Thank you! So, this means, you divide each sample value on a channel by (1 / 0.177828)?
No, you are taking my approximate a description a bit too literally.

The sample pre-amp is just one out of many factors leading to the final total amplification amount of every channel, and it's user-configurable. In XM by default it's 48 (found on the General Tab) but it can be changed. The pre-amp is stored as an OpenMPT extension in XM files so other players will ignore it completely and just use whatever pre-amp they want. XMPlay normalizes modules for example, so the pre-amp is calculated while the file is loading. For a tracker this is not a viable approach, though, it makes only sense for players.
And does it make a difference for the sound scaling the sample values on each channel while mixing or scaling the final sample value after mixing?
It depends. If you literally want to add a separate amplification for the scaling to each channel, then latter is less computationally expensive and introduces less floating-point requantization errors than doing it individually for each sample (every multiplication adds more errors to the final result, so you want to keep the number of operations that require requantization down). However, you can of course just use the same "trick" (really shouldn't be called a trick as it's a trivial optimization that you will find in any good mixer) as OpenMPT and have a single amplification factor that is applied to each mixed sample that itself is a multiplication of actual note volume, panning and the final scaling. This would end up with a single multiplication per audio channel per sample.
So if your mixer currently looks like this:
// Mixer
v = sample_value;
v *= note_volume;
v *= panning_left;
v *= global_scaling_factor;
out_left = v;
// same repeated for right channel
Then it would make more sense to optimize it to this:
// Somewhere outside of the mixer where the volume of each channel is calculated
volume_factor_left = note_volume * panning_left * global_scaling_factor;
volume_factor_right = note_volume * panning_right * global_scaling_factor;
...
// Mixer
out_left = sample_value * volume_factor_left;
out_right = sample_value * volume_factor_right;