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MOTU 1248, 8M, 16A Thunderbolt interface Audio Interfaces
Old 9th April 2015
  #1411
Gear Maniac
 

Quote:
Originally Posted by mrmiller View Post
Yup! All the interfaces support 64/64 channels in/out over USB with the latest 1.2.0 firmware. You can easily link two 16As with an ethernet cable and get a full 32/32 analog I/O. Add in the ADAT on both boxes and you could use all 64/64 channels to/from the computer. Pro Tools, however, limits non-Avid devices to 32/32 I believe.



The 8 analog ins on the back do not go through the pre-amps—only the 4 mic ins do. That said, all analog inputs do go through a trim stage that you can control to adjust the level as needed.



Absolutely! The OSC and HTTP API lets you do everything the web app itself can do, including updating routes. Changing a route is essentially realtime though it does take a handful of samples to disconnect and hook it up.







We are listening—sorry I haven't responded before now. We're trying to find the best way forward! We make firmware updates on a fairly regular basis that not only fix issues but add major features, e.g. increasing the channel count over USB from 24/24 to 64/64 at 1x rates and adding optical S/PDIF support. Much of this is driven by feedback from you all, so thank you for pointing out issues like this!

I spoke with the DSP team and I think we've got some good solutions in the works. We're going to add a preference to disable the auto makeup gain globally. As well, we're planning on changing the built-in presets to have a default ratio of 1:1 instead of 2:1.
Great thanks for clarifying this. very useful info. the specs of these units is pretty unique in this price range so if the little issues will sorted I'm definitely going to get a few.
Old 9th April 2015
  #1412
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emrr's Avatar
Quote:
Originally Posted by mrmiller View Post
We're going to add a preference to disable the auto makeup gain globally. As well, we're planning on changing the built-in presets to have a default ratio of 1:1 instead of 2:1.
Sounds like a step forward, thanks.

I still haven't seen any comment about direct hardware playthrough for DP. Anything there?
Old 10th April 2015
  #1413
Here for the gear
 

If I have my Macbook Pro located in a separate room and attached to my 1248 via Ethernet cable, is there a way to monitor the signal with the headphone jack on my MacBook without loading a DAW program?
Old 10th April 2015
  #1414
Quote:
Originally Posted by emrr View Post
I still haven't seen any comment about direct hardware playthrough for DP. Anything there?
We're actively thinking about ways of adding a similar feature to DP for the AVB line. The architecture is so different compared to the CueMix FX boxes, though, that it's a little tricky and takes some doing.

Quote:
Originally Posted by mamerica View Post
If I have my Macbook Pro located in a separate room and attached to my 1248 via Ethernet cable, is there a way to monitor the signal with the headphone jack on my MacBook without loading a DAW program?
You can route whatever's going to your Monitor Outs back into your computer as well. The complication is then listening to it. It doesn't need to be a full-fledged DAW, but you need something that will allow you to route an input to your computer's headphone outs. Even the built-in mic isn't audible unless some software is playing it back to the outputs.
Old 10th April 2015
  #1415
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emrr's Avatar
Quote:
Originally Posted by emrr
I still haven't seen any comment about direct hardware playthrough for DP. Anything there?
Quote:
Originally Posted by mrmiller View Post
We're actively thinking about ways of adding a similar feature to DP for the AVB line. The architecture is so different compared to the CueMix FX boxes, though, that it's a little tricky and takes some doing.
Thanks.

I notice an issue on my system with 'monitor through efx': I cannot open a plug window in DP while recording without it leaving a glitch in what's being recorded. It's impossible to do any sort of background prep work while audio is coming into the system. High buffers, very low CPU usage, etc. This was not true with 'direct hardware playthrough', you could multi-task and preload for expected needs, so they'd be ready to adjust on first playback. I struggle to understand the reason for this glitch, it would seem incoming recorded audio would be isolated from any processing streams, since that is the monitoring side, downstream on the flowchart.
Old 10th April 2015
  #1416
Here for the gear
 

The trim controls of the 1248 and 16a analog outputs are variable from 0 to -24 dB.

Are the trim controls analog or digital? If digital: do they work though the DAC's internal data path?
Old 10th April 2015
  #1417
Quote:
Originally Posted by skjensen View Post
The trim controls of the 1248 and 16a analog outputs are variable from 0 to -24 dB.

