MOTU UltraLite-mk5 Review (Audio Interface)

[mdsimon2]

If it works like that, it's a bit surprising. With most interfaces, setting the clock source to S/PDIF will be needed, but I was expecting that it should change the sample rate automatically as I think that my 20 years old MOTU is doing it. I will check if I'm not wrong (and on other interfaces I have)

Awesome job posting feedback on something you didn't test, really useful.

Remember that the sample rate families (44/48, 88/96, 176/192) present different channel counts so switching between them is not trivial. I should also say you can set the sample rate other ways besides Cuemix and the front panel, for example you can switch CamillaDSP configurations to do this. However, there is no automatic rate switching based on the TOSLINK input.

Michael

 

[Music1969]

Awesome job posting feedback on something you didn't test, really useful.

Remember that the sample rate families (44/48, 88/96, 176/192) present different channel counts so switching between them is not trivial. I should also say you can set the sample rate other ways besides Cuemix and the front panel, for example you can switch CamillaDSP configurations to do this. However, there is no automatic rate switching based on the TOSLINK input.

Michael

Thats Disappointing - RME ADI-2 DAC and Pro (latter supports 4 channel out) does auto detect and slave to SPDIF source sample rate

 

[mdsimon2]

Thats Disappointing - RME ADI-2 DAC and Pro (latter supports 4 channel out) does auto detect and slave to SPDIF source sample rate

RME clocking seems a lot more sophisticated than the MOTU UL Mk5. My Fireface 800 also automatically changes sample rates based on SPDIF input if the clock mode is set to AutoSync. I also really like how in AutoSync clock mode it will default to the internal clock if it isn't able to sync to your desired input. This is helpful when changing inputs as you don't need to worry about changing clock sources like you do on the MOTU UL Mk5.

Michael

 

[Grooved]

Awesome job posting feedback on something you didn't test, really useful.

Remember that the sample rate families (44/48, 88/96, 176/192) present different channel counts so switching between them is not trivial. I should also say you can set the sample rate other ways besides Cuemix and the front panel, for example you can switch CamillaDSP configurations to do this. However, there is no automatic rate switching based on the TOSLINK input.

Michael

Awesome job reacting like that
In case you did see it, I only said that other interfaces can do it, and that I'm a little surprised that MOTU decided to make this one not able to do such a thing.
I'm not saying "I'm surprised this guy didn't succeed to make it work because I'm sure the interface can do that, even if I never tested this interface", which would have been giving a feedback on this device without testing it.

Maybe you just had a bad and reacting like that made it a better one?

 

[PureLIN]

NONE of their devices support surround windows playback devices.

Antelope don't even support WDM when using thunderbolt (only ASIO).
I recommend you use VoiceMeeter, basic or banana is both free.
Install it will create virtual output(8ch)/input(2ch) ,can send to/from ASIO device.
Support EqAPO by it's bus patcher, and even can share you DAC/ADC through network by VBAN(not very stable though, but for daily playback is okay)

 

[anphex]

Antelope don't even support WDM when using thunderbolt (only ASIO).
I recommend you use VoiceMeeter, basic or banana is both free.
Install it will create virtual output(8ch)/input(2ch) ,can send to/from ASIO device.
Support EqAPO by it's bus patcher, and even can share you DAC/ADC through network by VBAN(not very stable though, but for daily playback is okay)

This sounds really complicated and prone to errors. Could you create a more detailed tutorial or link one?

 

[Grooved]

Antelope don't even support WDM when using thunderbolt (only ASIO).
I recommend you use VoiceMeeter, basic or banana is both free.
Install it will create virtual output(8ch)/input(2ch) ,can send to/from ASIO device.
Support EqAPO by it's bus patcher, and even can share you DAC/ADC through network by VBAN(not very stable though, but for daily playback is okay)

Did you tried "ASIO Bridge" that comes with "HiFi Cable" ?
This the one I used in this kind of cases, more than VoiceMeeter

 

[PureLIN]

This sounds really complicated and prone to errors.

If you not use VBAN then it's quite easy to setup. Just download the basic version, install, open it and set the A1 output to ASIO device, then set windows default output to voicemeeter input, and then you are good to go.
For EqApo, is same as install it on other device, the only different is you don't need to restart for it to take effect.
If you want to use more function, just check some video in youtube, ex:

Did you tried "ASIO Bridge" that comes with "HiFi Cable" ?

Used before, I'm not sure if it has 8ch output option, and I need VBAN and bus insert (it can run not only EQApo but also other applications, even your own code).

 

[Grooved]

If you not use VBAN then it's quite easy to setup. Just download the basic version, install, open it and set the A1 output to ASIO device, then set windows default output to voicemeeter input, and then you are good to go.
For EqApo, is same as install it on other device, the only different is you don't need to restart for it to take effect.

I will give it another try, thanks

Used before, I'm not sure if it has 8ch output option, and I need VBAN and bus insert (it can run not only EQApo but also other applications, even your own code).

Yes, you can set the 8ch under "ASIO Route" on each side, but there might be less options than in VoiceMeeter

 

[NTTY]

Sorry but English is not my strong point ... someone kindly explains to me what this wording in the manual means?

