SPDIF vs Analog....
Im going to give a short very basic overview of the signal flow of the Kemper and some ideal monitoring situations for beginners.
This will be very basic, and there will be points to make arguments for and against some points. It is ideally for beginners, so if you want an "in depth discussion" / argument about anything I say, go to gearslutz or something 
------------------------------------------------------------------------------------------------------------------------------
In theory at least, SPDIF should be the superior sound, as it is passing the audio through 1 less conversion process.
(in laymans terms):
SPDIF: Guitar -> digital conversion within Kemper -> output as digital via SPDIF into soundcard -> soundcard sends digital signal to DAW for mixing (the incoming signal is already digital, so no conversion needed before sending the signal to the DAW)
Analog: Guitar -> digital conversion within Kemper -> conversion to analog at Kemper output stage -> soundcard converts the incoming analog signal to digital -> soundcard sends digital signal to DAW for mixing
Generally speaking, whenever a signal gets converted to or from analog, you lose a tiny bit of quality. In other words, the less conversion processes you have in your signal chain, the better.
Now, the thing is, the human ear cannot really hear the tiny audio loss from the sound conversion process of analog to digital (or vice versa) in most situations.
The only place you MIGHT hear a difference is in a really well sonically-designed room with high-end speakers and in a perfect listening environment, which very few users have....and even then, any noticed difference could be subjective in a psycho-acoustically sense (read: all in the users head!).
So generally speaking, digital vs analog should not be noticeable. Any difference is 99% likely down to human error, ie, a setting accidentally left on, or a plugin one thought was disabled...
------------------------------------------------------------------------------------------------------------------------------
Now onto monitoring:
When you load up your Kemper and connect via your DAW, with a lot of soundcards there is a front-end monitoring panel, that is almost DAW like in itself! It can get a bit confusing, but think of it as a latency-free monitoring tool for now. Then you have your DAW, and finally you have your soundcard again before it is sent to headphones / speakers.
So it looks like this:
Audio into soundcard via SPDIF from Kemper -> "Latency-Free Monitoring" (LFM) Software -> Soundcard -> heaphones / speakers
................................................................................L-> DAW (Logic, ProTools, Cusbase, etc) -> soundcard -> heaphones (HP) / speakers
So the signal here is sent to BOTH the LFM and DAW software respectively AT THE SAME TIME. (the "L->" is a mis-aligned break-out arrow)
(What I mean by "Latency-Free Monitoring" software (LFM) is for example, RME Totalmix for RME hardware, UAD Console for UAD Appollo, Maestro 2 for Apogee, Mbox for Mboxes, MREditor for MR816X/CSX, QUAD Control Panel for Roland Quad Capture, etc etc...)
Now, the signal passing through your LFM and DAW software do not hit the soundcard at the same time. Often, the DAW is slower due to audio buffers etc that the DAW populates first, amongst other things. If you put plugins on your track in your DAW, this further slows down the signal getting to your soundcard due to the extra processing of the plugins. For now, lets presume that we are not mixing and have no plugins on the DAW track.... it is just setup so we can record our takes (as it ideally should be, though perhaps bar a level monitoring plugin or something).
Due to the time difference of the audio getting to the soundcard via the LFM and DAW routes, the combined signal output to your HP will sound different than the sound coming straight out of the HP jack on your Kemper, for example. The 2 signals combine at the soundcard at different points in the original wave form that was first sent from the Kemper to the soundcard, and cause "phasing" to occur (aka "comb filtering" to some)
For more info on phasing, this explains it quite nicely:
http://www.soundonsound.com/so…articles/qanda-0613-3.htm
Note that if you record in your DAW anyways, and then listen back to the recorded audio that passed through your DAW, you will not actually hear the phasing sound that you heard through your HP while you were recording. The DAW has no knowledge of the LFM audio stream, so only the DAW audio stream was actually recorded. The phasing wont actually be printed within your DAW and you wont lose that awesome solo you just played! 
So what can we do about it? How do we stop it?
Simple...
Simply go into your LFM software OR your DAW and mute the audio stream being heard. This will stop that audio stream being sent to the soundcard (and out to your HP / speakers) and what you hear in your cans should be what you get out of the Kemper.
Which one is best? Well, there are Pro's and Con's for both.
Personally when I record guitars, I usually mute the LFM as I dont monitor with latency-free effects, that a lot of LFM software allow. Since Im tracking guitars, I usually already have the drums and bass tracks in my DAW to play along with, so for me it makes it the best option.
When I record vocals, I do the opposite...I mute the DAW stream and monitor via the LFM as I like to add some reverb to my headphone mix. This does not mute any other tracks from the DAW...all you do is, in your DAW setup a new track, arm it, and mute the output of that track within the DAW. Presto, you can heard your LFM headphone mix with your monitoring reverb, AND the instrument tracks from your DAW at the same time while recording!
Just dont forget to unmute it when you're mixing .... (you'd be surprised...you really would! 
)
------------------------------------------------------------------------------------------------------------------------------
I HTH some folks with basic signal flow and setting up the ideal monitor mix for you.
to the OP, have a look at the "Space" section of the headphones. Ive noticed that if this is on, it can be imprinted into the SPDIF output going into your daw. Possibly also your analog too, but I connect via SPDIF and not analog so Im not sure about that. Either way, try turning it off.
HTH