Are the trim controls analog or digital? If digital: do they work though the DAC's internal data path?
The output trims are digital and use the DAC's internal trim.
Old 11th April 2015
  #1418
Here for the gear
 

Don't mean to change the subject but a friend that knows I use MOTU equipment sent me a translated quick review (I guess) of the 1248 from the German Sound & Recording magazine:
Audio Interface › SOUND & RECORDING That link is for those of you who can read German. College German was a long time ago for me!

So here's the translation he sent me: Written by Dr. Andreas Hau , March 30, 2015

"Looks like a normal audio interface, but inside lies a revolution. AVB agent technology can be networked easily and remotely controlled via a Web browser multiple interfaces. Full "on the side" MOTU having a "State of the Art" audio technology incorporated - all at an affordable price. We should look at!

Achieving loop test (output connected to input) AD + DA conversion in the sum of a dynamic of 117.5 dB; the total distortion is only 0.0002% - the best value we've ever found!"

Old 11th April 2015
  #1419
Gear Head
 

[QUOTE=akeel;10958105]16A user here, had the unit for 2 months, switched over from a lynx aurora.
No complaints, solid drivers and the unit is simply one of those that you plug in and forget about, workflow with this is very smooth.

Question...
Running cubase 8/pro tools on a windows 7 64 PC am I able to daisy chain two of these units for 32 I/O yet? Connection via Usb[/QUOTE

Now that you're using your motu avb after switching from an aurora, will u consider the sound of the motu as superior to the aurora or is it just like the difference between apple and orange based on personal taste? Do you care to describe the difference between the two in sound quality as 'subjectively' perceived by you?
Old 11th April 2015
  #1420
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dandeurloo's Avatar
Quote:
Originally Posted by mrmiller View Post
The output trims are digital and use the DAC's internal trim.
Sorry if this is a stupid question. At what level should they and the adc be set at for the technically cleanest sound?
Old 11th April 2015
  #1421
Here for the gear
 

reverb

I'd like to add my voice to those who are disappointed with the reverb. It just doesn't seem to make any sense to me and I really have a hard time getting to sound decent. Someone else made a comment about the compressor like "whoever designed this has not been using studio equipment"... the reverb seems the same. Please look at common reverb controls and stick to a proven design !

If someone has figured out how to get this thing to behave, please share some of you techniques for dialing it in. Thank You.
Old 11th April 2015
  #1422
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emrr's Avatar
Quote:
Originally Posted by bduke View Post
I'd like to add my voice to those who are disappointed with the reverb. It just doesn't seem to make any sense to me and I really have a hard time getting to sound decent. Someone else made a comment about the compressor like "whoever designed this has not been using studio equipment"... the reverb seems the same. Please look at common reverb controls and stick to a proven design !
I'm not wild about the way the reverb sounds, but it works, and I actually find the controls to be fairly standard, with the hi and mid controls a nice extra touch beyond the standard 'high damping' control. In my experience the reverb has to be set with much less reverb time than default. I'm not sure who uses 60 second reverb time, or even anything over 5-6 seconds frequently. Stereo spread is nice to have, and modulation takes you more into Eventide territory. Not that it sounds like Eventide.

Back to the compression offerings. I would seldom put an LA-2A type optical leveler on a group or mix bus. Vocals and guitars, maybe, sometimes. It'd be a ton more flexible if you could select the compressor or the leveler for either position, versus being locked into the places they are now. Might be a DSP bottleneck. Me, custom presets for starting would be RMS compressor on everything. Switch to peak mode if needed for protection or efx use, switch to leveler if it's the specific color you want.

None of that is more important than Direct Hardware Playthrough!
Old 11th April 2015
  #1423
Here for the gear
 

Quote:
Originally Posted by mrmiller View Post
The output trims are digital and use the DAC's internal trim.
This is great, it means that output can be trimmed without audio degradation!!!

Are there other ways of accessing the DAC's internal volume control?

Is it possible to link trims for several channels to scale them up and down in parallel?
Old 12th April 2015
  #1424
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Cornvalley's Avatar
A nice addition would be a talkback mic assignable to any one of the analog ins. Add dim feature to an assignable keystroke. Also a mono button on the Main Outs would be helpful.
Old 12th April 2015
  #1425
Gear Head
 

Quote:
Originally Posted by Cornvalley View Post
A nice addition would be a talkback mic assignable to any one of the analog ins. Add dim feature to an assignable keystroke. Also a mono button on the Main Outs would be helpful.
+1
Old 12th April 2015
  #1426
Gear Head
 

I am considering buying one of these. I actually started a thread, choosing an audio interface, but saw this and since it is relevant I thought I would post.

