So here's something I just noticed recently when recording a podcast.  Although I have direct input monitoring through my interface (Saffire Pro 40), I can't get it to kick up much volume, so I always enable Input Echo on the armed track.  I set my buffer settings to 64 samples whenever I record, so while that obviously introduces a bit of latency/slapback echo, it's barely noticeable.
 
When enabling input echo and monitoring through headphones, my voice had this weird splashy effect, almost as if I was sending it to a harsh spring reverb or something.  The recorded audio sounded just fine so it's just a monitoring thing, but it was really distracting.
 
This is the first podcast I've recorded, and whenever I've recorded vocals in the past, it has always been with a cue mix going into my headphones in addition to the input echo of my armed vocal tracks, PLUS I typically send the track to a reverb and monitor that as well.  SO, it could very well be that input echo has always had this type of behavior and I only noticed now that I'm just monitoring dry vocals.
 
However, when setting up mic placement to record acoustic guitar a few days later, I noticed the same thing. I've set up acoustic guitar stuff before and didn't notice it then.  Of course, maybe I'm just listening for it now.
 
Anyway, not a deal breaker since the recorded audio sounds fine, just wondering if I'm off my rocker or not. :)

Disable your "Direct Monitoring" if you are monitoring the track/DAW.
 
Input Echo allows you to monitor the live track input/output from Sonar.
 
Direct Monitoring features on the hardware allows you to hear the input signal BEFORE it hits the DAW software/track.
 
If you have them both going at the same time you will hear both signals at the same time causing that "delay"because the signal from the track will be slightly delayed due to latency between the hardware and the DAW.
 
Cheers.

To illustrate this...
 
Direct "zero latency" monitoring...
 
Input > Interface > Headphones/Monitors (a direct, unprocessed signal that doesn't go through the DAW/Sonar so the signal hits your ears very quickly)
 
Input Echo from a track in Sonar...
 
Input > Interface > Sonar Track (input/output) > Sonar Main Output > Interface > Headphones/Monitors (this takes a little longer for the signal to return to your ears... especially as you add any effects. This is what's referred to as latency).
 
You put those two signals together the second signal (from Sonar) will be delayed because it has farther to travel and is being manipulated by more digital crud. This is usually measured in milliseconds so instead of an "echo" effect (unless the latency is REALLY bad) it just sounds weird and "phase-ey".
 
Muting one or the other will remedy the problem.
 
However sometimes with certain routing schemes it simply needs to be tolerated while recording but does not appear on the final track. This does not sound like one of those situations though. Just disabling the Direct Monitoring OR the Input Echo on the target track will remedy the issue.
 
Cheers.

aaaand just in case this is the problem/confusion...
 
You don't actually need Input Echo enabled to record into an audio track. That is only there so you can hear what you are playing as you play it. If you don't need to hear the input signal or you are using direct monitoring from another source like your interface/mixer or otherwise then you can leave that disabled and the signal will still get recorded.
 
With MIDI tracks though, if you are using an external controller, it DOES need to be enabled to allow the MIDI signals through. That's called MIDI Echo.
 
Confusing? You betchyernutz it is. lulz

Perhaps I should've been a bit more clear.  I understand what latency is and the difference between input monitoring through an interface/mixer and input echo monitoring through the DAW.  As I mentioned, when I track at a 64-sample buffer setting, I'm not bothered by the slight latency/echo that exists if I'm monitoring directly through the interface and the software at the same time.  I monitor through effects whenever I track vocals anyway, so a slight delay is actually desirable for me. (But in this case I'm not monitoring through any effects.)
 
But the problem I'm trying to describe isn't echo, it's an ugly, metallic sound, as if I've got a bad spring reverb effect on the vocal, or if I'm singing inside of a coffee can.  It's not exactly "phasey" sounding, although you're right, phasing will occur whenever the same signal is replicated and offset, which IS what's happening when I direct monitor and input echo simultaneously.
 
The thing is, I can barely hear myself through the direct monitoring.  Input echo is where I get the real volume, and again, what I'm hearing doesn't sound like typical phasing.  And I've never noticed it before...all I've noticed in the past is a slight echo as a result of the latency, not this ugly, splashy sound.  There again, I usually monitor through a fairly aggressive reverb effect so that would likely mask such anomalies anyway.
 
Since I can barely hear the direct monitor signal anyway I'll go with your suggestion and disable it so I'm ONLY monitoring through the DAW via input echo and see how that works.  Thanks for the feedback; I'll report back after I have a chance to try it out.  

Yup... check it out. I know the sound you are referring to and I am quite sure this is due to having both DI monitoring and Input Echo going at the same time.
 
There ARE other possible causes though but to determine those causes the DI/Input Echo thing needs to be eliminated.
 
If that does not work then I would say perhaps there is a "driver mode" issue/conflict (I've had that weirdo sound incurred when mismatching drivers or simply choosing the wrong driver). Are you using ASIO or another driver mode? Are you using any multiple interfaces (such as a USB mic and your interface/soundcard at the same time)? Are you using internal routing programs such as Voicemeeter?
 
