thepianist53
I have the older Ultralite MkIII (firewire only) and am still pretty happy with it. Anyone feel I would get great gains if I upgrade to this one?  I do mostly virtual instruments (medium projects), but also some vocals and occasional tracking of "real" instruments on an individual or duo basis. Been happy with MOTU thru the years, and the only other I've considered as an upgraded would be the RME UCX.
 

The Ultralite AVB has a slightly lower noise-floor (~-110dB) and slightly better sounding converters.
The AVB's round-trip latency is slightly lower (4.9ms at a 64-sample ASIO buffer size/44.1k).
 
The onboard routing/mixing/DSP is similar to RME's Total-Mix.
 
If you're primarily working with virtual instruments, I'd continue using the MkIII.  

Well initial reports are in:
The latency certainly goes nice and low - but I'm getting crackles at anything under 12ms RT or so.
This is my first USB Interface, so I might have to tweak the co0mputer a little - I'll do some homework.
 
In the "Control Panel" settings, I guess the "USB Streaming mode" is for the hidden buffer settings, and then the ASIO buffer size is self explanatory. I can do various combinations of these - so I'll play around. Is there any advantage to making, say, the hidden buffers as low as possible and playing with the ASIO buffers? Or vice versa?
I'd guess that I might just have to juggle the two to see what can give me the lowest latency without crackles.
 
Also - the web interface (to set up the Interface routing etc) is a bit of a PIA. It loses connection all the time. I gave it a ip address - rather than using DHCP, but then the firmware update doesn't work (it reboots in DHCP mode looking for the firmware update, and doesn't find it)
 
These may well just be teething problems - so I'll play around some more - and report back.
 
The sound is certainly really good. So far I'm using my guitar in to a Lehle Sunday Driver - and then in to the "mic" input. The Lehle doesn't really output a line level signal - so this seems better than just going in to a analog input (I haven't tried the "guitar" input yet either)
 
I'm getting my head around the routing, and how SONAR reports the analog inputs - so it's really just down to getting rid of the crackles at low latencies.
PS - my Delta66 worked fine down to 7.8 ms RT latency (128 samples), and actually was even pretty good at 64 samples (around 5-6 ms RT latency) but I stuck with 128 usually, to keep the CPU usage down.  

Hi John,
 
I can run at 4.9ms total round-trip latency with zero glitches.
Make sure you're not connected to a USB3 port (especially not a 3rd party USB3 controller).
 
FWIW, You can create a shortcut to the IP address (which will open the control-panel application).
You can even use the same icon.

Make sure you're not connected to a USB3 port (especially not a 3rd party USB3 controller).

Thanks Jim.
OK - there's clue number one! ;-)
I specifically bought a PCI card with 2 x USB3 ports on it - to use just for this.
I'll try a "normal" USB2 port - and I'll google around to see what I can tweak. As I said, I've always used a PCI card for audio - so there might be something else obvious I can fix.
I'm sure this is going to work out great - and I fully expected to have to do a little work to get there.
This forum just makes it a little easier ;-)
 
For the ip address - yeah, it's no big deal to type the ip address in to the browser (it's DHCP, so technically this can change). This is not ideal for the less computer literate though ;-)

==== UPDATE ====
Hopefully this will help anyone else going for this interface.
I noticed that there are certain combinations in the Audio Control Panel that do not work. You do actually get a warning, but I hadn't noticed it. I was wondering why my SONAR sounded horrible, and slow - like it was at the wrong sample rate! In fairness - it *does* warn you... (I wasted a lot of time with this)
 
I moved the USB connector to a USB port on the mobo, and it's not sharing a "Generic USB Hub" with anything else.
It did seem pretty much the same on a USB3 port on the mobo too.
 
==== Audio Control Panel settings ====
I've set "USB Streaming Mode" (the hidden buffers) to the lowest setting - and then adjust the ASIO Buffer Size.
- 64 samples (4.9ms RT latency) - Works OK without crackles but only with very few audio and/or soft synth tracks.
- 128 samples (7.8ms RTL) - Is much better. There are *occasional* crackles, but I'm going to work at this setting and see how it goes.
- 256 samples (13.6ms RTL) - This is still very usable. I'm sure that latencies like this bothered me with the Delta66 (playing through an amp sim plugin). Maybe it's a placebo effect because 13.6ms is 13.6ms no matter which audio interface you play through - but others have reported that they're OK playing with latencies like this too. Anyway - there are no crackles or anything at this setting.
=========================
 
The audio quality is great - although I'll pay more attention to this through the week when I actually use the Ultralite for tracking. I made sure I can return this interface "just in case" I had problems running a USB interface - but I think this is definitely going to be a keeper. I think the fact that I can't get the latency a little better is more due to my PC setup than the MOTU drivers or the hardware itself.
 
It also seems to be connecting to the web interface OK today too - but as you say Jim, it's easy enough to just type the ip address (from DHCP) in the browser, and that *always* works.
 
== I'll maybe play around with the DSP mixer and stuff sometime, and find out what "audiodesk" is, but for now I think this is going to work out well.
 
I'll report back later in the week.
Thanks again Jim for your input, and help. This seems like a good low latency AI.

One more quick question  - We *do* still record at 24 bit (and I uses 44.1KHz sample rate) don't we?
Or do modern audio interfaces record at 32 bit nowadays?

24 bits.  32 bits can be used for upsampled masters, but no hardware actually uses 32 bits as I understand it.
 

You can't actually make a DAC that'll resolve 24-bit of dynamic range, much less 32-bit. The inherent noise you get in components means that actually achieving 144dB of range is essentially impossible. As such 24-bits is more than we need, going to 32-bit doesn't do anything except record more noise.
 
There are DACs that take 32-bit inputs but that is a combination of marketing BS, and simple convenience (since most computer buses are 32-bit) not any actual usefulness.

24 bits.

Thanks to both of you for the sanity check. I just had this thought - "I wonder if things have moved on...." ;-)

========== UPDATE ========
Here's another update - after tracking guitar all day at the second lowest latency setting (128 samples - 7.8ms RTL) this audio interface sounds terrific, and performed flawlessly.
I even rigged up the mixer to strap a (LA2A) limiter over the guitar input (well, I'm using the XLR mic input to record guitar).
 
The web interface goes a little wacky when you access the mixer - but I figured out that it just needs to be minimised, and then open it again and it's fine ;-)
 
Anyway - I think (unless something stupid happens in the next few days) I'll be keeping this - and will highly recommend it.
 
Thanks again to Jim R - for starting the topic...