First off, I am an amateur. Well I do get paid for playing out on occasion but not as a studio tech. So I do question the wisdom of spending this much on an interface. It's definitely a want, not a need.

Here are the things that make this appealing to me.

1. Thunderbolt. It's new, it's fast. Hopefully it will be around for a while.

2. 16 line level inputs that will interface nicely with my mixer which has 16 direct outs.

3. Decent metering.

One thing that I see as a potential problem is my PC is in another room to keep the heat and the noise over there. It looks like a $300 investment in a suitable cable in addition to the cost of the unit. That is a critical consideration for my amateur budget.

The second concern is that while this thing is feature packed I don't know that it will outperform, audio quality-wise, other interfaces that cost significantly less. For example the Allen & Heath ICE-16 which uses Firewire. The unit costs less and the cable for the A&H will cost about $30 and Firewire is theoretically better than the usb 2 interface I'm using now although I have read that Firewire can be finicky and is a dieing standard.

There are much cheaper alternatives as well like the Presonus Firestudio Project. I can get 2 of those for around $800. I have no doubt both the Motu and A&H are higher quality but does that actually produce a better result especially considering it will be used very lightly in my home compared to a pro studio.

In fact my existing Tascam us-1800 is a 24/96 device. Sure it is a budget piece of gear but will there actually be a significant improvement in audio quality? Is the increase in dollars more a matter of function than audio quality? Will it make a difference when it is connected to my other lower end gear like my Mackie 1604-vlz pro board?

Obviously I am not the demographic that Motu is aiming for with this device. Still I am an audio junky and this thing looks way too cool.
Old 13th April 2015
  #1427
Here for the gear
 

Quote:
Originally Posted by bduke View Post
I'd like to add my voice to those who are disappointed with the reverb. It just doesn't seem to make any sense to me and I really have a hard time getting to sound decent. Someone else made a comment about the compressor like "whoever designed this has not been using studio equipment"... the reverb seems the same. Please look at common reverb controls and stick to a proven design !

If someone has figured out how to get this thing to behave, please share some of you techniques for dialing it in. Thank You.
I am bummed that no one has a good suggestion for shaping this reverb to sound good. Seems like a big step backwards in sound quality from the MK3 MOTU devices I have. At this point I am thinking I will just stick with the ultralight MK3 since it sounds better for mixing guitar and voice with effects. All the other improvements sound awesome on paper, but now using the ultralight avb for weeks as a live mixer, the mix still sounds better with the uktralite MK3 since the reverb is an important part of the sound with acoustic guitar and voice.

I have been watching this thread hoping for a response from Mr. Miller ensuring us that the reverb is just as good in the AVB devices and with some tips for tweaking the reverb sound. Still no feedback aside from a few people saying they don't care about the DSP. At least I know from the several other responses here that others are also unhappy with the AVB reverb sound so I am not imagining things. Can anyone offer help about the reverb before I return or sell the ultralite AVB?
Old 13th April 2015
  #1428
Here for the gear
 

Quote:
Originally Posted by Guitarmuzic View Post
I am bummed that no one has a good suggestion for shaping this reverb to sound good. Seems like a big step backwards in sound quality from the MK3 MOTU devices I have. At this point I am thinking I will just stick with the ultralight MK3 since it sounds better for mixing guitar and voice with effects. All the other improvements sound awesome on paper, but now using the ultralight avb for weeks as a live mixer, the mix still sounds better with the uktralite MK3 since the reverb is an important part of the sound with acoustic guitar and voice.

I have been watching this thread hoping for a response from Mr. Miller ensuring us that the reverb is just as good in the AVB devices and with some tips for tweaking the reverb sound. Still no feedback aside from a few people saying they don't care about the DSP. At least I know from the several other responses here that others are also unhappy with the AVB reverb sound so I am not imagining things. Can anyone offer help about the reverb before I return or sell the ultralite AVB?
I've made some progress with settings and probably the most important thing to remember is that you are using the high pass filter and other EQ controls in the reverb's channel strip to provide the dampening that you need. I'm just so used to dampening/diffusion/density controls that are tuned/optimized for the reverb that it took me a while to wrap my head around it. The high and mid controls are starting to grow on me and I've been able to get some creative and interesting sounds but I'm not feeling satisfied quite yet... maybe my expectations are too high.
Old 13th April 2015
  #1429
tft
Gear Nut
 

all my tests to tweak this avb mixers reverb are not very promising, at least for me.
when using the eq/leveler on the reverb-bus, i can bend the sound to something acceptable.
but why so much hassle and dsp resources for just getting a reverb sound decent?

