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Modules – PulseAudio
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/
The address used to listen for SAP announcements, defaults to 224.0.0.56. It can be either a multicast group or a unicast address. latency_msec The desired latency due to local buffering (the network latency and buffering at the sender's end are beyond PulseAudio's control and knowledge, so they aren't counted here).
PulseAudio/Examples - ArchWiki - Arch Linux
https://wiki.archlinux.org/title/PulseAudio/Examples
sap_address=0.0.0.0 is important to prevent pulseaudio from trying to use multicast, which doesn't work at all over wifi. Use latency_msec to tune the receiving buffer size on the remote end. If you find the audio is spotty, try increasing this number.
audio - Is Pulseaudio able to receive RTP Multicast from ...
https://unix.stackexchange.com/questions/392728/is-pulseaudio-able-to-receive-rtp-multicast-from-any-source
1. This answer is not useful. Show activity on this post. I am not sure what exactly you tried (you didn't specify), but I can get two pulseaudio servers to communicate via multicast RTP in the following way. On the sender, pacmd load-module module-rtp-send source=name_of_mic_source destination_ip=232.43.211.230 inhibit_auto_suspend=always.
networking - PulseAudio RTP unicast poor sound quality ...
https://unix.stackexchange.com/questions/471930/pulseaudio-rtp-unicast-poor-sound-quality-frequent-pops
Since then I have added latency_msec=1000 to module-rtp-recv on each receiver. On the sender, I am thinking about adding rate=44100 channels=2 format=s16le. However, those are already the defaults on all devices: PulseAudio Version: 12.2; Default Sample Specification: s16le 2ch 44100Hz; Also, all are synchronized with an NTP server:
linux - ALSA vs PulseAudio - Latency Concerns - Stack …
https://stackoverflow.com/questions/29245583/alsa-vs-pulseaudio-latency-concerns
Thus it can offer lower latency than other applications using the same buffer size, but more importantly, this allows it to adjust the latency dynamically without having to stop and reconfigure the device. Other applications could do the same, but it's easier to use PulseAudio than to implement that buffer handling again.
GitHub - VittGam/mtrx: Transmit and receive low-latency ...
https://github.com/VittGam/mtrx
Change receiving latency with -e if needed; If having problems try sudo ./mrx -d pulse; On OpenWrt and/or with cheap USB audio cards without PulseAudio, if it doesn't work try mrx -d plughw:0,0; It shouldn't be needed anymore, but it might still be useful, so this is a working /etc/asound.conf file for OpenWrt with cheap USB audio cards; Bugs
Amazing performance boost by tweaking pulseaudio …
https://www.reddit.com/r/leagueoflinux/comments/dr2qye/amazing_performance_boost_by_tweaking_pulseaudio/
1 year ago. Archived. Amazing performance boost by tweaking pulseaudio settings? (Needs confirmation) TL;DR: My rhythm gamer senses told me to tweak pulseaudio latency settings and it somehow made LoL less stuttering. Edit: Using Esync also seems to help. My system: Arch, i3-gaps, no compositor in-game, using optimus-manager to use only Nvidia.
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