gswitz
Yes.. The track gain is set to a useful range rather than complete range.
Clip gain is automateable. Not sure how low that goes. I've never pushed lower bounds on these parameters.
You can mute tracks and clips.

Its not that I ever need to go that far down in level when using the gain knob. I just thought to myself the other night, what if I bring the knob level all the way down? When I did, I noticed the audio was still there. It slightly confused me... But with this post and everyone's responses, I think we have cleared the confusion. 
 
Thanks again all!!!

In the old days you used the gain to "set" the levels so the analog fader had the most resolution.  That is toward 0 dB - you get large changes in vol toward the bottom of the fader travel with small movements.  At fader unity (0 dB) you get the most control, since fader movement changes the vol less. It was ergonomics and control, like tight steering in a car.
 
And a compressor operates completely different than normalizing.  Normalizing raises the entire track by the same value.  If you have a -6 dB peak but most of your track cruises along at -12 dB (average), normalizing the track to -3 dB only raises the average level by +3, so your average peak is at -9 dB.  The internal ratios and the average vol remain in lock step.  If you use a 2x1 ratio in a compressor, to raise the peak level to -3 you would need +6 dB of gain, while your average vol peak for the track will go up by the +6 dB (more or less), thus squishing the level difference between the loudest and softest vol on the track.  This is why when you slam a signal through a comp/limiter it looks like a block of sound, instead of peaks and waves.  They have been compressed together.  It has a lower peak level although the average sound level is higher (so it sounds louder, usually).  Then you can raise the overall level via the output gain and still not hit the red with random peaks since they have been, relatively, shaved down to humps. Depending on the other settings on the compressor, it can sound a lot like normalizing, or not, even tho both are louder.
 
@

On an analog mixer, you first put the Volume fader to the 0dB mark, then have the singer sing into the mic whilst you set the gain knob so that the VU meter averages at around 0dB. The preamp behind the gain knob is a very high quality, low noise design. You want it to do the heavy lifting.
 
Once all singers/artists have their channels set, ideally you now have the same volume when the faders are at the same position. Gain does not get touched again, except to correct for the inevitable singer that sings much softer when testing.
 
During the show/recording, you should now only need to move the level faders. And they are for fading, not boosting. Analog desks have a more simple design for the level fader electronics, so you don't want to have to go higher than 0dB because then it starts to add gain.
 
Sonar simulates this conceptual model with the difference that instead of 0dB, in the digital world you average at around -12dB, which is why the LANDR article speaks of working around -10dB to -18dB

jpetersen
On an analog mixer, you first put the Volume fader to the 0dB mark, then have the singer sing into the mic whilst you set the gain knob so that the VU meter averages at around 0dB. The preamp behind the gain knob is a very high quality, low noise design. You want it to do the heavy lifting.

Like all rules though there are exceptions - e.g. when e.g. a processor or tape recorder is patched to the channel inserts. Then the initial gain also needs to be set with a view to sending whatever's in the insert the right signal level for it to operate as desired.

jpetersen
Once all singers/artists have their channels set, ideally you now have the same volume when the faders are at the same position. Gain does not get touched again, except to correct for the inevitable singer that sings much softer when testing.

It's not just singers. A surprising number and variety of musicians have the (bad) habit of playing tentatively and quietly while sound-checking then letting rip once the gig starts. This includes guitarists and bassists who turn their amps up as they walk on stage and keyboard players who won't leave their volume controls alone. Or, worse, use several patches during a performance but have never bothered to make sure those patches are all at roughly the same volume, or that their pads or basses don't drown out their leads.

It's just one of those things that makes live sound so......"interesting".

(edited to sort out originally botched formating)

AT
Best practice is to capture as close to how you want it to sound, at a good level.  That makes everything else easy and leaves "mixing" to adjusting levels and pan and reverb and other such ear candy.
 
If the track's level is too low, for whatever reason, either the gain knob or normalizing can be used to raise it higher.  Either of those have the same effect, raising the entire signal, peaks and noise.  Compressing, well, compresses the internal dynamics of a sound, squishing the differences between the loudest signal let through and the softest part, including noise, esp. if you have to raise the output level on the comp.  Compression evens out the signal differences between the loudest and softest parts of said signal.
 
Normalizing isn't supposed to be a normal, everyday tool to make up for bad tracking techniques, but a tool to fix problem tracks.  That said, if your process involves normalizing everything to make mixing easier since every track as loud (lead guitar, kick and triangle), that is fine, but seems to me the long way around and is a way to introduce unnecessary noise.
 
However, gain staging is mostly an analog problem in driving a signal hard enough to overcome inherent noise without saturating or distorting.  Once it is in the computer you can raise a signal or normalize or compress it w/o needing to introduce any artifacts or adding noise (you'll just be raising up whatever is already there).  For me, if the captured signal needs more level to work into a mix, I'll use gain for a little bump, but normalize when I need more than a couple of dBs of gain.
 
