I think I understand more clearly now. Quality wise I should be more concerned with sample rate and bit depth, when it comes to bit rate this is essentially the factor of lossless flac compression, like a zip file for example.
Correct. The most 'damage' to digital audio is usually done when converting/resampling the original bit depth or sample rate.
So you are wise to keep an eye on it if those are changed, either at converting, or at playing back. (which by the way can easily happen on a Windows PC depending on your audio and soundcard settings)
If you want to learn more about that, search for terms such as bit-perfect, asio, wasapi, exclusive mode.
And back to the question as your header states:
Just leave flac at -5. Or possibly set it to 0 or -1 when doing bulk encoding. This will cut the time required for encoding more than in half, with only a very minor effect on the file size.
In all cases, 16bits audio will remain 16bits, 24bits will remain 24bits, etc.