If the limiter in any recording device is operating in the digital domain in almost any way, there's not much point in using it unless you do not intend to post-process the recording (in other words, if you are simply going to listen to it as is, then there's a point, but as soon as you pass it through a DAW, then there's no point).
For any 'dual path' system to be worthwhile to someone intending to post-process the recording, then it would have to operate in the analog domain, which would mean that you'd have to have two preamps per channel, one working with less gain than the other. Then if the preamp with the higher gain clipped, you could use the one with the lower gain. I believe this is done with some systems but only for mono recording - you have the right channel operating at (say) 12dB less gain than the left channel, so you have something to fall back on if the left channel clips.
In the R-44 (for comparison) the limiter works by the lowering the preamp gain by 12dB (whenever the limiter is switched on) and then in the digital domain, digital limiting is applied followed by 12dB of digital gain. There is absolutely no logical difference between this and simply setting the analog gain 12dB down manually yourself, and then applying a VST limiter in your DAW afterwards with 12dB of makeup gain.
Bear in mind that once the signal has been digitised in the recorder it's simple for the processing to look ahead at the data and limit the level entirely seamlessly. Limiting in the analog domain is that much harder to achieve (perfectly) because the analog circuitry typically doesn't "know" that a peak has been encountered until after the fact.
I don't want to restart the endless arguments about the merits of 16 bits vs 24, but it's generally agreed that 'under-recording' using 24 bits, then normalising later, means that any need to limit or otherwise compress normal musical signals toi avoid analog clipping goes out the window, due to the very large dynamic range in the digital domain that 24 bits provides.