alternately you could try unicast rtp.
On Wed, Dec 26, 2018, 12:31 PM David Davidović <david@xxxxxxxxxxxx wrote:
Hi,
I'm running a system-wide PulseAudio instance on a headless box connected to my speaker system. The sink is broadcast via Zeroconf and I use it to play music and movies via WiFi from my laptop. The headless box is connected to the WiFi router via a wired connection.
Unfortunately, every couple of minutes or so, the sound stutters for a short time, then returns to normal.
Does anyone have any suggestions as to how to debug the source of these issues? My best bet is WiFi latency. I tried increasing the buffer size (via default-fragments and default-fragment-size-msec) on both the client and server machine but I haven't seen any considerable improvement. The actual network packets seem to always be around 1.5K in size, which would be around 4ms of uncompressed sound with 32-bit stereo samples at 44.1kHz, and that's without any overhead. If my calculation is true, this would explain the stuttering, as the network latency towards the machine is at a baseline of ~2ms but can spike sometimes due to nature of WiFi.
I'd be happy if there was a way to e.g. tell the native-protocol-tcp module to buffer more of the audio and play it, as I don't mind the increased latency; I just want to get rid of the stuttering and all applications I use manage PulseAudio latency compensation quite well.
Has anyone else had this issue and resolved it?
Many thanks for your help.
Best,
--
David Davidović
david@xxxxxxxxxxxx
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