oshcar wrote: 
> thankyou for this response. there is alot of good information here. i am
> still processing it. will respond again.

ok

yes -- i want to avoid degradation

i understand the LSB

quote "It doesn't but if you have to perform an operation on 16 bit
sample (e.g. gain for volume adjust) and then any 16 bit operations
usually ends up with a 32 bit answer"
i dont want to perform any software operations
so why do i have to pad to 32 bit ?
even if i set ALSA f at 16 to drop the LSB later on before sending to
dac -- what do i do if playing a 24/96 file ? -- then i have to manually
change f to 24 every time ?

quote "Use ALSA "hw" devices and ALSA will not do any s/w conversions"
it is my understanding that there is more to this driver setting -- for
example generic 5102 = output setting hw:sendhifiberry -- something like
that --- my understanding is that there is more in this driver placed
there by hifiberry developer --- if you clear this field -- default
values are used -- are you saying the default values are doing some kind
of software calculations ?

i have no desire to use sox

quote "So if you have a 16 bit input, use ALSA "hw" and have
analogue/external volume control - then within ALSA 16 bit sample will
be changed to 24 or 32bit by adding LSB zeroes in case there is
processing and those zeroes will be removed when output to DAC - if
there was no "processing", the 16 bit audio sample value will be
unchanged from input."
i am confused by this -- i never get 16bit output unless i force with
f=16 ( Rpi driver ) -- this whole chain of events doesnt seem to be what
is actually going on to me
are you saying the 16bit input is being padded to 32bit by ALSA ( not
squeeze code ) waiting to see if any processing happens ( by sox i
assume ) then ALSA again drops LSB to match the value i put in ALSA
setting f ?

whatever the case is above -- if i know that i wont be doing any
software conversions --- why do i have to go through all this ? -- this
goes back to my original post --- 16bit in = 16bit out , 24bit in =
24bit out

Also -- if i pad a 16bit input to 24bit and send that to a dac that
accepts 24bit -- have i increased the dynamic range ?


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