3.6 When ?

I can’t wait to post my fake latency with my Audient tonight. I bet they’re lying liars too!
Audient uses TheSycon.de drivers. They (TheSycon) most certainly misreported driver buffer size back in 2017 with the Behringer, Midas, and SSL (Edit: and I think the MOTU USB drivers for the 2 and four input MOTU interfaces) drivers.

Do a real loopback test and let's see what they show today
 
Here's an RME Digiface USB, a digital only device, let's measure thru some Focusrite octopre converters

RME reports 1.3 in and 2.4 out at 64 samples for roughly 178 samples

Running a cable results in 183 samples or 3.8ms. That sounds suspiciously close to 178, and probably way too fast for D/A and A/D. Not exactly sure what is going on in this case.

One thing to note as @Sascha Franck said is the absolutely INSANE difference in Round Trip Latency at 64 samples between the RME and Line 6 Helix drivers measured here


RME cable.png
 
RME reports 1.3 in and 2.4 out at 64 samples for roughly 178 samples

Running a cable results in 183 samples or 3.8ms. That sounds suspiciously close to 178, and probably way too fast for D/A and A/D. Not exactly sure what is going on in this case.

Does the RME report the same in case you don't connect the Octopre?
 
Does the RME report the same in case you don't connect the Octopre?
If I just loopback an adat cable it gives 178 samples. The other thing is there's a bit of fuzziness at the start of the return waveform with analog but identical to the output with the digital cable, so it could be a bit smeared
 
Audient uses TheSycon.de drivers. They (TheSycon) most certainly misreported driver buffer size back in 2017 with the Behringer, Midas, and SSL (Edit: and I think the MOTU USB drivers for the 2 and four input MOTU interfaces) drivers.

Do a real loopback test and let's see what they show today
I respect you guys love this shit, but honestly, I don’t really give enough fucks to go through the exercise. The Audient is working fantastic for me, with no additional drivers loaded. Logic is perhaps/maybe/probably lying, but I don’t really care, as long as the results are good.
 
Fwiw, to at least somewhat take this latency discussion back to something HX related: There's two things making me wonder.

1) Latencies are better under Windows (I knew this already, the latest post of Orvillain just remembered me). It's still pretty bad under Windows (most interfaces will get well below 10ms with 64 samples @ 44.1, even if they're TheSycon powered), but there's a whooping 3ms of a difference between Windows and macOS.

2) The best latencies can be had on macOS running the HX devices "driverless", just using Core Audio, built into macOS. Can't recommend this, though, as you might be running into troubles (such as in running multiple audio sources from your computer into the HX or when recording at different sample rates but 48kHz, etc.).
It's pretty weird because typically, as soon as companies provide a dedicated driver, performance will be better than via Core Audio. At least that's the case with every other interface I ever tested.

Maybe some Windows folks may want to check whether ASIO4All is resulting in better performance (or at least better latency figures) as well. With cheap interfaces, this sometimes has been the case (not too sure about today's situation).

Needless to say, I think this could be improved considerably. Not sure whether it's possible within this generation of Line 6 modeling devices, they may as well have just slapped in the cheapest audio interface chip available (absolutely not unlikely, btw.)

---

After all, I am perfectly aware that onboard audio interfaces in modelers are most often (well, pretty much always) something just thrown in as a kind of afterthought. At the very least, modelers are defenitely not built around the idea of them being used as high performance audio interfaces. So all you can hope for is that the interfaces do at least work stable and not cause recording offsets. Otherwise user can just be happy that they're there, having an onboard audio interface pretty often is better than having none.

But imagine there was a better interface in your modeler, delivering the same latency values as just, say, Motus M series (they're possible the best VFM devices these days, as Motu apparently has considerably upped their driver game).
You probably wouldn't have to buy a Tonex pedal to enjoy captures within your HX ecosystem. You'd just get a 2-in-1 convertible laptop (best solution for live players as computer keyboards are the antichrist of stage organisation) and possibly run Gig Performer hosting each and every plugin you'd like on it. Not happy with the reverbs on your modeler? Well, chose one of the tons of (sometimes even free) excellent ones. Or just use an IR based reverb and roll your own. And, as with Tonex, you could just run each and every amp plugin alongside what's coming with the hardware. And in case the main workload was still done by the hardware, even a rather cheap convertible laptop would do, setting you short just around 500 bucks. Add a VST host (for a start, Reaper would likely be just fine) and enjoy each and every VST plugin there is.
For me, that'd be sooo much best of both worlds...
 
