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MOTU 1248, 8M, 16A Thunderbolt interface Audio Interfaces
Old 3rd October 2016
  #2761
Quote:
Originally Posted by barbaroja View Post
Would like to know the jitter figures in picoseconds for the on the 16A and 1248. Not published tho.
Would be nice, I agree. The THD+N measurements (-110 dB, unweighted) will include distortion from jitter, though, so it can't be too significant.
Old 3rd October 2016
  #2762
Gear Maniac
 
barbaroja's Avatar
 

Yeh I know it's more of a simple number and sound quality depends on way more stuff. But it does add to the equation and most major manufacturers publish it.
Old 3rd October 2016
  #2763
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Quote:
Originally Posted by barbaroja View Post
Yeh I know it's more of a simple number and sound quality depends on way more stuff. But it does add to the equation and most major manufacturers publish it.
It also adds to the equation of course, but let me expand, why I think, its more of marketing gimmick.. Which I typically just translate to the claim - "We've somehow upgraded clocking at our latest model", good to know.
Realistically it's not just one figure sourced from their oscillator vendor, because master clock frequency is being modulated by multitude of different frequencies.. and it's significantly affected by the component powering, so you need ideally phase noise plot in particular circuit under normal operating conditions (eg. audio streaming, some audio modulation).. even with that, after you'll reach some level, it's still not clear, how it will/might affect perceivable quality of the conversion.
Also distribution from crystal oscillator is alter important.. it's quite fragile signal and non-careful treatment and other circuit influences can easily deteriorate it.
So where this measurement can be obtained.. maybe at leg of converter chip.. cool, which converter chip.. if you have device with 16 I/Os for example.. maybe few different types and models (not just between different interfaces by different vendors, but also in the one unit).. then inside of each chip type are sections with programmable clock dividers, which might also affect final conversion performance. Uff..
So, that's why I think, it's somewhat naive to draw some conclusions from one published figure without the context.. For meaningful clock evaluation, you'll need to set some methodical approach and measure several different units under the test exactly same way, ideally at one bench with the same (terribly expensive) device. Some might be required to mod for reaching of right measurement points.
Finally, back to the subjective equation.. and perceived audio improvement evaluation.. assuming, no corners will be cutted, how do you rate clocking improvement in whole attribute mix for some device.. It's more or less important than dynamic range, distortion level and its harmonic structure.. Do you think it's more important to put more expensive oscillators into device or money will be better spend at finely tuned discrete I/V stage after DAC chips.. how about independent power supplies for different sections or overengineered PCBs..
So if you take two products from comparable category, each vendor and his engineering team, might take different design emphasis and focus (eg. Antelope - weirdly described clocking).. or balance the mix to different direction, because they might believe it's important.
But for some specs shootout it's very hard to pull out just one attribute and assume how it would sound in comparison to other similar device.

Michal
Old 4th October 2016
  #2764
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Janzoulou's Avatar
Quote:
Originally Posted by wilkinsi View Post
I can't use a buffer setting lower than 512 without getting pops and clicks, though the fact I'm only using a Macbook Pro "13 Retina with 2.7GHZ Dual Core processor and 8GB RAM might have something to do with that. Its connected via Thunderbolt to my 112D. I use MOTU AudioDesk 4.0 as a soft DAW, with everything set to 44.1k. The 112D has plenty of digital gear connected to it.
Here too: System macPro 6.1 3,7gHz Quad, 32GB RAM OSX Yosemite + Logic Pro X
MOTU 112D (connected via TB) & 24o (connected to the 112D via AVB) latest firmware and driver (not beta)
The machine is only doing audio, no other applications running in the background.

Lowest Buffer size I can handle without pops/clicks is 128.
I contacted support but so far no idea on their side.
Any Ideas?

