Lets all TEST our Interface driver for offset

Note added May 15 2016- Sorry my pictures seem to have gone offline So original I stored in Photobucket is gone... I'll just put a new one I made here for now, but it is not the same so my text might not be correct.
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This sort of started with another thread where a us1800 owner was not happy about his tracks being around 2,000 samples out of alignment.  I started doing some loop back test of my own. I now believe everyone owes it to them selves to run this test and see if your audio interface drivers are telling the truth to Sonar. 
What I'd like is for different people to post a screen shot of the same test. Maybe we can learn something, maybe this is pointless sound nerd stuff. 
 
Drivers work with Sonar to adjust playback latency so your new audio tracks are perfectly in sync with the original tracks used for playback. It's called offset.  Therefore monitoring at your interface will result in all new tracks lining up as long as you play your part with tight timing. If this sync if off by say 10 - 40 ms you'll be scratching your head and start thinking you suck. I'm good now thanks to my Scarlett, but this was not always so in the past.. read on
 
The loop back test: 
Simply you patch the audio output back to an input and record any super transient track. You zoom way in and see if they are lining up. 
 
I used a midi drum track and froze session drummer. This will create an audio track that should line up to the sample with the midi track.  
Now insert a new audio track and make sure not to turn on input echo, better yet set it's output to "none". 
Select the track the input, I'm using my back panel line inputs therefore the level was at unity. - Arm the track and hit record.  I'm only looking at the starting offset. You are welcome to see what happens after 3 minutes too. 
I'm using 44.1 / 24 bit 
It's hard to read the track titles but the from top to bottom is the midi track, the frozen synth then the looped back tracks using different driver modes if available. 
I first tested my Tascam us1641 on behalf of the us1800 user to see if I got similar results. I tested with the normal buffer setting and then the highest buffer setting. Both were the same being around 400 samples early.
I then tested WDM mode( bottom track) and it was 800 samples late. MME mode wouldn't let me loop back. 

 
I then tested my Scarlett 6i6 and was please to find it was really really right on the money. I tried different buffers as well but that seems to make no difference. It would not run in WDM mode, fine by me. I even zoomed in way closer and could not see any offset. So which interface are you going to use? Right. 

So now I'm real curious about this and fired up my office PC which has this circa 2003 Card Deluxe PCI audio card. My son said it was used in a radio station. Turns out it's better than I realized. I never got into it because like most PCI cards it's bare bones 2x2 1/4" TRS and SPDIF.   They even had W 7 -64 driver for it.  I got the same results in both ASIO and WDM mode. It wouldn't do the loop back in MME either. 

Last but not the least is this cheapo Berhinger USB box that came with the mixer. It has 2x2 RCA. 
I'm using it just fine for live performance playback. It's plug and play and actually works great. Great as long as you don't mind using MME mode with Sonar. I tried ASIO and WDM and it said no can do. 
But as you see it also is late by 400 samples. 

 

If the offset used by RLA is larger than reality, the result will be shifted further back in time than it should be.  Accurate measurement requires that RLA be zero during the test.  To do this, uncheck 'Use ASIO Reported Latency' and make sure 'Manual Offset' is '0'.
 
This will let you determine the actual RTL for comparison with what ASIO reports.

Zeroing RLA they way I described is for measurement/verification.  Not the way to run for production.
 

 
I just tried this out with my Mackie Blackjack.
 
Reported ASIO latency is 864 samples.  To line things up as close as I could, I had to set the offset to 698 (Use reported latency unchecked).
 
For some reason, I'm not able to achieve a perfect match but am either ~3-4 samples early or late by changing the offset amount by 1 sample.  So an offset amount of 1 sample moves the track audio by ~6-8 samples.
 
So it looks like my interface/driver's reported latency is off by 166 samples.

I have a 16X08 due to arrive tomorrow.  I'll have a mini review with some latency numbers in a few days as I get time

Here's a quick one with the 16x08 that got here today.  Orange is the source, below the left channel looped back.  The recorded track is 28 samples early, indicating to me that the reported latency is 28 samples greater than reality.  I'm not gonna futz with this any time soon  :)
 

 
This shows the defaults when I set it to 48K, 64 sample buffer

 

Johnny, I finally got some time to try this.
For the test, I used Sonar Professional 32 bit, AKAI EIE Pro (ASIO) set to 128 sample buffer size.
It looks like the AKAI is coming in 10 samples early. I can live with that.



