I'm telling you from experience.
I used default settings the first time I ever tried it (which included 16 bit). Heard the results and said nope this doesn't work at all.
Noticed the 16 bit. Noticed the output difference business. Did what I said above.
Now I say the short answer is it's a HD to HD virtually lossless conversion.
I checked the 16bit waveform. There were three instances of digital clipping in the whole album. They are so short and far apart that they would not have been noticeable by ear.
But converting the Stravinsky iso again, still using foobar, this time it worked as 88/24. I don't really know why -- I was surprised. But such are the mysteries.. it may be due to the DSDIFF plugin I installed last time (seems unlikely), or to the actual location of the iso and .cue files output from iso2dsd -- I noticed when I moved the iso into a new folder, it no longer converted. Anyway, those clipping I saw before is gone because there's so much more headroom now, but there is actually still an instance , the loudest peak in the whole album, where it looks as if the original digital file is slightly overloaded, either because of how it the ADC was done, or something done during mastering. The peak is slightly 'flattened' even though it's well below digital full scale. Again, it would not likely be noticeable in normal listening.
When you load a file into Foobar, you can right-click (control-click) on the file - select 'convert' and then '...'
You'll see an output format tab to click on in the window that opens.
It's a clunky Windows app. You have to poke around for the controls.
I'm familiar with Foobar controls, thanks, it's been my player for many years now. If you're interested in the particulars: the standard foobar (v1.3.10) WAV output conversion bit depth parameter is set to 'auto' by default, which typically converts to 24 bit for a DSD source (as it did in all other cases except the Stravinsky disc). The SACD plugin has its own parameter settings for each mode of output (PCM, DSD, DSD+PCM). For PCM modes the settable parameters are sample rate (44100 is default, 88200 is my choice, though it's overkill), volume (i.e. 'gain compensation', +0 to +6dB increase, I vary this according to +0 peak level), and mode (Multistage (32fp) default is my choice, Multistage (64fp), Direct (32fp, 30 kHz lowpass), Direct (64fp, 30 kHz lowpass), Installable FIR (32p), Installable FIR (64fp)).
Anyway the issue is moot; as I wrote above, while 24 bit it output didn't work before for this disc, it does now, without me having changed the WAV conversion or SACD plugin output parameters at all.