in my understanding, it should sound natural and unobtrusive just out of the box, then get tweaked to sound special, if that's needed.

some specifics:
- to get a good mono to stereo reverberation with sufficient stereospread, i have to crank the spread and modulation all the way up, but then it waves around way too fast and obvious, throwing the signal from left to right. it sounds as if this is just a simple modulated panner.
to make this design more useful it needs at least an additional time parameter for the modulation to make it slower or faster.

- the different frequencyranges to tweak the reverb are in theory a good addition, but how they are implemented leaves questions. normally you would like to tweak low range and highrange seperately, to reduce muddiness/boom and tame harsh high reflections for example.
but why do we have mid and high?
i really don't get it, why the low range is fixed and the mid and high can be reduced. this is against all logic in the context of reverb.

- why aren't there different reverb models to choose from, to fit different needs?
seems like an easy way to get this device to sound better, maybe ...


as mentioned by some here, other motu devices seem to have decent reverb, which makes this design even more of a question.

as said earlier, i still like what this unit does for me in all other regards (i have a 112D). all the connection options, routing, the whole mixer apart from the reverb (update for compressor is coming (automakeupgain switchable), as stated by mr. miller).
it is so well designed in all aspects, that i just need to call for "DECENT REVERB"

rant over, peace!
Old 14th April 2015
  #1430
Here for the gear
 

Quote:
Originally Posted by bduke View Post
I've made some progress with settings and probably the most important thing to remember is that you are using the high pass filter and other EQ controls in the reverb's channel strip to provide the dampening that you need.
Scratch the idea of using the high pass filter since they didn't include it in the reverb's channel strip. I noticed this when I had some content with a considerable amount of bass. Even after using the low shelving at -20 I still had some low level rumble (when monitoring the reverb's wet signal). I could use one of the mid filters to further reduce the rumble but it feels like a hack.

MrMiller, please consider adding the high pass filter to the reverb and group's channel strip.
Old 14th April 2015
  #1431
Here for the gear
 

Hi mrmiller, i have been following this thread (for what seems like forever) from the start since i began looking for a whole new DAW/interface-setup. It is truly very encouraging to see someone knowledgeable like yourself here to take care of the customers questions (misconceptions and fears). This is my first post here, so bare with me...

I am now at a crossroads and need to commit into a purchase. I have been holding off as long as possible in order to get some more info about the Thunderbolt support on Windows. Im building a live-rig where the lowest possible input-->output latency is of critical importance. Are there any news or advances regarding TB/Windows. I know drivers will not probably roll out tomorrow, but if i could get some kind of indication of the status of win/TB, then the 1248 is still my no. 1 contender.

In this live-rig i might have to use a second PC to do some of the heavy VST-amp-sim lifting since i will be at 24/96. Is it possible to use the ADAT´s of the 1248 to connect to a second PC in a "real time roundtrip matter"? What i want to do is this:

1. Connect the whole band into the pre´s of the 1248. Send some of these channels to the "main DAW" that the 1248 is connected to.
2. Send the rest of the inputs directly to the ADAT-outs (bypassing DAW/computer) and going into the second PC´s ADAT-in for real-time (well, low latency) processing there.
3. Those processed signals then get routed back into the 1248/main DAW via second PC´s ADAT-out into 1248´s ADAT-in, and then merged with the processed audio of the main DAW at stereo out.

Presuming i have the same latency and buffer settings on both computers, would this work? The routing in itself is purely digital, so that would not result in any latency due to AD/DA, right? Since its not a matter of splitting individual instruments across the computers i guess i would not have to deal with any phasing issues?

Thanks mrmiller and everyone else for this great thread! Lets keep it rolling.
Old 14th April 2015
  #1432
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swartzfeger's Avatar
It's down to the 1248 and the Ensemble 2 for me, the biggest thing being for me is low-latency monitoring. I usually program a scratch track in SS4 then DI bass and guitar over that. For whatever reason, my timing with previous setups was always terrible unless I bounced my track in Logic X. And I don't want to be forced into screwing around with a software mixer like I was with my Apollo. That's part of the Ensemble's appeal -- plug it in, arm track in Logic, done. Is the 1248 the same?