And fwiw, if you compare nesting digital devices to adding a computer, in case the interface part was decent, you'd end up with latencies that are on par, less conversions, less cables and what not.
HX family device + Tonex pedal in a loop = around 7ms of latency. That's quite likely already more than what you'd get when using a decent interface under 44.1kHz and 64 samples buffersize.
Add to this that you'd always have your editor available as well.
 
According to DAWBench currently, any interface with a decent driver is going to have a (usually slightly) lower round trip latency number under windows than Mac OS for quite a few reasons, but could change in an instant if apple culture changed. DAWBench Podcast has lots on this sort of thing
 
According to DAWBench currently, any interface with a decent driver is going to have a (usually slightly) lower round trip latency number under windows than Mac OS for quite a few reasons

Yeah, I know. But 3ms isn't exactly "slightly". We're usually talking about values in the 0.2-0.5ms realm, from all I know.
 
Some audio interface drivers report their latency accurately... others do not.
If latency is reported accurately (by the driver), your DAW application's listed RTL will be correct.

Just what I said.
And fortunately, the percentage of interfaces reporting correct numbers has been vastly increasing.
 
Except when the DAW reports round trip latency.

The accuracy of DAW reported round-trip latency depends on the audio interface driver.
If the audio interface driver reports latency accurately, the DAW reported RTL will be accurate.
Believe it or not, many audio interface drivers do not report latency accurately.
This is why you want to perform a loopback test... to verify there's not an "offset" when recording.
Most DAW applications let you manually enter a time offset (in samples) to ensure audio is properly aligned.
 
FWIW, I wouldn't blindly trust any audio interface.
You only need do a loopback test one time (to check for any offset).
If there's not, you're good-to-go.
If there is an offset (between previous and newly recorded audio), you need to measure it (in samples)... and manually enter it in your DAW.
Once done, you don't need to think about it.

I'd do a loopback test anytime you switch audio interface.
 
The accuracy of DAW reported round-trip latency depends on the audio interface driver.
If the audio interface driver reports latency accurately, the DAW reported RTL will be accurate.
Believe it or not, many audio interface drivers do not report latency accurately.
This is why you want to perform a loopback test... to verify there's not an "offset" when recording.
Most DAW applications let you manually enter a time offset (in samples) to ensure audio is properly aligned.

Fwiw, recording offsets aren't necessarily related to false RTL numbers. With my cheesy Zoom G3, the reported latencies are exactly the same as what I measured manually, yet it's causing huge recording offsets (435 samples).
 
If you're running a digital only audio interface, there's no way for its driver to accurately report latency.
It can't factor the latency of your A/D D/A.
 
Checking for offsets is the first thing I'm always doing, too. The Helix Floor in Core Audio mode (hence running driver-less) was causing around 90 samples offset, with installed drivers it's fine.
 
Lots of USB capable devices (guitar processors, drum modules, keyboards, etc) include an audio interface for convenience.
Keep in mind these companies don't specialize in audio interface design and drivers.
It's not realistic to expect performance equal to the best audio interfaces currently available.

If RTL matters to your workflow, opt for a quality dedicated audio interface.
 
A lot of older line 6 pedalboatds, for sure the XT Live, still had drivers on par with Line 6's older interfaces like the toneport GX...not RME by any means but far better than what's on the Helix now
 
I use a lot of EQ to shape my core tone or match recorded tone, it would be very handy to have a parametric EQ with more bands.
I don't mind if it takes x2 or even x4 times the DSP of the current parametric block which is only 3%.
6 bands without frequency range limitation would be awesome.

Here's a great idea, add up to 10 parametric bands according to my needs in the same parametric eq block.
That would require developing "Dynamic Block DSP" while the block is actively in the routing chain.
 
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