Cheers, Fredi
Old 4th October 2016
  #2765
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Quote:
Originally Posted by msmucr View Post
It also adds to the equation of course, but let me expand, why I think, its more of marketing gimmick.. Which I typically just translate to the claim - "We've somehow upgraded clocking at our latest model", good to know.
Realistically it's not just one figure sourced from their oscillator vendor, because master clock frequency is being modulated by multitude of different frequencies.. and it's significantly affected by the component powering, so you need ideally phase noise plot in particular circuit under normal operating conditions (eg. audio streaming, some audio modulation).. even with that, after you'll reach some level, it's still not clear, how it will/might affect perceivable quality of the conversion.
Also distribution from crystal oscillator is alter important.. it's quite fragile signal and non-careful treatment and other circuit influences can easily deteriorate it.
So where this measurement can be obtained.. maybe at leg of converter chip.. cool, which converter chip.. if you have device with 16 I/Os for example.. maybe few different types and models (not just between different interfaces by different vendors, but also in the one unit).. then inside of each chip type are sections with programmable clock dividers, which might also affect final conversion performance. Uff..
So, that's why I think, it's somewhat naive to draw some conclusions from one published figure without the context.. For meaningful clock evaluation, you'll need to set some methodical approach and measure several different units under the test exactly same way, ideally at one bench with the same (terribly expensive) device. Some might be required to mod for reaching of right measurement points.
Finally, back to the subjective equation.. and perceived audio improvement evaluation.. assuming, no corners will be cutted, how do you rate clocking improvement in whole attribute mix for some device.. It's more or less important than dynamic range, distortion level and its harmonic structure.. Do you think it's more important to put more expensive oscillators into device or money will be better spend at finely tuned discrete I/V stage after DAC chips.. how about independent power supplies for different sections or overengineered PCBs..
So if you take two products from comparable category, each vendor and his engineering team, might take different design emphasis and focus (eg. Antelope - weirdly described clocking).. or balance the mix to different direction, because they might believe it's important.
But for some specs shootout it's very hard to pull out just one attribute and assume how it would sound in comparison to other similar device.

Michal
Interesting view. Thanks for sharing.
Old 4th October 2016
  #2766
I just bought a 16A.. It's great.. One thing I found different between demoing the 1248 and the 16A is the buffer size only goes to 1024 using usb, vs the 1248 which went much higher.. Can't remember... it was either 4096 or 8192 something much higher..

My Rme digiset does 4096 and it's almost 10 years old.. Not happy about the 1024 buffer settings as reports were that everything was the same throughout the AVB line.

Sound quality wise it's great..
Old 5th October 2016
  #2767
Jca
Gear Nut
 

Quote:
Originally Posted by wilkinsi View Post
That's odd. I've never been charged for AudioDesk 4.0. I checked Activity Monitor, and couldn't find anything wrong. AD4 and Falcon only up to 30% of the CPU each at the most. Before I lost my job, I was seriously considering a Mac Pro (the "trash can" lookalike) with 64GB RAM. Just as well that I didn't, from what you're saying. Am I to assume that there isn't any hope for anyone running this software, because they will always have these problems, no matter what spec their Mac is?
As far as I know Audiodesk is still "free" as in included with a new interface.

I had the popping and clicking problem using an Ultralite AVB over ethernet at 96K. Only thing that fixed it was dropping to 48K.
Old 6th October 2016
  #2768
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Monkey Man's Avatar
 

Curious, guys:

The UltraLite mk4 has just been announced, with all the features we're now familiar with of the AVB line, even the "web apps" and remote control, but without AVB.

Is it a naive question for me to ask whether or not this could indicate that there's a chance that a simpler system for multiple interfaces, minus the "additional" complications / intricacies of the AVB one, could be on the horizon?

I ask 'cause due to my lay-buy having been lost 6 months ago, my already-delayed purchase of a 16A and two 24Ai units has had to be put off even further. I'm admittedly daunted by and apprehensive about the whole AVB thing, requiring only a simple CueMix-style setup so as to be able to monitor realtime groups of my MIDI ROMplers and so on. It'll be a set-and-forget affair if I can help it.

Sorry if this is the dumbest question of all time...
Old 6th October 2016
  #2769
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loopy's Avatar
 

Can someone (Mr. Miller??) from MOTU comment on what the specific differences are between the "Host Buffer Size" and Host Safety Offset" settings are?

There is currently a thread in the Interface Low Latency Database thread where we are discussing this and the documentation is a little vague so some of us are looking for additional information.