Looks like I'll be starting fresh with page 1 to see if there are any comparisons with Ensemble's DMA latency (hint hint) :D
Old 14th April 2015
  #1433
Gear Maniac
 

So, does anybody have experience with the new high channel count usb driver in windows?
. Does the new usb driver work with heavy sessions or do you get dropouts? I am using cubase with orchestral sessions and would like to sum 32 tracks to my summing mixer....

Can the new motu line deliver high performance with low latency in usb?
Old 14th April 2015
  #1434
Here for the gear
[QUOTE=Sammix;10962960]
Quote:
Originally Posted by akeel View Post
16A user here, had the unit for 2 months, switched over from a lynx aurora.
No complaints, solid drivers and the unit is simply one of those that you plug in and forget about, workflow with this is very smooth.

Question...
Running cubase 8/pro tools on a windows 7 64 PC am I able to daisy chain two of these units for 32 I/O yet? Connection via Usb[/QUOTE

Now that you're using your motu avb after switching from an aurora, will u consider the sound of the motu as superior to the aurora or is it just like the difference between apple and orange based on personal taste? Do you care to describe the difference between the two in sound quality as 'subjectively' perceived by you?
Hi Sammix,

To provide some context...

The motu is being used to send 16 channels out into a desk. I have used the Lynx for a number of years and have never really had any major issues with it, maybe some small niggles here and there but nothing major. I was rather hesitant in switching to the Motu as I had gotten some negative feedback from a number of people as well as having read some bad reviews online (For older Motu units).

Ideally I needed upwards of 32 channels running out into the desk, cost wise this would have been a significant outlay on what is now a rather dated pair of Lynx units.

So I bit the bullet on the Motu and have not looked back. The sound quality to my ears (This is subjective, some people may completely agree and others may totally disagree) is Not a night and day difference, I would go so far as to say that they are definitely comparable. The lynx for me has a hyped High end although it definitely has a very open sound. The Motu on the other hand is very transparent and open while translating the lows well, the high end is less hyped. I believe this is a matter of taste rather than a simple "one is better than the other".

The bottom line for me was simply the pure value for money I was getting on what is "New technology" and a far greater number of channels of conversion.

I am now planning on adding a 24ao to the 16a, I believe this says it all...
Old 15th April 2015
  #1435
Gear Maniac
Quote:
Originally Posted by swartzfeger View Post
It's down to the 1248 and the Ensemble 2 for me, the biggest thing being for me is low-latency monitoring. I usually program a scratch track in SS4 then DI bass and guitar over that. For whatever reason, my timing with previous setups was always terrible unless I bounced my track in Logic X. And I don't want to be forced into screwing around with a software mixer like I was with my Apollo. That's part of the Ensemble's appeal -- plug it in, arm track in Logic, done. Is the 1248 the same?

Looks like I'll be starting fresh with page 1 to see if there are any comparisons with Ensemble's DMA latency (hint hint) :D
I can't answer your question for sure, but I would imagine the answer is "yes".

One big difference between the 1248 and the Ensemble is the 1248 only has four mic pres, whereas the Ensemble has eight pres.

Personally that makes a pretty big difference to me. So that has me deciding between the Ensemble and the 8m.
Old 15th April 2015
  #1436
cbm
Gear Head
 
cbm's Avatar
 

Quote:
Originally Posted by swartzfeger View Post
That's part of the Ensemble's appeal -- plug it in, arm track in Logic, done. Is the 1248 the same?
The latency w/ the 1248 is as good as, or better than, anyone's, I think. Within fractional ms, at any rate.

As far as the plug&play nature, there is one extra step needed for initial setup of the 1248. This involves downloading the discovery app, and selecting the preset that you want. The discovery app queries the network to find MOTU AVB devices, and opens the confing page in your browser. I was a little dubious about using a browser for this sort of thing, but in practice it's been really nice. I've been using the web-based mixer control to allow for players to grab individual monitor mixes.
Old 16th April 2015
  #1437
Here for the gear
 

Quote:
Originally Posted by swartzfeger View Post
It's down to the 1248 and the Ensemble 2 for me, the biggest thing being for me is low-latency monitoring. I usually program a scratch track in SS4 then DI bass and guitar over that. For whatever reason, my timing with previous setups was always terrible unless I bounced my track in Logic X. And I don't want to be forced into screwing around with a software mixer like I was with my Apollo. That's part of the Ensemble's appeal -- plug it in, arm track in Logic, done. Is the 1248 the same?

