r/linuxaudio

Most voice input on Android - SpeechRecognizer.createSpeechRecognizer(context) calls — gets routed to Google's network-backed recognizer. I wanted that path to run locally, so I wrote one.

The service hooks the framework's SpeechRecognizer API. Once it's set as the default, any app calling createSpeechRecognizer(context) (no ComponentName) ends up in our pipeline and gets back transcription that never left the device. Pipeline is Silero VAD + Parakeet TDT v3 (114 languages, ~890 MB INT8) on ONNX Runtime with NNAPI.

Honest caveat: Gboard, Samsung Keyboard, and Google Assistant ship their own recognizers and skip the system default. So the default-IME voice button on most phones won't go through this. What does: accessibility tools, custom dictation UIs, and anything calling the framework API directly.

Models download on first use (~1.2 GB) via a foreground WorkManager job so it survives backgrounding. After that, fully offline.

Setup + demo APK: github.com/soniqo/speech-android

audio.soniqo:speech:0.0.9 on Maven Central

Library:

Happy to answer questions about the binder lifecycle, the foreground worker setup, or why SpeechRecognizer is such a tarpit of edge cases.

Before buying the MPC live 3 I tried to find if the built in interface would work on Linux but couldn't find an answer. There were hints and I suspected the MPC to be class compliant but there were drivers to be installed for windows and macos although it should just work on iPad and iPhone.

Anyway, it just works. 24 channels of audio in and out. 3 midi ports.

I would expect the new One G2 and the flagship XL to be the same.

I’ve got a cheap USB mic (Jmary MC-PW10, chip is Medeli, USB ID 0a67:d156) that works flawlessly on Windows and on my Android phone, but on my Linux laptop it won’t work reliably, and I’ve run out of ideas.

The frustrating part: same mic, same cable, same USB-C-to-USB-A adapter, same type of USB-A port. On Windows and Android it’s instant and rock-solid,including switching modes with the button.

On Linux it either won’t enumerate, or connects and then disconnects within a second, and pressing the mic’s mode button reliably drops it. Since the physical connection is identical across all three, it looks like a Linux USB-stack issue rather than a cable/power problem.

Setup: Dell Latitude 7480, Linux Mint 22.3, kernel 6.17, Intel xHCI, PipeWire. (The laptop’s USB-C port is dead, so I’m stuck using the USB-C-to-USB-A adapter — can’t test a direct USB-C connection.)

Two failure signatures I see: - Enumerates cleanly, then disconnects ~1s later at rest, sometimes with no error, sometimes with error -71 (device not accepting address / can't set config) or error -32. - Pressing the hardware mode button → usb_set_interface failed (-19) → disconnect.

When it does stay connected it works perfectly (shows up in PipeWire, records fine in OBS at 24-bit/48kHz), so the hardware and audio are fine.

Already tried (no luck): powered hub + external 5V, three different cables, autosuspend disabled, usbcore.quirks=0a67:d156:gi (detects but drops) and :gik (breaks detection entirely), and snd_usb_audio quirk_flags=0x02/0x200. Already on a recent kernel (6.17).

What I’m hoping someone knows: a usbcore.quirks or snd_usb_audio quirk_flags value that makes Linux tolerate this chip, or whether forcing the port to USB 2.0 helps this class of device.

Full writeup with all the logs and dmesg excerpts here: [https://unix.stackexchange.com/questions/806610/usb-mic-medeli-0a67d156-works-on-windows-android-but-fails-on-linux-xhci]

Has anyone gotten a Medeli/JinAudio USB mic like this stable on Linux? Any quirk value or trick appreciated.

All very experimental, mostly vibe-coded, rather unintuitive UIs, but there is already plenty to have fun with. Time to play, see what works and what doesn't.

https://github.com/danja/downspout

Feedback appreciated. Better still, please improve them youself! (Anyone kitted out and willing to build for Windows/Mac, let me know, I'll add you as a collaborator).

I used a lot of the same algorithms in earlier lv2 plugins on an album I just released : https://github.com/danja/attone

* bassgen: a transport-aware MIDI generator

* p-mix: a transport-aware probabilistic fader

* e-mix: a transport-aware Euclidean fader

* rift: a damage effect, invented by Codex

* drumgen: MIDI drum pattern generator

* cadence: MIDI harmonizer and comping generator

* counterpointer: a transport-aware MIDI counter-melody generator

* gremlin: a chaotic glitch instrument with live performance gestures

* gremlin-driver: a MIDI modulation and action sequencer for gremlin

* ground: long-form MIDI bass generator

I've been testing Tape 16 and trying to fix two issues I'm having with it before I decide to purchase it. Using JACK as my audio backend I get choppy audio on monitoring and playback. But when switching over to ALSA as the backend and using what shows up as "JACK Audio Connection Kit" as my input and output source, I get clean audio on monitor and playback. Switching back to JACK and playing back the audio recorded through ALSA then makes the ALSA audio choppy. This happens on my Debian 13 system but not on my Kubuntu 24.04 system; everything runs smooth on Kubuntu.

I built up the Debian system manually as a base install with I3 on top. The Kubuntu system is Kubuntu with the Ubuntu Studio Audio Package installed. Could my Debian system be missing something that I may not have installed that's causing this to happen or is this a bug in the software?

A second issue, which I can do without right now, is that it won't send out MIDI Clock. Also, something that works on my Kubuntu system.

I'd like to note that I currently run Reaper on the Debian system and everything works perfectly.

I appreciate any and all help and/or suggestions.