Thread is over here:

Audio Interface - Low Latency Performance Data Base

Thanks much!
Old 6th October 2016
  #2770
Here for the gear
 

Hope someone can give a definitive answer to this question - it's an area I'm by no means an expert on.

I want to use all 16 ins and outs of my Motu 16A for external sends and returns from Cubase. Currently I use outputs 15 & 16 as the main feed to my monitors.

That means I need some other way of getting an extra pair of outputs from the 16A.

What I would like to know is whether I can connect this Schiit Audio, Headphone amps and DACs made in USA. (The Uber version with Toslink) to the optical outputs on the 16A or not.

Seems in some places I read that SPDIF over optical is possible on the 16A, and some places not. Some stuff seems to suggest it's ADAT only, but in other places that SPDIF is supported.

Can anyone who actually knows please inform me one way or the other?!

Cheers

Rob

Edit - of course, just after posting I see this for the first time : https://www.gearslutz.com/board/10640501-post891.html

Has anyone actually made it work though?

Last edited by rob_gould; 6th October 2016 at 04:30 PM.. Reason: New information
Old 7th October 2016
  #2771
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dandeurloo's Avatar
Quote:
Originally Posted by rob_gould View Post
Hope someone can give a definitive answer to this question - it's an area I'm by no means an expert on.

I want to use all 16 ins and outs of my Motu 16A for external sends and returns from Cubase. Currently I use outputs 15 & 16 as the main feed to my monitors.

That means I need some other way of getting an extra pair of outputs from the 16A.

What I would like to know is whether I can connect this Schiit Audio, Headphone amps and DACs made in USA. (The Uber version with Toslink) to the optical outputs on the 16A or not.

Seems in some places I read that SPDIF over optical is possible on the 16A, and some places not. Some stuff seems to suggest it's ADAT only, but in other places that SPDIF is supported.

Can anyone who actually knows please inform me one way or the other?!

Cheers

Rob

Edit - of course, just after posting I see this for the first time : https://www.gearslutz.com/board/10640501-post891.html

Has anyone actually made it work though?
I use it everyday.
Old 7th October 2016
  #2772
Gear Nut
Quote:
Originally Posted by loopy View Post
Can someone (Mr. Miller??) from MOTU comment on what the specific differences are between the "Host Buffer Size" and Host Safety Offset" settings are?

There is currently a thread in the Interface Low Latency Database thread where we are discussing this and the documentation is a little vague so some of us are looking for additional information.

Thread is over here:

Audio Interface - Low Latency Performance Data Base

Thanks much!
Hi all,

The Host Safety Offset is a setting provided on Windows for fine tuning the latency beyond the normal buffer size increments. Here's how we describe it in the Pro Audio Readme:

"This setting allows
you to fine tune host latency. Larger offsets allow the driver
more time to process audio as it transfers to and from the
hardware. Lower settings produce lower latency, but if you go
too low, your host software may experience performance
issues"

The full documentation is available here: http://cdn-data.motu.com/manuals/avb...0Read%20Me.pdf

I hope this helps!
Old 7th October 2016
  #2773
Gear Nut
Quote:
Originally Posted by Chris Perra View Post
I just bought a 16A.. It's great.. One thing I found different between demoing the 1248 and the 16A is the buffer size only goes to 1024 using usb, vs the 1248 which went much higher.. Can't remember... it was either 4096 or 8192 something much higher..

My Rme digiset does 4096 and it's almost 10 years old.. Not happy about the 1024 buffer settings as reports were that everything was the same throughout the AVB line.

Sound quality wise it's great..
Buffer settings are the same throughout the AVB product line. The maximum available Host Buffer Size for these devices is determined by the setting of the sample rate. Higher sample rates will yield higher maximum buffer sizes. For example, when running at a sample rate of 44kHz you will have the option of choosing a buffer size of 1024. Running at 88k will offer a buffer as high as 2048 and at 176k you will be able to choose a buffer as high as 4096.
Old 7th October 2016
  #2774
My RME runs at 4096 a 44.1

The 1248 allowed picking a higher buffer setting even at 44.1 as far as I can remember..

A Zen studio I rented was the same..
Old 8th October 2016
  #2775
There's definitely something about the 16A that's different.. I'm calling up mixes using Uad plug ins that the 1248 and my Rme have no cpu hit on.

