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.
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