I have a Pulseaudio networked sound setup, where one laptop is connected to my HiFi through a mini-jack, and have pulseaudio configured to export the local sound device, while another laptop is configured to access the remote sound device for audio playback. It's a nice setup, except that buffer underruns frequently occur, resulting in really annoying fallouts in the sound.
Using
PULSE_SERVER=<hifi-laptop> <audio-player>works better, but there are still underruns, and I'd like to keep the ability to dynamically switch between laptop speakers and the stereo.
I've tried to tweak all sorts of parameters (including setting tsched=0 on both machines), but since this setup is only ever going to be used for music playing, is there a way to force the pulseaudio/ALSA internal buffer size (on the wifi laptop) to something large, say 3-5 seconds and thus sidestep any problems?
