So I decided to experiment different driver settings for my 01v96i
WDM\KS
Right away I get an error "The Following drivers either do not support the current audio format, or are in use by another application.  Please choose whether you want to disable them or use them anyway.  Now to add windows is using the 01 as it's sound card also.  I choose Use Anyway. In the device section there is only one input and output option for the 01v96i.  From a track perspective, there are no options for inputs and for outputs there is Left, Right and Center.  I am disappointed :(
 
WASPI
Right away I get an error "The Following drivers either do not support the current audio format, or are in use by another application.  Please choose whether you want to disable them or use them anyway.
The only device that shows up is Line (Yamaha 01v96i) which is how it shows up for my computer as a sound source. From a track perspective, there are no options for inputs and for outputs there is Left, Right and Center.
 
 
ASIO works properly
 

My experiments with using WDM mode had mixed results depending on the audio interface and the driver. 
I need to do this test again as a lot has changed in my system since including the updated Tascam drivers. 
Below is my us1641 loop back test with a midi track triggering session drummer. I then run my output back and record to a new track. I tried different modes and differnt buffers. Buffers made no differance at all. The offset stays the same at any setting. But you can clearly see that ASIO was early, WDM was way late. 
The test of other interfaces gave varying results. My Scarlett was bang on and did not have a WDM mode option, , my Card Delux PCI was bang on in both modes. My Behringer usb 22 was way off. 
 
http://www.cactusmusic.ca...scam%20test%20copy.jpg
Top track MIDI patten
2nd TRack Audio bounce of midi 
3rd track ASIO normal buffer setting loop back recording
4th track ASIO highest buffer setting loop back recording 
5th track WDM loop back recording 
 
But as I originally said, you can use WDM to record a band live using the multiple interface. My results where the tracks lined up perfectly. You don't normally need the channel count once the main session with drums etc is finished so you switch back to one interface running ASIO to continue with editing and overdubs. 

I use the Behringer XR18 - Which has 18 channels input (17 & 18 are a stereo pair 1-16 XLR, 1 & 2 Hi Z capable).
6 Aux outs, Monitor Out, Main, L/R Out.  All connected via USB 2.0.
 
Works great.  It is also a digital Mixer!
 
Here is link:
http://www.music-group.com/Categories/Behringer/Mixers/Digital-Mixers/XR18/p/P0BI8
 
I send the Monitor Out to Rane HC-6 headphone amplifier.
 
It also has the Ultranet built in which can send 16 channels to a user adjustable device:
http://www.music-group.com/Categories/Behringer/Signal-Processors/In-Ear-Monitor-Systems/P16-M/p/P0421
 
 
The one drawback of the XR18 is it only supports 24-bit at 44.1K or 48 KHz.  It does say it has 114 DB of dynamic range with 0.8 ms latency though! YMMV!

BTW BMOG, your music sounds really good!  When I get home I plan to get Sailing.  Love that song and what you did there!
 

patm300e

TheMaartian
Still trying to track the video down and will post it if/when I find it, but what it does is show the use of dual Tascam US-16x08 interfaces to provide 16 mic pres over a single USB cable. The US-16x08 defaults to a straight-through 8 mic preamp, feeding the 8 XLR inputs directly to the 8 line outputs, when no USB is present. The 8 line outputs are cabled to the second US-16x08's 8 line inputs (IN9-IN16). With this setup, you do lose the ability to directly plug a guitar/bass into IN9 or IN10. But we all use DIs with XLR outputs, right? 

Tascam Video can be found here:
http://tascam.com/product/us-16x08/getting_started/ 

Ahhh. It wasn't a video. It's the story below it about the home studio of Masayuki Muraishi. Thanks! I just didn't scroll down far enough. Doh!

Cactus Music
My experiments with using WDM mode had mixed results depending on the audio interface and the driver. 
I need to do this test again as a lot has changed in my system since including the updated Tascam drivers. 
Below is my us1641 loop back test with a midi track triggering session drummer. I then run my output back and record to a new track. I tried different modes and differnt buffers. Buffers made no differance at all. The offset stays the same at any setting. But you can clearly see that ASIO was early, WDM was way late. 
The test of other interfaces gave varying results. My Scarlett was bang on and did not have a WDM mode option, , my Card Delux PCI was bang on in both modes. My Behringer usb 22 was way off. 
 
http://www.cactusmusic.ca...scam%20test%20copy.jpg
 
But as I originally said, you can use WDM to record a band live using the multiple interface. My results where the tracks lined up perfectly. You don't normally need the channel count once the main session with drums etc is finished so you switch back to one interface running ASIO to continue with editing and overdubs. 

