If I keep my recording chain lossless for instance..
Record/CD/Dat/R2R -> Deck with Analogue or Digital Output -> Computer with Software Set to capture in PCM/Wav/AIFF/DSD/Other non lossy -> Converted to Flac in *Any* software that offers that as a format...
It will be lossless. Has to be.
I say *any* software that offers Flac because they all pass the process off to Flac.exe/dll (or at least the very exact same conversion Math) in one way or another - they have too or it couldn't be a classed as Flac and could lead to compatibility issues...
But If I stick crap in I'll get crap out. lossy in - lossy frozen in time in a lossless shell out - but still lossy.
A little side note about Opus codec, Opus is a great codec, technically incredible - it's lossy but has full 20-20khz stereo bandwidth at 128kbps but the thing with it is if you convert a 128kbps opus file to wav or flac it 'can' fool checking software into thinking its true lossless. This is why spectrograms are useful tools as they tell you so much about the waveform, Opus has a tell tale 20khz Hard cut off, high kbps Mp3 shelves at 16khz and 18khz
Some official lossless releases are trash, somewhere around the early 2000s when digital remastering and digital brick walling became a thing - sometimes they were from lossy archive masters or the remastering process had a lossy link in the chain.. Classical CD's were pretty bad and anything that was remastered from the 50s etc..
Now as for recording volume, as long as you aren't clipping the signal, record! Once its done find the normalize function and run that over the file- what that does is find the loudest peak and set that as 0db reference (or what ever you pick as reference -0.2db is ideal) and raise the rest of the track up to match. You'll get the same result as what you would if you had recorded it louder, It won't introduce crap because we are in the digital realm here - it wont add anything that isn't already there. But use Normalize, not set gain or anything like that Normalise retains the integrity of the Volume throughout where as othere methods can cut off anything over 0db killing transients and effectively brick walling the file..
If your recording chain is 32bit (unlikely right now but maybe one day soon) you can even record totally clipped and normalise back down to 0db and not lose any transients - it's amazing to see!
Little observation - There seems to be a mistrust of noise reduction - it's a shame because this especially has been solved from a technical standpoint for years now. Every movie/Tv show and every individual sound in those, every song in recent years, every voice recording have all been treated and you'll have not known. Its really that good now. Its just the operator using the tools that can go overboard or not understand the operation correctly that can mess it up.
Sorry for the ramble - too much coffee today