With the 16A.. it's taxing the CPU.. they are not the same. What's up with that?
Old 8th October 2016
  #2776
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Quote:
Originally Posted by Chris Perra View Post
There's definitely something about the 16A that's different.. I'm calling up mixes using Uad plug ins that the 1248 and my Rme have no cpu hit on.

With the 16A.. it's taxing the CPU.. they are not the same. What's up with that?
That's definitely weird.. I'd expect all AVB lineup should use same USB audio interface hardware and drivers.. It doesn't make much sense to develop all that individually for bunch of similar products, which apparently differs only at I/Os.
Firmware and driver releases are pretty much in-line for whole product range with similar changes between the versions.
From its launch two years ago, it was quite a lot of refinements under the hood and it's possible, that some changes were introduced to the buffer handling or its range of settings.
So maybe some previous firmware version at 1248 allowed mentioned larger buffer setting, but later they modified that and 16A with possibly newer firmware has different range.. I don't know.
Other thing is, in my experience with different setups.. there is typically some diminishing return with buffer size depending on the project even if interface driver allows that.. eg. it's not uncommon, that increase of buffer length over certain value (say 1024) doesn't really bring any performance improvement and everything only lags more.

Also I'd recommend to do some more methodical testing with different interfaces and trace system processes usage.. for example during multichannel playback of exactly same projects with comparable settings among interfaces to somehow sort out the feel thing, when really doing different tasks. Typically, when audio streaming isn't working well, there's possible to spot some some excessive usage of some kernel task.
Finally if you're in the doubts about 1248 and 16A differences, you can try to directly reach MOTU support, which should clear that out.

Good luck!

Michal
Old 9th October 2016
  #2777
Gear Nut
 
wilkinsi's Avatar
As regards to not seeing the buffer settings in the AVB web browser, I have contacted MOTU Support. I upgraded to OS X 10.12 Sierra yesterday. Having uninstalled and reinstalled the AVB driver, opening the browser showed it had retained my previous audio routing settings. Still no sign of the buffer settings. I can only adjust buffer settings in UVI Falcon. I can't find any buffer settings in AudioDesk 4.0.
Old 10th October 2016
  #2778
Quote:
Originally Posted by wilkinsi View Post
As regards to not seeing the buffer settings in the AVB web browser, I have contacted MOTU Support. I upgraded to OS X 10.12 Sierra yesterday. Having uninstalled and reinstalled the AVB driver, opening the browser showed it had retained my previous audio routing settings. Still no sign of the buffer settings. I can only adjust buffer settings in UVI Falcon. I can't find any buffer settings in AudioDesk 4.0.
As far as I know there are no buffer settings in the WEB UI for OSX as these are Windows only. OSX buffers are controlled by the DAW/Audio application. In Audio desk it's "hidden" under Setup-Configure Audio System-Configure Hardware Driver...-Buffer Size.
Old 10th October 2016
  #2779
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wilkinsi's Avatar
Quote:
Originally Posted by rolandvg99 View Post
As far as I know there are no buffer settings in the WEB UI for OSX as these are Windows only. OSX buffers are controlled by the DAW/Audio application. In Audio desk it's "hidden" under Setup-Configure Audio System-Configure Hardware Driver...-Buffer Size.
Despite being connected via Thunderbolt, I was previously able to access those buffer settings. Thanks for letting me know about AD4.
Old 10th October 2016
  #2780
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wilkinsi's Avatar
I don't suppose there's somewhere in the AVB web browser, AD4 or the Macbook itself that measures how much latency exists?
Old 10th October 2016
  #2781
Quote:
Originally Posted by wilkinsi View Post
Despite being connected via Thunderbolt, I was previously able to access those buffer settings. Thanks for letting me know about AD4.
The setting can sporadically be available in OSX. I've had it pop up when hot-pluging my 8M, but according to the manual it's Windows only.
Old 13th October 2016
  #2782
Gear Nut
 
Rafter Man's Avatar
Quote:
Originally Posted by wilkinsi View Post
As regards to not seeing the buffer settings in the AVB web browser, I have contacted MOTU Support. I upgraded to OS X 10.12 Sierra yesterday. Having uninstalled and reinstalled the AVB driver, opening the browser showed it had retained my previous audio routing settings. Still no sign of the buffer settings. I can only adjust buffer settings in UVI Falcon. I can't find any buffer settings in AudioDesk 4.0.

