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Clint Martin

Lowest Latency USB interface?

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My Onyx 1200F just gave out, but I've been using a Zoom UAC-8 as my primary and just routing the extra Onyx inputs into it via ADAT.

I like the Zoom and it works well for me but since I need extra inputs I went for the Presonus Quantum which blazes at RT latency around 3 msec at 64 samples and now I run the Zoom inputs into it via ADAT.  But it costs a $ grand and uses Thunderbolt, so you need to be sure you have a good MOBO with a thunderbolt header and a matching AIC card made by the same manufacturer.  Thanks to Jim Roseberry for his encouragement and tips on this issue.

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On 2/3/2019 at 6:57 AM, Twisted Fingers said:

Jim this is very useful information for me. I have an ASUS X99 DELUXE II motherboard with the ASUS/INTEL ThunderboltEX 3 PCIe card. It's good to hear that, once the prices drop to within my price range, I'll be able to use a Thunderbolt interface and benefit from the extremely low latencies they are capable of.

That combination works great.  Used it myself for a good while...

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On 1/30/2019 at 11:52 PM, Jim Roseberry said:

USB audio interfaces with lowest round-trip latency are RME and MOTU (both sub 4ms).

 

I used to have a MOTU 2408 Mk3 and 24I/O connected via PCIe-424 audiowire.  Both would run at very low latency and if the product wasn't too loaded, I could run at 64 samples.  When I sold my studio, I purchased a 896Mk3 Hybrid, as I wanted some built-in preamps, plus I add more via ADAT when I need them.  I have never, ever been able to get anywhere near 4ms whether connected via USB or FireWire (TI Chipset).  I find I have to run with 256 samples at a minimum and often more to avoid dropouts or distortion.  I really miss the low latency I got with the audiowire gear.

However, if you're saying you can get sub 4ms with the MOTU USB stuff, maybe there is hope?

Only reason I have steered clear of the new MOTU interfaces is the crap-shoot that is thunderbolt on Windows.  Also, I currently run a Ryzen rig, which, of course, does not support thunderbolt.

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Wile were talking Thunderbolt Cards /Motherboards : Im running  A Asus z370a motherboard.. I originally  purchased the Asus EX3 thunderbolt card and after a little fiddling got it up and running.. but I had read that the Gigabyte thunderbolt card was easy to install and offered more features ..so I orderd one

https://www.gigabyte.com/Motherboard/GC-ALPINE-RIDGE-rev-20#ov

the connectors are much better and the unit came with cables to hook up to asus gigabyte and asrock motherboards .. this unit is rock solid  ..1 driver ..there is also  : https://www.gigabyte.com/Motherboard/GC-TITAN-RIDGE-rev-10#kf

Titan ridge offering even more features !.. I would seem that Universal Audio is onboard with Gigabytes and supporting them : this board came up as a Has It All Board ! https://www.gigabyte.com/Motherboard/Z390-DESIGNARE-rev-10#kf

Note As Jim said if your running a thunderbolt ll interface ..as I am you need a  converter : https://www.startech.com/ca/Cables/thunderbolt-3-cables/thunderbolt-3-usb-c-thunderbolt-adapter~TBT3TBTADAP       I payed 85.00 cad for mine !

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Since updating to the new Universal Control driver my old Presonus 44VSL can easily work at 64 buffers, and that's with my PC online with Windows firewall active.

At 48khz I believe 5.3ms roundtrip is pretty impressive.

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Sitting in my chair now, I have more latency between me and my amp than 5 ms.

 

Super low latency is cool though. I appreciate it. For me, I love using synths and using tons of voices with low latency drives the processor hard b/c the voices all show up on one core.

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4 hours ago, Bill Ruys said:

I used to have a MOTU 2408 Mk3 and 24I/O connected via PCIe-424 audiowire.  Both would run at very low latency and if the product wasn't too loaded, I could run at 64 samples.  When I sold my studio, I purchased a 896Mk3 Hybrid, as I wanted some built-in preamps, plus I add more via ADAT when I need them.  I have never, ever been able to get anywhere near 4ms whether connected via USB or FireWire (TI Chipset).  I find I have to run with 256 samples at a minimum and often more to avoid dropouts or distortion.  I really miss the low latency I got with the audiowire gear.

