32 Channel Interface or Two Different Interfaces at once

Like most it is a 2 channel stereo USB interface.
The board you have is a 16x 16 audio interface,,, that's a lot of channels. not many studios need more than that.
Are you still have trouble figuring out the routing for headphone cues?

Any takers on using two interfaces at the same time?

JohnKenn
Bmog,
 
Maybe too ignorant to be ignorant, and from the vantage of hard wired PCI cards, but...
 
Two different M-Audio cards on my system wired together via SPDF.  Expands 8 channels to 10 channels, and honestly have never had to use them all at one time. Anticipated what would the need be with a group of acoustic musicians with multiple mics on a drum or something. Never happened so far to go beyond the 10 channels.
 You can likely sync two sources, each at 16 channels to get 32 at a time with excellent clock sync. Same mfg should allow the same driver to be shared with both units. Question is, do you really need 32 channels at once? Potential is there for the right price, but is it needed or practical.
 
John
 
 
 

Thanks for the reply dont really need 32 but I need more than 12 and I ran across this while researching does this apply to Plat?

Those running Cakewalk's Sonar from version 2.2 onwards have the option of using ASIO drivers, but choosing the WDM/KS (Windows Driver Model/Kernel Streaming) driver option instead lets you assemble a composite interface from any combination of the stereo inputs and outputs that appear in the drop-down Sonar list. Since WDM/KS drivers bypass Microsoft's kernel mixer (which is normally used to mix the outputs from multiple audio applications into one stereo stream), this option can provide very low latency. However, quite a few musicians also seem to have used the feature to assemble an ad hoc arrangement of interfaces from different manufacturers. As long as you lock their clocks together in some way (see 'Locking The Clocking' box), they should be able to exist in perfect harmony.

http://dt7v1i9vyp3mf.clou...4jy=&itok=sBmcp7gK

I have had bad results with WDM driver for Sonar

Update, I see the new us16x08 is now different than the us1641 and us1800. It now has NO SPDIF but has the full 16 analogue inputs and 8 analogue outputs.   
Everything else seems the same. 

I guess this answers my question but now I am cuious to the advantage or disadvantage of asio vs wdm? http://forum.cakewalk.com...aspx?m=3079857&p=1

Jim Roseberry

BMOG
Any takers on using two interfaces at the same time?

By definition, ASIO only allows a single audio interface driver to be running.
Some companies get around this by allowing several different audio interfaces to use the same ASIO driver.
In this scenario, the multiple audio interfaces function as a single unit.
 
You can use WDM to address multiple audio interfaces, but then you have clocking issues, higher latency, etc.
Not an ideal solution.
 
If you need 32-channels of I/O, consider a Behringer X32 with a Lynx AES-16 with AES-50 port.
The Lynx card provides 32-channels of I/O connected to the X32's AES-50 port via a single CAT-5/6 cable.
Super low round-trip latency... and rock-solid performance.
 
If you're not looking for an actual digital mixer, have a look at the new MOTU AVB series.
You can run multiple units under a single ASIO driver... to configure just about any type/number of I/O required.
Excellent fidelity, super low round-trip latency, rock-solid drivers

Thanks Jim I am still wrapping head around this topic but getting closer. For example based off your theory on ASIO because I have one 01v96i working fine using ASIO if I got a second 01v96i connected via ADAT I would use ASIO for both or would I need to use WDM?

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? 

BMOG
Thanks Jim I am still wrapping head around this topic but getting closer. For example based off your theory on ASIO because I have one 01v96i working fine using ASIO if I got a second 01v96i connected via ADAT I would use ASIO for both or would I need to use WDM?

multiple interfaces via ASIO works ONLY if the driver was originally designed to work with more than one device of the SAME brand.
 
Roland used to do this with the VS system that also tied into the Octa-Capture (VS-700 and OctaCapture together gave me 24 IN @ 44.1 kHz, 20 IN @ 96 kHz)
 
I believe RME provides this option for all their interfaces (??)
 
MOTU with their AVB series is taking it to usability peaks. 2 of these units just replaced my older Roland system, options are mind blowing (although I sat several hours figuring out the best way to route everything)

BMOG

Rob[atSound-Rehab]

BMOG
Thanks Jim I am still wrapping head around this topic but getting closer. For example based off your theory on ASIO because I have one 01v96i working fine using ASIO if I got a second 01v96i connected via ADAT I would use ASIO for both or would I need to use WDM?

multiple interfaces via ASIO works ONLY if the driver was originally designed to work with more than one device of the SAME brand.
 
Roland used to do this with the VS system that also tied into the Octa-Capture (VS-700 and OctaCapture together gave me 24 IN @ 44.1 kHz, 20 IN @ 96 kHz)
 
I believe RME provides this option for all their interfaces (??)
 
MOTU with their AVB series is taking it to usability peaks. 2 of these units just replaced my older Roland system, options are mind blowing (although I sat several hours figuring out the best way to route everything)

I am a fan of Motu to connect your motus which driver did you use?

I'm using the latest AVB PRO driver with the super low latency - although this still beta and some users have connections issues with it (which I only see rarely)
 
The "official" USB only driver supports the same features as regards routing, but at somewhat higher latency

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/ 
 
 

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. 

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

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?

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)

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