can't really answer your question with the information at hand, but I can give you some ideas...
Back in the mid 1990s (I think) Louis Fielder, one of the engineers at Dolby Labs, wrote a landmark paper titled something like Dynamic Range in Digital Systems. In this paper he suggested that the proper reproduction of a musical performance could require as much as 120 dB of dynamic range. That's a BIG number, a ratio of 1,000,000,000,000:1 (yes, that's 10^12 - feel free to check my math!)
That is theoretically possible, and while some chips claim that dynamic range, it is more of a marketing number than a real number.
Let's think about that for a minute... a really quiet recording space would be built to have a noise floor on the order of NC20, or about 20 dB-SPL broadband, average. And it is generally accepted that the threshold of pain occurs somewhere around 130 dB-SPL (depending on age, physical condition, and the content of the noise.) So it would seem that we are already in trouble<G>!
So what's practical? Or realistic?
You have to start with the room! How quiet is it in your room. You'll need to borrow (or buy) a real sound pressure meter if you want to know the answer, but you can get by with your recording gear if all you really care about is the S/N ratio in your recording.
But whatever you end up with is, well, what you end up with. The noise in the room is the noise floor, it's only going to get worse as you pass it through transducers and electronics, and especially modern (read low power) A/D converters. For sake of argument, lets say that the average SPL in your recording space, across the entire audio band, is around 30 dB-SPL. So even if you could hit (and survive?) 130 dB-SPL you'd still only have 100 dB of dynamic range... IF none of the subsequent stages distorted or created noise of their own... and they all add noise!
Practical... gotta remember that word!
I usually monitor so that the wide band, average level (slow response) is around 85 dB-SPL. Remember, this is what is delivered to my ears, this still has nothing to do with the electronics, that's a whole 'nother matter! And we, by which I mean "I" keep throwing that word average in...
It turns out that the human ear is sensitive to both average and peak levels. And that is one of the places where people tend to misunderstand poor Louis's assertion. He was thinking about the whole kettle of fish, and especially peaks. Peaks take on a couple of forms in music. There is a peak when you strike a snare drum that is very short in duration, but depending on the drummer and the drum, it can be REALLY LOUD for a fraction of a second.
And that' what Fielder was trying to preserve. Well that, and the natural "crest factor" of complex waveforms. And that's what you need to consider as you set up your system.
We really don't know much about the energy in the noise floor you are measuring in Sonar. We don't know who the contributors are, or what they contribute, but since you probably can't do much about it, and 90 dB down for an average noise level is quite decent really, let's forget about it, and worry instead about the upper end.
You probably know that you can not exceed 0 dBFS - which is the maximum level that the hardware and/or software can represent. What you may not know is that level is going to be a different physical level at different places in the system, and that can be a problem. You should also keep in mind that ALL dB measurements represent power ratios, even though we use them to represent sound pressure levels and voltages and whatever. And this means that they are, by definition, average or RMS (root-mean-squared) measurements. They are not peak measurements, and they can't be peak measurements.
The trick then, is to find a spot somewhere below 0 dBFS where you can operate safely, and by safely I mean:
1) you never exceed 0 dBFS for more than a couple of samples
2) you don't clip analog stages before or after the computer.
Some folks think that you should treat a digital system the same way we treated analog tape. That's not a good idea. With analog tape we had a huge noise contribution from the tape, and probably the tape electronics. So we tried to operate as close to 0 VU (an entirely different measurement system) as we could. We don't need to do that any more. Digital storage and transmission do not add noise to the signal, only the channel, and the channel gets tossed away!
So unless you are recording the song of the Egyptian Fruit Fly (and since you mentioned amplifier hiss you probably aren't), the key here is to find a level in the digital domain that works for you. You need some headroom to accommodate peaks, and you want to stay as far away from the noise floor as possible... without exceeding 0 dBFS or clipping.
Sensing a pattern here<G>...
I tossed a fair bit of information that you may or may not need, but I hope it demonstrates the futility of sweating S/N ratios, or even dynamic range, in a modern recording environment. The bottom line is that the noise that is inherent to your recording chain is going to be there. If you record your tracks at a level significantly higher than the noise floor then the noise will still be there, but listeners probably won't hear it.
One last thought - a lot of this is pschological, or psycho-acoustic. When I was still working with tape we would use a trick called spot-erasing, where we would erase the noise in between the notes - ok, seldom got that silly, but if the guitar amp was humming a bit we'd erase the tape if the guitar player wasn't playing. It tricked the mind into thinking that the dynamic range of the PERFORMANCE was greater than it really was. (When inexpensive automation became popular we'd automate the mutes to do the same thing.)
Have fun... and don't spend too much time worrying about S/N ratio... or reading this post for that matter!