Quote:
Originally Posted by rolandvg99 View Post
As far as I know there are no buffer settings in the WEB UI for OSX as these are Windows only. OSX buffers are controlled by the DAW/Audio application. In Audio desk it's "hidden" under Setup-Configure Audio System-Configure Hardware Driver...-Buffer Size.

I asked about this a while back (probably no more than within the last 5 or so pages of this thread), and the MOTU rep replied that this wasn't included for the Mac OSX version because it wasn't needed. I was curious because I was on Windows 7 when I first purchased my 16A over USB about 18 months ago (and i've been on Mac OSX over Thunderbolt for a little over 12 months now.)

While I don't fully understand the science behind it, I was assured it was a non issue and that I shouldn't expect to see it appear on any updates for the foreseeable future.

And since i've experienced virtually no freeze ups or lock outs since my transition to running the unit in Mac OSX over Thunderbolt, I'm able to accept that sentiment at face value, provided the setup continues to remain as stable as it has been.

And just to clarify for any Windows users reading into this, that's not to say the occasional drop outs and error messages that would say to the effect of "the unit cannot locate the device" were due to it being through Windows and/or over USB. I suspect it was probably because the updates to the drivers and firmware had just continued to improve over time, as i've observed in this thread that others who have owned the box roughly as long as myself have experienced similar results.
Old 13th October 2016
  #2783
Gear Nut
 
Rafter Man's Avatar
Quote:
Originally Posted by rob_gould View Post
Hope someone can give a definitive answer to this question - it's an area I'm by no means an expert on.

I want to use all 16 ins and outs of my Motu 16A for external sends and returns from Cubase. Currently I use outputs 15 & 16 as the main feed to my monitors.

That means I need some other way of getting an extra pair of outputs from the 16A.

What I would like to know is whether I can connect this Schiit Audio, Headphone amps and DACs made in USA. (The Uber version with Toslink) to the optical outputs on the 16A or not.

Seems in some places I read that SPDIF over optical is possible on the 16A, and some places not. Some stuff seems to suggest it's ADAT only, but in other places that SPDIF is supported.

Can anyone who actually knows please inform me one way or the other?!

Cheers

Rob

Edit - of course, just after posting I see this for the first time : https://www.gearslutz.com/board/10640501-post891.html

Has anyone actually made it work though?
I also work in Cubase (version 8.5 Pro) and do this all the time on my 16A through OSX over Thunderbolt, and it took me a ton of diligence to read up on everything since there was basically no information that I could find that addressed exactly what I was seeking to do.

The way i understand how the optical ports work is that the Optical B input/output banks are only for ADAT, and that any peripheral you connect this to would need to likewise be ADAT compatible.

With the Optical A input/output banks, you can run them as either ADAT, or as TOSLink.

Since i'm almost completely out of available analog inputs/outputs on my 16A, I needed to find a way to digitally connect my Yamaha SPX2000 effects processor unit so I could open up another pair of analog inputs/outputs for my ever growing outboard collection. Unfortunately, the SPX2000's digital inputs/outputs are not ADAT compatible, as it uses an AES/EBU connection instead. However, there's one product on the market (it's discontinued, but you can still find it on Amazon, eBay, Sweetwater, etc.) called the Hosa ODL-312 that can convert the AES/EBU connection to that of a TOSLink optical, which the 16A's Optical A input/output banks can be formatted to accept. All you need to do is go into the MOTU Discovery App and find the "Optical Setup" section on the first tab called "Devices". Just select the "Format" field next to the input/output for Optical A and change it from ADAT to TOSLink, and then you're good to connect two channels to any piece of gear using SPDIF TOSLink and can also connect this to AES/EBU if you get the Hosa ODL-312 converter like I've been using.

As far as the Optical B input/output bank is concerned, since you're limited to it being an ADAT connection only, in addition to there being no product on the market i'm aware of that can convert an ADAT signal to any other digital medium, i'll probably never get any use of it since i'm planning to buy another 16A and can connect them via the ethernet port.