Looks like I'll be starting fresh with page 1 to see if there are any comparisons with Ensemble's DMA latency (hint hint) :D
I'm a new 1248 user. This was also of great interest to me. I can tell you that the 1248 has slightly higher latency than the Ensemble 2 by about 30 samples. I normally record at 48kHz and that means an additional latency of 0.625ms vs the Ensemble 2 at that sample rate. It isn't a huge difference, and I feel there are several advantages such as DAC quality where the 1248 bests the Ensemble 2. I could be wrong, but I don't see any technological reason that the MOTU 1248 couldn't boast the same low latency that Apogee advertises. DMA is just a fancy marketing term for what Thunderbolt provides inherently. MOTU has historically used a safety buffer as part of their drivers, and I believe that's where this 30 sample discrepancy comes from.

Perhaps Mr Miller could offer some insight as to the difference.

I do see that MOTU recently released a new driver that significantly reduced the latency of this unit to what I am reporting now. This type of continued development is much appreciated. Perhaps we could see parity between the Ensemble 2 latency and the 1248 in the future?

From my own perspective, recovering something as small as those 30 samples could really help when using something like UAD plugins in live mode with Pro Tools. My UAD Thunderbolt Satellite Octo requires its own safety buffer of 32 samples each way (64 roundtrip), so all of those little buffers can add up in a hurry if you are very sensitive to latency. I have workarounds such as using direct monitoring when it presents issues. Still any additional improvements would help. A slight improvement here could eliminate the advantage of the UAD Apollo for example. I prefer the converters of the 1248 to the Apollo myself, plus routing and channel count is much better, so this wouldn't be a trivial feature for UAD plugin users I believe. I hate to nit pick over 30 samples, but this is gearslutz and most of us obsess over the last 5-10% of performance per routine.
Old 16th April 2015
  #1438
Here for the gear
 

I am curious if any 1248 users have experimented with an external WC such as the Apogee Big Ben or Antelope Isochrone OCX. Do you notice any improvement over the 1248 internal clock?
Old 16th April 2015
  #1439
cbm
Gear Head
 
cbm's Avatar
 

Quote:
Originally Posted by MediaBrain View Post
I hate to nit pick over 30 samples, but this is gearslutz and most of us obsess over the last 5-10% of performance per routine.
Unless you're tracking in phones, I don't see how a 30 sample latency could make a difference. Even with phones, that still seems like too small a difference to care about. That's the same as moving your head about half a foot further from a sound source.
Old 16th April 2015
  #1440
Here for the gear
 

Quote:
Originally Posted by cbm View Post
Unless you're tracking in phones, I don't see how a 30 sample latency could make a difference. Even with phones, that still seems like too small a difference to care about. That's the same as moving your head about half a foot further from a sound source.
Under normal conditions this isn't a problem, and 30 samples by itself certainly isn't a killer. But there are certain situations right on the boundary of being acceptable. For example certain VIs don't behave well at low buffer settings. Or in the case of UAD "live mode" there is that annoying safety buffer of yet another 64 samples roundtrip plus whatever the plugin requires. In these scenarios, increasing the buffer or incurring UAD safety buffer plus plugin delay already pushes latency to the edge of an acceptable limit.

So yes, by itself 30 samples isn't a huge deal. It's all of the little system delays that add up. 0.5ms being ~ half a foot isn't a big deal for a virtual piano, but when cumulative delays add up to several milliseconds it can be problematic for a sensitive player. If you are used to hearing player perspective from the piano bench, it can be frustrating when the piano suddenly sounds 6-8ft away from you. Vocalists tend to be much more sensitive as well. For a guitar player through an amp, this wouldn't be an issue as that distance would seem pretty normal.

Workarounds have been using a separate laptop to run the VI (treating it like an audio instrument and recording midi data in parallel for later sound tweaking), holding off on UAD plugins for post work, direct monitoring, etc. I typically use a HEAR system following an analog monitor board pulling signals directly for the talent. I can apply effects or inserts this way with virtually no latency and also fulfill the occasional funky request without printing.

I realize those 30 samples alone won't solve these issues that we have to work around as engineers, but if everyone improves product performance a little bit.... First world problems I know. We are leaps and bounds from where we were even a decade ago.

Last edited by MediaBrain; 16th April 2015 at 04:35 AM..
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