The Analog ALSA audio output is upsampling all my 44 Khz FLACs to 48 Khz by default, and messing up the sound quality.
Is this a bug?
Is there a way to change this?
These are the sound parameters when music is playing:
~$ cat /proc/asound/card5/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 1024
buffer_size: 16384
and this is what I was expecting to have (and what I have in Volumio player for example):
~$ cat /proc/asound/card5/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 5513
buffer_size: 22050
if I change the output on Kodi from Analog to SPDIF (which doesnt make much sense on stereo), I do get 44 Khz but this time the format is altered from S16_LE to S24_3LE (?!?), so I don’t have my bit perfect stream anymore, and I can hear the differences on the speakers:
access: RW_INTERLEAVED
format: S24_3LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 2205
buffer_size: 8820
The right way to play 44 Khz FLACs should be with:
access: RW_INTERLEAVED
format: S16_LE
rate: 44100 (44100/1)
And this I can’t have with Kodi. 
Help please!!!!