@

This is the analogue approach and does not work for the digital paradigm.  Yes get your levels right before going into the box, however...if you aren't recording analogue, and here is the rub...you need to be aware of both ways depending on your practice.  Aim for -6db RMS on the Sonar meters for analogue recordings, but don't set and leave...just like a cake or a painting, think about the layers.  So for some rock, the vox once it's inside the box should be 0 VU say on the console emulator VU.  Same with drums, but the bass could be -6 on the VU meter.  And if you set and leave, you can't set a mix up like this.  This is what the gain pot is for in Sonar, Sonar is the best mixer in DAW land because of this.
 
If your not sending the audio out of the box after it's been recorded or if its totally electronic, think about the cake and use the gain pots to set a whole mix in the above way.  And here is the important step, don't worry about overs...it does not matter about clipping once your in the box, because of 64FP.  And this is how you make a digital mix sound analogue-fat and warm .  Just ask if you don't understand. 
 
All my work is on my websites, which are in my signature.  I believe my work speaks for itself now.
 
Peace Ben :)
 

Really its not a good idea to clip any audio even with a FP audio engine. True the routing within wont have a problem  and with FP there is no true clipping but some plugins can't handle overly hot signals those above 0dB. Besides there is no logical reason to allow audio to clip.
 
A technique I rely on is before I start to mix I lower all faders to infinity. I then raise one at a time to set a balance. If I find I'm in the red on a track to get it in balance I go back and lower all the others. Remember faders go both ways.    

John
Really its not a good idea to clip any audio even with a FP audio engine. True the routing within wont have a problem  and with FP there is no true clipping but some plugins can't handle overly hot signals those above 0dB. Besides there is no logical reason to allow audio to clip.
 
A technique I rely on is before I start to mix I lower all faders to infinity. I then raise one at a time to set a balance. If I find I'm in the red on a track to get it in balance I go back and lower all the others. Remember faders go both ways.    

No and take this the wrong way John, but this is old school thinking.  And I'm not talking about constantly in the red either, just don't worry about clipping...just like on a analogue desk.
 
I do the opposite to John, I leave all my faders at zero, mix all the instruments e.g. console emulator, EQ, compression, then balance the mix to fit the quietest track.  This generally means the master bus is over by 3 or 4 db, I then use the gain pot with Linear Phase EQ, then Tape Sim...to give me an average of -3db RMS headroom for mastering.  This is how you use the gain pot. IMHO.
 
But John is right, and it can work this way...although I'm highly doubtful you will get the same fullness of mix this way.  And it is how an analogue engineer would mix in the digital realm.  Whereas I'm a digital engineer, using analogue mixing techniques.  
 
Now there will be some huffing and puffing about this concept, this is analogue engineer vs digital engineer, but as a digital aesthetics expert now...and I will get around to writing a paper and publishing this at some point, there is a definite difference between the two paradigms.  Again the proof is in the pudding, all my music is in the below links.
 
Cheers Ben 

BenMMusTech

John
Really its not a good idea to clip any audio even with a FP audio engine. True the routing within wont have a problem  and with FP there is no true clipping but some plugins can't handle overly hot signals those above 0dB. Besides there is no logical reason to allow audio to clip.
 
A technique I rely on is before I start to mix I lower all faders to infinity. I then raise one at a time to set a balance. If I find I'm in the red on a track to get it in balance I go back and lower all the others. Remember faders go both ways.    

No and take this the wrong way John, but this is old school thinking.  And I'm not talking about constantly in the red either, just don't worry about clipping...just like on a analogue desk.
 
I do the opposite to John, I leave all my faders at zero, mix all the instruments e.g. console emulator, EQ, compression, then balance the mix to fit the quietest track.  This generally means the master bus is over by 3 or 4 db, I then use the gain pot with Linear Phase EQ, then Tape Sim...to give me an average of -3db RMS headroom for mastering.  This is how you use the gain pot. IMHO.
 
But John is right, and it can work this way...although I'm highly doubtful you will get the same fullness of mix this way.  And it is how an analogue engineer would mix in the digital realm.  Whereas I'm a digital engineer, using analogue mixing techniques.  
 
Now there will be some huffing and puffing about this concept, this is analogue engineer vs digital engineer, but as a digital aesthetics expert now...and I will get around to writing a paper and publishing this at some point, there is a definite difference between the two paradigms.  Again the proof is in the pudding, all my music is in the below links.
 
Cheers Ben 

Actually its not analog at all. Its wisdom from CW. Noel made the same statement and for the same reason.
 
The reason I threw in the mixing procedure was to emphasize the idea of controlling the level from the beginning. It also lets one hear what each track is doing. What it may need and how it might sit in the mix. 
 
 If you think of a mixing console as a pallet you don't mix all your colors all at once. All you get by doing that is mud. 

So for the guys who normalize your tracks how are you doing this?

IF YOU are recording in 24 bit,
there is no reason to normalize.
mix with tons of headroom.