However, if you're saying you can get sub 4ms with the MOTU USB stuff, maybe there is hope?

Only reason I have steered clear of the new MOTU interfaces is the crap-shoot that is thunderbolt on Windows.  Also, I currently run a Ryzen rig, which, of course, does not support thunderbolt.

The 896mk3 Hybrid interface (and all the Hybrid series) was (round-trip latency wise) a step backward from the original 896HD.

The original 896HD yielded 5ms round-trip latency at a 64-sample ASIO buffer size 44.1k

The Hybrid series added onboard DSP processing/mixing... and that increased round-trip latency to ~6.5ms at those same settings.

 

To get sub 4ms round-trip latency from MOTU USB, you have to be running one of the newer AVB models (or spin-offs).

The newer MOTU USB drivers allow you to tweak the safety-buffer size.

 

FWIW, Thunderbolt under Windows is not a crap-shoot. 

You just have to make sure you've covered all the details.

MOTU was one of the  first companies to have (release version - not beta) Thunderbolt drivers for Windows that support "PCIe via Thunderbolt" (allowing PCIe level performance).

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I found that using an interface with a DSP chip and onboard effects solved my latency problems. Of course, it depends what you are using the DAW for. For guitar, I tried amp sims but latency destroyed that idea, I also didn't like the sound. My computer is old, 4th generation i5, 8gig ram. Now I use pedals to get the sound and run it into my Steinberg UR44 where I can route it out direct to the headphone while recording, no latency, I can also add reverb to my headphones, since I don't use pedals for that and add it in afterwards from the DAW. I can also use a HPF and some equalization/compression, through the UR44 before it hits the DAW, used sparingly it's great.

For keyboards/drums, I use a nektar midi keyboard controller and Kontakt/Komplete and other VST's. Unfortunately, there is no way to avoid the latency with this setup. However, I have found the Steinberg drivers to be the best so far out of what I've tried on my system so I have no complaints. Don't really notice it, whereas i did notice it on the other audio interfaces i tried.

For vocals/harmonies, same as guitar, record straight in add compression or equalization to suit and any reverb in the phones, output through the DSP chip and no latency.

I could never get the ease of use and sound I wanted from USB powered interfaces. I prefer the wall wart powered, DSP mixer and effects chip type interfaces (generally, if they have a DSP chip, they need external power). But again, it depends what you are doing with the DAW. This is the best setup I have found for my uses but it may not suit others. It's also not that expensive to set up. More money for mics, guitars/pedals and VST's.

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2 hours ago, Tezza said:

For keyboards/drums, I use a nektar midi keyboard controller and Kontakt/Komplete and other VST's. Unfortunately, there is no way to avoid the latency with this setup. However, I have found the Steinberg drivers to be the best so far out of what I've tried on my system so I have no complaints. Don't really notice it, whereas i did notice it on the other audio interfaces i tried.

In the case of playing soft-synths, you're dealing with one-way (playback) latency.

Monitoring audio tracks in realtime thru software EFX/processing, you're dealing with full round-trip latency.

The UR44 (IIRC) yields about 7ms total round-trip latency at a 32-sample ASIO buffer size 44.1k.

 

Some of the absolute latest generation of AmpSim plugins are sounding pretty decent.

  • Helix Native sound pretty good.
  • The new version of TH-U (from Overloud) is going to have a feature similar to Kemper's ability to "profile" actual mic'd amps/cabs.
  • PRS Super Models sounds good (IMO) if you use different Cab IRs

 

Onboard DSP to process/route/mix/loop-back-record can be extremely useful (if you use it).

If you're after lowest possible round-trip latency, onboard DSP will slightly increase it.

Part of the reason Quantum can achieve such low round-trip latency; it has zero onboard DSP for routing/mixing/loop-back-recording.

All monitoring has to be done via software.

 

Why so fixated on lowest possible round-trip latency?

In the case of Quantum, since all monitoring has to be done via software, it's critical.