I wish the Optical B input/output bank could also be formatted to accept a TOSLink connection or was just simply SPDIF, as that would allow for much better versatility, but I guess I can't really complain too much about that since I think the 16A is by and far the best product out there at such a price point, for those that already have high end mic pre's and have zero need or desire for an audio interface that has any other analog connection besides Line inputs/outputs. I wish more manufacturers would make note of this, because I surely cannot be the only one who feels that way.
Old 18th October 2016
  #2784
Here for the gear
 

Fellow MOTUers

I wanted to pass on some experiences I have had with my system lately. All machine details listed below in my sig. I had been experiencing a ‘pop’ on audio channels 3-16 (never 1 or 2) about every 11 to 14 minutes while recording. After the loud ‘pop’ timing would be off and there would be significant latency in the system. Often audio would become distorted following the ‘pop’ as well. I tried many things to find the source of the ‘pop’ including rollback to Win7, PT 12.4 and PT 12.5, several revisions of the MOTU drivers, unplug all peripherals and extra drives, lower video resolution, turn off all power saving, …

After a long weekend of debugging (not making music) it turned out the problem was the USB2 ports on the motherboard. Note, the USB2 ports work perfectly with all other peripherals; the ports just seem to have an approximately 12 minute cycle on some kind of interrupt that produced the loud ‘pop’ and messed up timing.

To solve the problem I switched the MOTU 1248 to a USB3 port and all other peripherals to USB2 ports. This stopped the 11 minute cycle ‘pops’.

Finally, in combination with the latest MOTU drivers and PT 12.6, I am getting great low latency performance. I have stable operation while recording at 24/96 with 64/64 buffers. At these low buffers I get more stable PT CPU usage (i.e. no spikes) if I set the processor affinity at 6 of 8 cores when loading PT. This frees up 2 cores to deal with system load. At higher buffers like 256/128 this is not necessary. Interestingly, the CPU usage meters for the system cores show much higher and erratic CPU load with the MOTU plugged into the USB2 port. When the MOTU is plugged into a USB3 port the CPU load on the 2 system cores is about 20% lower and the CPU usage meter is stable.

Hope this helps someone.

ejinbc

>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Machine: Assu P8Z77i MB, 3770K CPU, OC @ 4.5GHz with Noctua NH-D14 air cooling, 16 GB RAM, 256 & 512 GB SSDs, 4 TB Data drive, Gigabyte 950 video HDMI 2.0 @ 4K resolution, all USB peripherals plugged into USB2 ports EXCEPT Motu 1248 which is plugged into rear USB3 port, Turbo Mode Off in BIOS, C-states Off in BIOS, WiFi Off in BIOS, BT Off in BIOS, On-board sound Off in BIOS, HPET Off in BIOS. Never online.

Midi: MAudio MidiSport 8x8/s, Command|8, Roland TD6, Edirol PCR M80, Korg SP250

Software: Windows 10 (all standard DAW optimizations), Pro Tools 12.6 - uninstalled App Manager & Web Services, Waves Gold Bundle + Waves HEQ, Native Instruments Komplete 10,

Session details for typical recording: 24 bit 96 kHz, 64/64 buffers, 16 Audio Tracks, 3 VI Instrument Tracks, 2 HEQ and 1 HDelay instance. PT 12.6 run with processor affinity at 6 of 8 cores.
Old 22nd October 2016
  #2785
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Can any of you guys answer a possibly dumb question about sample rate setting on the AVB line: does whatever sample rate you set affect the entire machine (ie, how the AD/DA converts the analog inputs and outputs, as well as the sample rate of the ADAT ports)?

More specifically, my question is this: if I choose to set the box to run at 96 kHz (for the presumed benefit of better conversion of the analog inputs), does that mean I won't be able to input an ADAT device that only outputs at 48 kHz?