Hello! As the title says. The volume only decreases normally when using slider from other program such as Youtube or In-Game audio settings, but seems to not work using keyboard shortcuts nor the GUI slider, it loses clarity but stays as loud. I messed with alsamixer before because the audio sounds weird after installing EasyEffects, but I was simply maximizing all the bars that I can slide without knowing anything. I did sudo alsactl init but the problem persists. How to begin troubleshoot it?

Operating System: CachyOS Linux KDE Plasma Version: 6.7.2 KDE Frameworks Version: 6.27.0 Qt Version: 6.11.1 Kernel Version: 7.1.2-3-cachyos (64-bit) Graphics Platform: Wayland Processors: 16 × AMD Ryzen 9 270 w/ Radeon 780M Graphics Memory: 32 GiB of RAM (30.6 GiB usable) Graphics Processor 1: AMD Radeon 780M Graphics Graphics Processor 2: NVIDIA GeForce RTX 5050 Laptop GPU/PCIe/SSE2 Manufacturer: ASUSTeK COMPUTER INC. Product Name: ROG Zephyrus G14 GA403UH_GA403UH System Version: 1.0

Command: lspci -k | grep -iA 3 audio grep Codec /proc/asound/card*/codec*

pcilib: Error reading /sys/bus/pci/devices/0000:00:08.3/label: Operation not permitted 01:00.1 Audio device: NVIDIA Corporation GB207 High Definition Audio Controller (rev a1) Subsystem: NVIDIA Corporation Device 0000 Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel

-- 65:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Radeon High Definition Audio Controller Subsystem: Advanced Micro Devices, Inc. [AMD/ATI] Radeon High Definition Audio Controller Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel

65:00.2 Encryption controller: Advanced Micro Devices, Inc. [AMD] Phoenix CCP/PSP 3.0 Device

65:00.5 Multimedia controller: Advanced Micro Devices, Inc. [AMD] Audio Coprocessor (rev 63) Subsystem: Advanced Micro Devices, Inc. [AMD] Audio Coprocessor Kernel driver in use: snd_pci_ps Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, snd_pci_acp6x, snd_acp_pci, snd_pci_ps, snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh, snd_sof_amd_acp63, snd_sof_amd_acp70 65:00.6 Audio device: Advanced Micro Devices, Inc. [AMD] Ryzen HD Audio Controller DeviceName: Realtek ALC256 Subsystem: ASUSTeK Computer Inc. Device 1044 Kernel driver in use: snd_hda_intel /proc/asound/card0/codec#0:Codec: Nvidia GPU ae HDMI/DP /proc/asound/card1/codec#0:Codec: ATI R6xx HDMI /proc/asound/card2/codec#0:Codec: Realtek ALC285

Not sure why my audio look like this it sound fine just a bit distorted
How ever when applying de-noise effect from the OBS it fixes the issue
I feel like its a software problem since my Røde mic is new (and it happens with different mics)

I have wanted to share a non-game app that I've been working on for a while. My background is in systems programming, but I am also an amateur music producer. This app is a response to my own pain points with other DAWs and modular synths. I have used Ableton, Reason, Fruity Loops, Cakewalk, and even Max/MSP back in the day, but each of them has a considerable learning curve or feels disconnected from certain forms of music.

Most DAWs give you timelines and a grid to place your clips, but this tends to force you into making music that sounds block-like and square.

I wanted to explore more organic musical possibilities for ambient, electronic, and generative music. Most artists who explore that realm end up building Euroracks or very complex custom software in PureData or Max/MSP, which introduces a considerable distance between the idea and the execution. I also needed a composer companion: an app where I could roll out a simple arpeggio in a scale and start jamming alongside it.

I started up writing my own sound processing modules in standalone C# as a custom synth engine library. I built it from scratch because I wanted it to be portable, cross-platform, and embeddable. To dogfood the library, I decided to embed it in a Godot app. Little by little, I built devices and modules until I had something I could share with others. I got carried away and ended up building a programmatic DAW in C# with a Godot UI.

I did this for myself, to make the music I have in mind. I also have some game ideas in my backlog, and this app was a step in the right direction for making music for those future projects, while getting to know the game engine I plan to use for them.

Links & Info:

• Steam (Coming Soon): https://store.steampowered.com/app/4765870/Sigilgraph_Audioworkbench/
• Website http://sigilaud.io

If you've installed a VST plugin recently and its GUI is just a black window under Wine/yabridge -- that's the JUCE 8 Direct2D/DirectComposition issue. Stock Wine doesn't implement the render path JUCE 8 switched to, so there's nothing to draw into. The tell-tale sign, if you launch from a terminal:

DCompositionCreateDevice failed: Not implemented. (0x80004001)

First, a caveat: if the plugin you want has a native Linux build, just use that (Pianoteq, for example) — this is only for Windows-only JUCE 8 plugins like Serum 2.

There is a fix — giang17 wrote Wine patches that implement the missing bits — but building patched Wine and wiring it into yabridge for just one plugin (without disturbing your other plugins) is fiddly. So I packaged it for Arch/CachyOS and wrote up the whole thing, including a prebuilt binary so you can skip the ~40-min compile, and the yabridge dispatcher trick that routes only the patched plugin to the patched Wine:

Full guide + prebuilt binary: https://github.com/mklnln/wine-d2d1-dcomp

Not on the AUR (new registrations are frozen after the June 2026 malware wave), so it's install-straight-from-the-repo for now — instructions are in the README.

All credit for the actual fix goes to giang17; I just packaged it and wrote it down. Happy to answer questions here, but anything about the patches themselves is best directed at their repo.

go check it out here and lmk https://www.capysynth.com/