Ron F. wrote: 
> My comment here could be construed as naive, but assuming the question
> is bit-perfect output from a Squeezelite instance running on RPi-OS,
> (output samples to stdout using -o -) then what happens if in LMS, the
> player's audio volume control setting option is fixed to 100% ??? I
> imagine the OP is already doing this and I missed the comment, just
> thought I would ask.
> 
> Edit: ALSA Master volume also set to 100% ?

Not at all. As mentioned volume fixed at 100% with replay gain off. But
here is the thing, if amplitude is changes between two takes then all
audio samples would change for that file unless we were juggling the
volume knob.
Here I get more than 40% of the data correct. And looking at my first
post the spectogram are mostly green but not all green as it should be
as we still are at the pure "-data transport layer-" of the audio with
no intended DSP. 

And given both playback from WAV and FLAC versions are done with the
same settings on both the transmitting side and recorder with the WAV
file comming in as bit perfect nothings changed other than LMS creates
the stream from what the FLAC application sent to 'stdout'. And I
believe I've proven the FLAC configuration on LMS do produce identical
file as the WAV as we already espected.

At this stage I'm looking at this fenomen as corrupted data. But why it
manifest itself mostly in the high frequency band I know to little about
digital audio to form a theory about. If anyone more experienced
programming audio could chime in?

Before stumbling into the code stack (thanks @Roland0) I wonder if I can
trick the convert.conf to sendt WAV to stdout? I'll look into that to
check if similar artefacts are added.


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