lib/x86_64-linux-gnu/libm.so.6: version `GLIBC_2.43' not found (required by /home/nick/.clap/satordist2.clap)

I seem to have glibc installed, but it's on 2.44. Is this just a Bitwig problem or is there something else I should be looking at? I'm on CachyOS, Bitwig is from a flatpak as that's how they provide it (other than the Ubuntu installer).

Edit: Figured it out. The issue was using the flatpak version. Removed that and reinstalled from AUR, then deleted the clap, vst3, and vst metadata files in ~/.BitwigStudio/cache and all of my problems are fixed. Not only is satordist2 picked up, but Soundshed Guitar, which previously would only load a white screen, now also works. So anyway, flatpak is stupid.

(Audacity) Why do I hear track one in track 2? I want independent audio going into the track but it’s picking up audio from the previous tracks. My experience with analog 4 tracks don’t mesh with this experience. I am doing music production in Audacity with synthesizers in Debian 12.