About upsampling.
I've found that upsampling in LMS works just fine using those line in
custom-convert.conf:
flc flc * *
# IRFT:{START=--skip=%t}U:{END=--until=%v}
[flac] -dcs $START$ $END$ -- $FILE$ | [sox] -q -t wav - -t flac -e
signed -C 0 -b 24 - rate -v -M -a -b 90.7 192000 dither -S
aif flc * *
# FT:{START=--skip=%t}U:{END=--until=%v}
[flac] -cs --totally-silent --compression-level-0 $START$ $END$ --
$FILE$ | [sox] -q -t flac - -t flac -e signed -C 0 -b 24 - rate -v -M
-a -b 90.7 192000 dither -S
wav flc * *
# FT:{START=--skip=%t}U:{END=--until=%v}
[flac] -cs --totally-silent --compression-level-0 $START$ $END$ --
$FILE$ | [sox] -q -t flac - -t flac -e signed -C 0 -b 24 - rate -v -M
-a -b 90.7 192000 dither -S
To reach this point I've tried MANY other options and at the end I came
to the conclusion that or I missed something (quite sure) or something
in LMS is broken (could be?), in producing pcm output when upsampling.
Seems like the receiving player (Squeezelite in this case) is not
informed that the stream has been upsampled (or changed in bit depth):
it always try to open it at 44100/16.
Same happen sending a <> 44100/24 Wav file via the "wav pcm * * -" line
from a win32 PC, but from macOs, Linux and win64 works as espected.
So, just to understand and in case stop loosing my mind:
Is LMS supposed to be able to play Wav Files with sample rate > 44100 or
bit depth > 16 ? From all platforms?
Is LMS supposed to send to players raw pcm streams (produced i.e. with
sox) with sample rate > 44100 or bit depth > 16 ?
If so, could you please provvide me with a working example (i.e
upsampling a wav file without converting it to flac)?
About starting the server from the preference pane.
Do you need some more informations?
Have a nice Easter and enjoy holidays!
Marco
____________________________________________________________
SB+, Klimo Merlino + Kent Gold, Monitor Audio Studio 20 Gold SE+, Klimo
reference and DIS Interconnect.
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