Quarter-inch line outputs are not cross-coupled. Therefore, when connecting them to an unbalanced input, use a TRS plug with the ring disconnected. Not floating the negative terminal will short it to the sleeve ground and cause distortion.

Which is the ring to be disconnected?

Hello everyone, I know this is old topic, but I did not see anyone publishing measurements.

So I followed the Rane paper to create my own cable (TRS with Ring not connected) and compared it to a standard TS cable. Motu says only the output will be impacted, and I can confirm that. So let's see if distorsion increases.

For the sake of clarity, I'll refer to TRS with Ring disconnected as "TRS(d)" in the rest of this message. So if I mention TRS, it means standard connector with Tip, Ring and Sleeve connected.

I did this test in loopback - TRS(d) out to TS in, TS out to TRS(d) in. I did the same with TS out / TS in as well. That TS out / TS in shows exact same results (performance degradation - see below) as with TS out / TRS(d) in, which demonstrates that the output is influenced by TS vs TRS(d) connectors, as per Motu's documentation.

I also repeated the same test, inserting a high definition analog device in the loop (a preamplifier, not to mention it). So in that case, the connections where TRS(d) to RCA -> RCA to TS, or TS to RCA -> RCA to TRS(d). The findings are the same with or without the preamplifier in the loop.

Enough blabla, there you go with the results (and I'm sorry, in that print screen TRS means TRS(d)):

This is a THD vs Level measurement with linear scale (because I prefer ), in dBr unweighted.
I removed frequencies below 200Hz as differences are nor relevant and also because I had too many FFT windowing errors screwing up the sweep, and I got bored trying to have a clean one at that resolution (6 points per octave). As you can see, yes there are differences but they are negligible.

I did a second sweep with lower resolution (2 PPO) and now you have the view from 20Hz. And this time it's log scale

Last note, find below a single FFT (1kHz) that shows a little more, even if these differences do not impact the THD since they are very low level noise. The green trace below (TS out) shows some spikes not present with TRS(d) out (which is the pink trace):

I would recommend to follow Motu's instruction but I must admit it's unlikely to make a sonic difference.

Enjoy your Motu!

 

[Davide]

Is there any way with the Mk5 to know the sample rate of the incoming spdif signal?
If it differs from the Motu clock, is it the DSP that does the resampling?

 

[Music1969]

Is there any way with the Mk5 to know the sample rate of the incoming spdif signal?

No, we discussed this a few posts above.

Unfortunately Motu does not detect incoming sample rate with SPDIF.

Some RME interfaces do

 

[Grooved]

No, we discussed this a few posts above.

Unfortunately Motu does not detect incoming sample rate with SPDIF.

Some RME interfaces do

Don't generalize... one of my interfaces is a Motu and it changes its clock automatically based on the SPDIF signal if SPDIF is set in the menu as clock source

 

[Davide]

No, we discussed this a few posts above.

Unfortunately Motu does not detect incoming sample rate with SPDIF.

Some RME interfaces do

I've read that the Mk5 doesn't switch by itself, my question is if you can simply see the clock of the incoming signal (which is sometimes not known in my case) so as to manually change the clock in CueMix

 

[Music1969]

Don't generalize... one of my interfaces is a Motu and it changes its clock automatically based on the SPDIF signal if SPDIF is set in the menu as clock source

I'm talking about the Motu of this thread obviously because that's what the question was... ;-)

I can't possibly have used every Motu interface on the planet

 

[Grooved]

I'm talking about the Motu of this thread obviously because that's what the question was... ;-)

I can't possibly have used every Motu interface on the planet

No problem, just that "Motu does not detect incoming sample rate with SPDIF. Some RME interfaces do" let me think that you were talking about all MOTU and RME interfaces and that only some RME could do it.

 

[Music1969]

I've read that the Mk5 doesn't switch by itself, my question is if you can simply see the clock of the incoming signal (which is sometimes not known in my case) so as to manually change the clock in CueMix

As I mentioned this particular Motu does not automatically detect incoming sample rate... so the only sample rate you will see on the front display or in CueMix is what you manually set yourself.

So it's not just that that the Mk5 doesn't automatically switch by itself, it does not even automatically detect incoming sample rate in the first place...

For auto switching, there first needs to be auto detecting.

 

[Davide]

As I mentioned this particular Motu does not automatically detect incoming sample rate... so the only sample rate you will see on the front display or in CueMix is what you manually set yourself.

So it's not just that that the Mk5 doesn't automatically switch by itself, it does not even automatically detect incoming sample rate in the first place...

For auto switching, there first needs to be auto detecting.

Ok, thanks to clarify.
At this point I wonder, if the sample rates between the incoming signal and CueMix don't match, is the DSP doing the resampling?

 

[Music1969]

is the DSP doing the resampling?

Yes has to be for an SPDIF source

Since SPDIF source has no idea what happens at the D to A clocking and conversion end, so it can't be the source end doing resampling

 

[PureLIN]

is the DSP doing the resampling

Some Audio interface do have resampleling for digital input, ex: lynx Hilo or old ver antelope zen tour (both can src for spdif), for when you want use device interal clock but need a digital input (in digital world, there should be only one mater clock).
But they are using a standalone src chip to do this, not DSP or FPGA.