I wasn't sure what to expect, because this is a relatively inexpensive unit, but I like this thing more every day.

Good stuff Bob. You just made me think of another Nerd test , to record the loop back and flip the phase to see if it nulls? Wouldn't that tell us if the A/D D/A etc were adding colour to the signal? 
 
Are you sure that's only 10ms? It looks like more to me but I guess It depends on how far in you zoom. With the Scarllet no matter how far in I zoomed they still lined up as example. 

My findings where that those setting had no bearing on the test results. I with each device I tested at the lowest buffer and the highest buffer setting and both were the same results. 
I posted a screen shot of my Scarlett settings in # 4 
 
Here's the Tascam

Cactus Music
My findings where that those setting had no bearing on the test results. I with each device I tested at the lowest buffer and the highest buffer setting and both were the same results. 

Copy that. And my last post may be straying from the original premise.
What I was looking for is the Driver Settings screen. This show how the reported latency shown in the Sync and Caching screen is derived.
The balance between Input and Output latency is what has me curious, along with your original question.
Both BobF and myself are showing drivers heavily weighted on output latency.
It makes sense to me, and I wonder how much this balance affects the stability of a certain interface with Sonar.
 
mettelus, cool. Looking forward to seeing the Saffire in action.
 
Kids, you CAN do this at home. Let's see what we've got.
 

With manual offset = 0 and 'Use Reported ASIO Latency' unchecked, you are measuring the full, actual, roundtrip latency.  Those settings are what Sonar uses to adjust (nudge?) your recorded audio to counteract the effect of latency.
 
The reason things line up better when 'Use Reported ASIO Latency' is checked is because Sonar is moving your recorded audio by that amount to compensate by the specified amount.
 
In my example, the reported latency is greater than actual by 28 samples according to measurement.  If I test further and find this to be consistent, I could enter -28 in the manual offset box to correct for this overestimate from the driver.
 

The internal vs external loopback is very likely to be a signal tap prior to the D/A converter, that is the 70-76 sample difference and pretty consistent between tests.
 
What confused me was with the "reported" 319 samples. Since it was off on the first run, I would expect the second to be offset by that amount, but for both tests (internal and external) the offset was 191 samples. Any insight into why this would be? The delta is 128 samples, which also happens to be the buffer that I was using at the time. I am not sure how this all "plays together" now.

FYI - I tried switching to WDM/KS for the Tascam 16x08.  Not good at all.  I can't even get it to do 48K via WDM.
 
 

I've never fully understood WDK mode. I used to think it was installed as an option with your ASIO drivers. Might be true of some drivers but obviously not all of them. The Tascam us1641 seems to let you switch but not my Focusrite. 
My Card Deluxe interface looses the SPDIF option if you switch to WDM mode. ANd you also won't have the control panel. So seems WDM mode is a very pour option as a driver even though Cakewalk recommends we try it if ASIO doesn't work. And asio4all is a WDM wrapper so this would explain why it is a crappy option most times too.
 
I think unless people are willing to run this test when they say things like "I'm using WDM mode with out any problems" I think they are unaware of what might be happening to their tracks syncing up. Some might not even notice. For vocals and lead guitar  2-3 ms offset might not even bother some people. It bothers me. 
I gave up on recording with Cakewalk Home Studio back in 2004-5 because it sounded way out of sync all the time. Turned out I was correct but it was my Creative Sound card, not Cakewalk to be blamed.
I returned to my Atari and my Yamaha MD8 because it synced up perfectly. 

Thanks, Johnny. Yes that is exactly what I was looking for.
Some difference in I/O latency makes sense. Input latency involves only the interface(at least it should).
If this is very low, it suggests that the interface is efficiently handling the A/D conversion for incoming signals.
The output latency is somewhat more relaxed, allowing the CPU to process effects, etc before handing the signal back to the interface for the D/A conversion. I get that.
The original drivers for the AKAI had an excessive output latency, similar to your Tascam, and it would lose sync after a minute or two.
I think the key here, and the premise of your original post, is that the numbers need to be reported correctly to Sonar. A few samples is no big whoop. Several hundred, and you are looking at a few ms, phasing issues, etc. I'm just glad I don't have to write the code for this stuff