Lets say you're using a Kemper Profiling amp... or something like Helix or HeadRush (all hardware guitar amp sims).

The Kemper itself can have up to 4ms round-trip latency.

The whole point of an audio interface like Quantum is to keep round-trip latency as low as possible.

If Quantum were yielding 4ms latency, add the Kemper's 4ms latency... and you're at 8ms total round-trip latency (while monitoring guitar).

That's a significant step backward compared to hardware based monitoring (8ms vs. near zero).

Since Quantum can actually get down to 1ms total round-trip latency, even with Kemper's worst case scenario, you're at 5ms total round-trip latency.

At 1ms total round-trip latency, Quantum makes software based monitoring effectively on-par with hardware.

 

Monitoring via software at 1ms round-trip latency hits the CPU hard.

High CPU clock-speed is critical... as this isn't a process well-suited for multi-threading.

 

 

 

 

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I have an obsolete Echo Layla 3G that uses a PCIe interface and I'm happy to get 11.5 ms rountrip latency recording at 24/96! The latency doesn't bother me at all when I'm recording guitar tracks, as an example.

Reading the low latency some of you are seeing with USB interfaces is really encouraging, as some day I will need to replace my audio interface and I already saw that current PCIe audio interfaces are out of my budget.

Edited by razor7music

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I have an Echo Layla 3G that I was happy with for a LONG time.  It still works, except for the Phantom power.  I'd love to sell it but not sure what to list it for at this point since Echo Audio stopped making such things.

Since I jumped ship from Sonar to Studio One and I had sold a bunch of other stuff on eBay and I got a great deal (at the time) on a Presonus Studio 1824 (new last year), I went with that.  In the end, I suppose it is "better" than the Layla in that the overall latency, even in a completed project is better.  I don't really thin kabout the latency and having to switch between multiple different settings.  In Studio One, I just leave it at 64 samples at 24/44.1 and don't get bothered with it through my simi-simple projects.  In Cakewalk, I have to move that to 128 samples to be usable for even just a single (same)  softsynth (SampleTank3).

Numbers that are reported through the software are:

Studio One at the above 64 samples at 24/44.1 = 2.954 Input + 2.49 Output = 5.44

This is workable for me playing guitar and bass and recording vocals.  Studio One has the option of what they call Blue Z and Green Z monitoring, but I honestly never bother with them - it's just a bit too complicated, really.

In Cakewalk, I can basically double those numbers as I have to go to 128 samples if I want to do any SampleTank3 stuff.  Since that is my primary meloduic Synth (along with Arturia Analog Lab), that's what I'm dealing with.

In Cakewalk, the screen at 64 samples says "effective latency is 1.5ms".  It's not usable and I don't really understand how 2.9+2.5 somehow is 1.5, but whatever.  It's not real.

At the end of the day, for ME (a not very great guitar player), any of these latency numbers are acceptable.  It's only the edrums where it is completely intolerable for me, even in Studio One at 64 - though I can go to 32, which is still too "soft" for drumming VSTs to me.  I have a Pearl Mimic Pro, which is effectively the same as a VST for monitoring and really, recording, since I loop the recorded MIDI (after massaging) back to audio in multi-track audio anyway.  The thing sounds fantastic and pretty much eliminates any drum playing VST issues I might have.  I used to just use the TD-20, but now with the Mimic I kind of have the best of both worlds.

I REALLY wanted to get the Quantum and a Thunderbolt card for the PC, but I was making the purchase on sold "stuff I had in the audio shed" and the Quantum + card was outside of that reach.  I still have a few things around to sell, like the Layla, an original Yamaha DSP Factory DS2416 card (which sells for a decent amount still on eBay), a pair of Roland VH-12 hi-hats, etc.  Between all of that stuff, I might have enough to get a Quantum + Thunderbolt card (if I sold the Studio 1824 too), but I might be tempted to just get a few more e-cymbals (like the ATV 18" ride) or something else.  

End result: I think my Presonus Studio 1824 is "good enough" and my dollars might be better spent elsewhere as a single-person-hobbyist. That was a longer post than I expected.