Thanks.
Old 22nd October 2016
  #2786
Gear Nut
 
Rafter Man's Avatar
You should run everything at the same sample rate. Maybe there's a way around this, but I wouldn't recommend it. You can run ADAT SMUX II at 96 kHz, as it's an option on the MOTU and many other devices on the market.
Old 22nd October 2016
  #2787
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Quote:
Originally Posted by Rafter Man View Post
You should run everything at the same sample rate. Maybe there's a way around this, but I wouldn't recommend it. You can run ADAT SMUX II at 96 kHz, as it's an option on the MOTU and many other devices on the market.
I know about SMUX, but I guess my real question boils down to the following:

I have a TON of gear to input into some combination of these AVB boxes (if I decide to buy them), so I'm trying to maximize my available inputs. I have 3 synths that happen to have ADAT outs, but I'm certain they only go up to 48 kHz because they're "vintage." I'd like to be able to connect them via ADAT, simply because then that would free up 3 pairs of analog stereo inputs for some of my other synths. However, would this mean that I would HAVE to set the MOTU boxes to run at 48 kHz? (And if I set them to 96 kHz, what would happen to those 48 kHz ADAT synths -- there would be no sound or something?)

Thanks for any advice.
Old 22nd October 2016
  #2788
Lives for gear
 

It's simple, any interface has just one active sample rate for its operation. So if you have some digital devices with 48k outputs and you'll switch an interface to 96k, it won't work.. inputs just loose it sync.
So either ditch the 96k idea or hook-up those synths via analog.

Michal
Old 22nd October 2016
  #2789
Lives for gear
 

Additionally, only way how to interface digital devices with different sample rates is use of some external SRC box. But for multi-channel ADAT operation, it will be rather pricey route.
For example this device is capable of that..
https://www.rme-audio.de/en/products/adi_192_dd.php
But it's questionable, if it's better way than getting additional 8 analog inputs.

Michal
Old 22nd October 2016
  #2790
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Hokut's Avatar
 

Wow, I am back.
I had a 1248 and a 16A for a few months but only now I was able to switch them on!
Anyway, only had time to do basic setup

MAC Pro 3.1 (2008) - Finger crossed it keep working for now
OSX Yosemite - Deciding if I should upgrade to El Capitan ? (Sierra does not support MAC Pro 3.1/2008... only MACs fro 2009)

All my hardware synths are connected (Inputs and Outputs) to 1238 and 16A which then go to Patchbays... did this when I first got the MOTUs, can't even remember now the details but luckily enough I did label the cables and I have connection plans saved.

Today
1) 1248 via USB to MAC PRO + CAT 5 to MAC PRO to update the Firmware - It worked
2) 16A via USB to MAC PRO + CAT 5 to MAC PRO to update the Firmware - It worked

3) 1248 via USB to MAC PRO and 16A via CAT 5 to 1248
I need to get a better network cable, CAT 6 but I only had an old CAT 5 at hand for today

Installed latest version of the MOTU AVB Discovery App.. and both devices show up. I did initial look around of the app and setup of the devices.
NOTE initially I set both devices to 96Khz for 32 channels. I intend to use the ADATs as 8 channels at 96Khz as well
I set the 1248 to internal clock and 16A to use clock from 1248

Then I fired up Logic simply to get a midi track going and have audio to come out of the 1248. The 1248 main outs go to my Active 2.1 system

I edited the Routing so that From Computer 1 and 2 went to 1248 Main Outs... and GREAT sound came out

Then I realised that both 1248 and 16A changed from 96Khz to 48Khz (Or was it 44.1K? I can't remember now)
I went into the Device App and tried to change it back to 96Khz BUT it won't let me do it, the clock icon flashes, value goes back to 44.1 (or 48 can't remember) and settles on that. It won't let me change it back to 96Khz


I run out of time so that's how far I got. Tomorrow I will carry on but first will need to do setup to implement MOTU Midi Express XT, then make sure that my hardware synths' audio does show up in Logic (via 1248 and 16A), make sure overall the whole setup with
1248+16A+Logic+Patchbays works... then I will go back to the 96Khz issue


But I wanted to ask... do you have ideas what the issue could be with not letting me switch to 96Khz? I was trying to set 96 on both devices and it worked at first. Could this be a problem with the CAT 5 connection between 1248 and 16A? Or maybe a limitation of AVB with Yosemite?

I will keep you posted with how all works out in my setup because I know there are a few people around with older MACs that may have similar issues or want to know how it works before they get MOTU interfaces... Finger crossed will get it all sorted
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