Your post inspired me to do a test of my own of a full loop back for latency.  Here is my setup let me know if I did it correctly.


Input track from 01 using AD 1 and AD2 to  track in sonar using Yamaha usb 1 and 2
Output is to a bus I named Keys which is going back into 01via channel 17 and 18
I can control the volume of the track using the channel slider of 17 and 18 from 01
I recorded some audio and midi tracks using the metronome as my way of seeing if I had latency issues and I did not.  Is this a true latency test?

TheMaartian
Ahhh. It wasn't a video. It's the story below it about the home studio of Masayuki Muraishi. Thanks! I just didn't scroll down far enough. Doh!

NP, they block most videos here at work, so I could not see the video, but do remember the story below.  I was considering buying the two Tascam 16x08s when I found my XR-18.  I replaced my interface AND a Mackie 1640 (read HEAVY) mixer with it.
 
 

BMOG
I recorded some audio and midi tracks using the metronome as my way of seeing if I had latency issues and I did not.  Is this a true latency test?

FWIW, That doesn't really test for latency... 
As long as the audio interface driver reports the correct amount of latency, the audio tracks are lined up properly in time.  Thus, they should all align properly... regardless of latency.
 
Sonar will report the round-trip latency of your audio interface... at a given sample-rate and buffer size.
 
If you're strictly monitoring via the digital mixer (not via software), then the latency when recording is a moot point.
You're not hearing it when recording... and it's compensated upon playback.
 
Two scenarios where latency is an issue:

• If you wish to monitor in realtime thru software based EFX/processing, you're dealing with round-trip latency.
• If you wish to play soft-synths in realtime, you're dealing with one-way (playback) latency.

 
Round-trip latency is the sum of the following:

• ASIO input buffer
• ASIO output buffer
• A/D and D/A converters
• The driver's safety-buffer (often hidden - can't be tweaked)

Jim Roseberry

BMOG
I recorded some audio and midi tracks using the metronome as my way of seeing if I had latency issues and I did not.  Is this a true latency test?

FWIW, That doesn't really test for latency... 
As long as the audio interface driver reports the correct amount of latency, the audio tracks are lined up properly in time.  Thus, they should all align properly... regardless of latency.
 
Sonar will report the round-trip latency of your audio interface... at a given sample-rate and buffer size.
 
If you're strictly monitoring via the digital mixer (not via software), then the latency when recording is a moot point.
You're not hearing it when recording... and it's compensated upon playback.
 
Two scenarios where latency is an issue:

• If you wish to monitor in realtime thru software based EFX/processing, you're dealing with round-trip latency.
• If you wish to play soft-synths in realtime, you're dealing with one-way (playback) latency.

 
Round-trip latency is the sum of the following:

• ASIO input buffer
• ASIO output buffer
• A/D and D/A converters
• The driver's safety-buffer (often hidden - can't be tweaked)

Thanks for the info

My test is not a latency test,,, it to test the accuracy of the offset that Sonar calculates so your tracks will be in snyc when you overdub. 
Because of latency Sonar needs to do this.
Sonar needs to calculate latency of the system and driver and adjust the playback so that overdubs are put down in sync with the existing tracks. If your input latency is say 10 ms your overdubs would end up being 10 sec late on the time line. Your output also has latency so what your playing along to has to be factored into the equation. 
 
 
 So the audio driver seems to be what Sonar has to rely on to figure out how much it will move things around. If the driver lies to Sonar it will result in the tracks being out of sync. This is not Sonars fault, it can only do what the driver is telling it what to do. 
So the loop back test is more or less a way to test the accuracy of driver. 
From what you say your system is working correctly. 
 
The test is simple as explained above. 
 
Use a MIDI drum pattern. 
Trigger any soft synth drum.  Bounce to audio. At this point you can see that these line up correctly.
 Then send that audio out of your interface and use a short cable to bring it back to any input. 
Record that input to an audio track and zoom in to see if the new recording lines up with the audio from the original. simple. Sonar allows you to adjust the offset if it is wrong,, but myself if it's wrong you need to buy a better interface. 
 
 
in my screen shot 
Top track MIDI patten
2nd TRack Audio bounce of midi 
3rd track ASIO normal buffer setting loop back recording
4th track ASIO highest buffer setting loop back recording 
5th track WDM loop back recording