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[desa]

I have a PC running a library server and a client attached to it using the library server key.

I configured the client to force the server to always convert audio (Client Options -> Audio Conversion -> Conversion: Always convert audio) to uncompressed WAV (Client Options -> Audio Conversion -> Encoder: Uncompressed).

In this configuration everything works as expected and starting or changing tracks takes less then a second, with any sample rate from 44.1 to 192 KHz. It would be fine if it didn't truncate 24 bit samples to 16 bit.

So, I changed the encoder to 24 bit (Client Options -> Audio Conversion -> Encoder: Uncompressed - 24 Bit) and problems began: now starting a track takes from 4 (5 min. 16/44.1KHz track) to 60 (9 min. 24/192 track) seconds, depending - not linear - on track length and sample rate. The behaviour is also a bit inconsistent as sometimes tracks start very quickly, just like they did with 16 bit encoding.

A 60 factor cannon be explained with raw transfer numbers, but I took some network measures just in case: copying a 2.8 GB files from server to client takes 44 secs.

A bug?

Update:
I also installed a third-party UPnP Server (Asset UPnP) on the same PC running the Library Server and I forced it to convert all contents in WAV 24 bit format. Pointing the MC client to this server gives no problems or delays with any sample rate and bit depth.