Don't worry about those questions until you test disabling the DIM or IE stuff. Just posting it preemptively.
 
As far as your DIM being too quiet issue... this of course can be dealt with by adjusting your levels. Instead of trying to make your direct monitor signal LOUDER try turning down the other signals. If this affects the signal that you want sent/tracked/etc then you can create a sub bus (or headphone mix/bus) using sends or maybe your interface has options for creating headphone submixes. That can be complicated stuff and I personally can't help with that without having my hands on the rig/seeing what's up but generally totally doable.
 
So... send everything EXCEPT your voice to a single bus (maybe separate from your Master depending on the situation), turn down the bus until it levels out with your direct input.
 
Problem is DIM signals are raw and unaffected so you try to compete with a bunch of compressed/limited/effected signals it might get overpowered (which seems to be what's happening). You turn down those signals en masse then the signal can cut better. If you can figure out the routing you could even sidechain a compressor to "duck" the signal when you start talking (like radio DJ's do).
 
 
 
 
BUT... that is a whole pile of wackiness that need not be worried about until you check out the dual signal thing. All that REALLY matters in the end is what the end listener hears. Everything else is comfort for the artists/speaker as they do their thing. If you can handle just listening to your voice through your skull (no input echo or dim at all) and can mix it in post production then you don't gotta worry about it at all.
 
Cheers.

I'm using the Focusrite ASIO drivers, and sometimes I've got an Eleven Rack online (I do have to change the sync on my Saffire to SPIDF whenever using the 11R otherwise I get crackles if the 11R is a slave), but aside from that it's always set to internal.  The Saffire is my only audio interface, although the 11R can function as one, but I never use it that way.  It's only there for recording, and it's never turned on when I'm tracking vocals anyway.  No USB mics or internal routing software, expect of course the Focusrite mixing software for my interface.  I've double-checked all of my settings and nothing appears to be out of place, so I'll give your recommendation a try first.
 
As far as the DM volume goes, I've already made adjustments within the Focusrite software to bring it up as much as possible, but as you mentioned, when I'm tracking against a full band arrangement, I can't hear myself at all unless I enable input echo, or bring everything else WAY down.  Thing is, even with everything turned off and I'm just tracking a dry vocal, I can still barely hear my own voice through DM, which is why I enabled input echo during the recent podcast project and came across this nasty sound.
 

Tried a couple different things:
 
1) Enabled ONLY direct monitoring through the Saffire Pro 40.  Very dry vocal sound in the headphones, pretty low volume (had to adjust the backing tracks output so I could crank the headphone output), but a clean sound.
 
2) Enabled ONLY input monitoring through SONAR. Latency not noticeable at 64 samples (not to me, at least), levels were much better, and the "springy" sound was a little less pronounced.  However, it still had a "processed" sound to it, not the clean, dry sound I got through direct monitoring. (I should note that I didn't have any console emulators or other effects active when monitoring/tracking.)
 
Ultimately, the recorded audio sounded just fine, no matter how I monitored it, so in the end it doesn't really matter. Disabling either DM or Input Echo did improve matters, as opposed to having them both enabled, but it still doesn't sound quite right going straight through SONAR. However, since MOST of the time I'm monitoring through a reverb anyway, I suppose it's a moot point.  If I start noticing slushy recorded audio, then I guess it's time to tear down the house and build it back up!

Yeah, if you have a reverb or any effects in the signal chain at all (like maybe you've got bus effects and the voice is going through that) that will all effect the sound unless you go out of your way to avoid it.
 
Think of your DIM as pluging into a mixer with no effects or anything and listening to your voice coming back to you completley pure and unaffected.
 
If you go through Sonar absolutely anything in the chain (from the interface input back to the interface output) will alter the signal. So if you've got you voice going to the same bus (like the master) as everything else and that bus has any effects on it (limiter/compression/etc) then it will affect your voice as well.
 
What you can do to get the purest signal possible is to switch the track you are recording your voice on to output directly to an output on your interface that leads to your headphones.
 
But without any compression or anything you may find that level too low just like the DIM signal. BUT you could apply specific compression directly to the track to alleviate that while not screwing around with any effects on your downstream busses where other material is going to.
 
Ugh... I hope that makes sense.
 
No sleep make Beepster go something something...

No sleep make Beepster go completely nonsensical.
 
Actually, not. You're making perfect sense.
 
As I stated before, I'm not running any effects on anything when monitoring through SONAR. No master buss effects, no emulations, and in this case, no reverb. I kept everything absolutely clean to try to determine where the splashy sound is coming from.
 
I've also routed the signal directly to my headphones out instead of going through the master bus just to see if there's something happening in the signal chain that I didn't miss, but it's the same thing.
 
No matter. The recorded audio sounds fine, which is what's most important.