It seems like avoiding resampling audio isn't possible when using pulse via tcp? Both my client and server are configured with avoid-resampling set to true, and my hardware supports 24/96. However, my audio is always resampled to 16/44. Is this something that just hasn't been implemented or do I have something set up wrong? Thanks. If this isn't possible, it would be great to be able to stream high-res music over the network.
Attachment:
signature.asc
Description: This is a digitally signed message part.
_______________________________________________ pulseaudio-discuss mailing list pulseaudio-discuss@xxxxxxxxxxxxxxxxxxxxx https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