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After reading this thread days ago I bumped my samples to 192 and my buffer to 48. Then I decided I didn't care and went on with life.

Today, I was bouncing a video while listening to playback and I was scratching my head why I heard crackling...

duh lol

48 samples while exporting video didn't work.

The point is, I hadn't heard it in so long I started trouble-shooting rather than just bumping up the buffer. lol

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10 hours ago, Blades said:

In Cakewalk, the screen at 64 samples says "effective latency is 1.5ms".  It's not usable and I don't really understand how 2.9+2.5 somehow is 1.5, but whatever.  It's not real.

A 64-sample ASIO buffer at 44.1k = 1.5ms (This is true no matter what audio interface you're using)

While true (and I know you already know this), that's not telling the whole story.

When you're monitoring in realtime thru software EFX/processing, you're dealing with round-trip latency:

  • ASIO input buffer (1.5ms)
  • ASIO output buffer (1.5ms)
  • A/D D/A (~1ms)
  • Driver's safety-buffer - this is the X-Factor when it comes to round-trip latency and it's often hidden (can vary radically)

In this example, we're already at 4ms... without factoring in the safety-buffer.

If the audio interface is one of the better makes, the safety-buffer will be small and round-trip latency will be ~5-6ms.

If the safety-buffer is large, round-trip latency can be more than double.

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Rendering video makes even a large/dense audio project look light-weight.

IME, The ASIO implementation in Premier, After-Effects, etc is not quite on-par with better audio applications.

 

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1 minute ago, Jim Roseberry said:

If the audio interface is one of the better makes, the safety-buffer will be small and round-trip latency will be ~5-6ms.

Yeah, I have been wondering if I made a mistake going for Roland's OCTA-CAPTURE. I mean, you would thing that Roland would make quality gear but my best RTL at 64 samples is 9.5ms.

But there is no way I can run with that setting and I have to go up to 384 samples giving me a RTL of 34.8ms

Not that I am complaining as I seem to be able to manage at that. I just get a bit envious when I see guys saying they can get down to 64 samples haha! :D

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The Roland audio interfaces have some nice features.

Lowest possible round-trip latency just isn't their forte'.

At a 48-sample ASIO buffer size 44.1k, round-trip latency is 7.4ms.

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23 hours ago, Jim Roseberry said:

In the case of playing soft-synths, you're dealing with one-way (playback) latency.

Monitoring audio tracks in realtime thru software EFX/processing, you're dealing with full round-trip latency.

The UR44 (IIRC) yields about 7ms total round-trip latency at a 32-sample ASIO buffer size 44.1k.

I usually run mine at 1024 buffer and at 96k.  Gives the DAW room to breathe. I get zero latency because I monitor through hardware. This works for vocals, acoustic guitar and electric guitar. I only change it when using VST's for bass/keyboards, then to 64 or 128, works fine. I don't pay a lot of attention to latency figures because I have found that can be deceiving. My Focusrite 2i4 was exceptionally bad here, telling me 9ms when it was more like 20ms! I rely on my ears to let me know what is acceptable.

 

23 hours ago, Jim Roseberry said:

 

Some of the absolute latest generation of AmpSim plugins are sounding pretty decent.

  • Helix Native sound pretty good.
  • The new version of TH-U (from Overloud) is going to have a feature similar to Kemper's ability to "profile" actual mic'd amps/cabs.
  • PRS Super Models sounds good (IMO) if you use different Cab IRs

Thanks for that info, I am sort of over amp sims now though. I like the Jazz/Blues fenderish clean sound, a bouncy amp sound. Sims struggle with that, I did numerous sessions of over 60 tracks at a time testing out various sims (not the ones you mention). The best sound I came up with was from the $50.00 American Sound Joyo pedal I compared them too. There is also the workflow issue, I don't like fiddling around with ir cabs, mouse driven knobs etc. Using the pedals, I don't even look at the computer and I have real knobs to tweak if something sounds off.

 

23 hours ago, Jim Roseberry said:

Onboard DSP to process/route/mix/loop-back-record can be extremely useful (if you use it).

If you're after lowest possible round-trip latency, onboard DSP will slightly increase it.

