Problem: consistent latency between microphone and playback - around 1s. The quality’s also not ideal for calls, with some reports of it being crackly but audible that I wasn’t picking up in local playback. There’s no latency or quality issue with incoming audio, and video/screensharing are also okay. I’ve had instances where the mic latency gets into the 5s+ range, but can’t reproduce that.
Consequence: I’m using a separate system for all audio/videoconferencing. Might not be a bad idea but it’s unexpected and not my preference.
First attempted 9-12 months ago, and I did some digging 6 months ago. I was hoping for more focus time or inspiration since then, but haven’t found either.
Same results:
- using Google Meet, Zoom, and a clean jitsi server install on my local network
- using the built-in microphone and a separate one plugged into the audio i/o port. I don’t want to get extra hardware in right now just for this testing.
- giving more resources to the audio+client vms
- changing the virtualization/memory balancing/PCI configurations
- using multiple WLANs (no ethernet option)
Same hardware with a live boot Ubuntu did not have the issue. iirc this was using pulseaudio and not pipewire. lsmod looked very similar, but I don’t do much with drivers and could have missed something.
I’m trying to avoid deep-diving into pipewire. I’ve followed some of the basic pipewire latency guidance like setting environment variables, verified that they’re set correctly, and can’t see a difference.
I tried to test at different layers with arecord -- | aplay
and similar things with pacat
/pactl
/pw-cat
. Some of these seemed to reproduce a similar lag even when configured to remove built-in latency, but it turned out they also reproduced it on the working Ubuntu system where audio calls were fine. So I didn’t find a cheaper way to test than setting up a call between two systems.
I haven’t yet tried testing with calls between different local VMs - looks like this should be okay to set up so it’s my next step.
I also wanted to test dummy input instead of the mic, but haven’t worked out how to do that yet.
I haven’t tested with Qubes on a separate system. I’m assuming it is somehow hardware-specific at this stage, mainly because I can’t find similar reports and my system’s fairly clean.
I’m on Qubes 4.2.3 with an audiovm. No other problems with it.
I’ve checked the forum and issue tracker and not spotted anything very similar or recent. I’d like to know if anyone has had a similar problem, and if there’s anything obvious that people who know their audio would check, or anything else likely to narrow down the cause.