Part of the reason Quantum can achieve such low round-trip latency; it has zero onboard DSP for routing/mixing/loop-back-recording.

All monitoring has to be done via software.

 

Why so fixated on lowest possible round-trip latency?

In the case of Quantum, since all monitoring has to be done via software, it's critical.

Lets say you're using a Kemper Profiling amp... or something like Helix or HeadRush (all hardware guitar amp sims).

The Kemper itself can have up to 4ms round-trip latency.

The whole point of an audio interface like Quantum is to keep round-trip latency as low as possible.

If Quantum were yielding 4ms latency, add the Kemper's 4ms latency... and you're at 8ms total round-trip latency (while monitoring guitar).

That's a significant step backward compared to hardware based monitoring (8ms vs. near zero).

Since Quantum can actually get down to 1ms total round-trip latency, even with Kemper's worst case scenario, you're at 5ms total round-trip latency.

At 1ms total round-trip latency, Quantum makes software based monitoring effectively on-par with hardware.

 

Monitoring via software at 1ms round-trip latency hits the CPU hard.

High CPU clock-speed is critical... as this isn't a process well-suited for multi-threading.

 

I personally wouldn't monitor through software, first there is the latency issue and then, when you start loading the project up with tracks, plugins and VST's, your throwing all that at the CPU as well as asking it to provide perfect latency. Monitoring through hardware means those things are not going to interfere with your zero latency, your also not going to get the horrid clicks, pops, stutters and crashes etc associated with overloading your CPU.

To be honest, I don't understand the obsession with software monitoring.  I must be missing something. Presonus takes the cake with it's Blue Z, Green Z and Native Z latency monitoring with it's drop out protection etc, then when you get problems you have to turn off plugins, freeze tracks and fiddle with dropout and buffers. Too complex and fiddly for me. 

But again, I guess it depends what you are looking for, I am not sure what Clints needs are. If you are using amp sims, then I can see the obsession with low round trip latency. But if your a solo performer looking to record vocals, guitars, keys in a small studio etc I would recommend hardware monitoring over software monitoring. An interface that has:

Wall wart, not bus power
Min 4 XLR's in
Mic, line in and direct in (instrument)
Pads, selectable HPF and phase switch on each channel
DSP chip mixer (for hardware monitoring)
Smokin great quality Reverb
Compression and Equalization options for headphones/print to DAW
Good drivers that don't lie about latency and have low latency for VST's

Different manufacturers make interfaces that have some or all of the above and they can also be picked up secondhand on Ebay for not much.
 

 

 

 

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2 hours ago, Tezza said:

o be honest, I don't understand the obsession with software monitoring.  I must be missing something. Presonus takes the cake with it's Blue Z, Green Z and Native Z latency monitoring with it's drop out protection etc, then when you get problems you have to turn off plugins, freeze tracks and fiddle with dropout and buffers. Too complex and fiddly for me. 

With the "Hybrid Buffering" approach used by Studio One (Logic and Samplitude had this years ago, ProTools got it at version 11), you don't have the big CPU hit from monitoring thru software. 

  • Tracks that are merely playing back are processed using a much larger buffer size.
  • Tracks that are being monitored thru software are processed using the small buffer (only while being monitored).

This gives you the best of both worlds.

When you can monitor thru software (no glitches) at 1ms round-trip latency, it pretty much makes hardware based monitoring moot.

You can monitor thru AmpSim plugins in realtime... or anything type of processing... and never give it a second thought.

When you go to play a virtual-instrument... one-way Playback latency is incredibly low.

 

 

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I sometimes use Sims, I like to use my VSTs in real time.

I do use my Amplifire 12 mostly these days, but I have my levels set. Hardware monitoring is louder and sounds different.

My Presonus allows me to work the way I want even if the latency reporting in Cakewalk may or may not be true. I'll defer to @Jim Roseberry on that one.

It's great that affordable options exist for those of us that rely on our limited income to find our passions. I bought my 44VSL 6 years ago for $399 and it's still working for me.

I can certainly afford to spend $1000 or so when the time comes to replace it.

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