Calibrating Input Level for Plugins

a very crude video, so apologies. on some models I adjust input levels to where it gets more similar. Of course this isnt taking pot tolerances or bright caps into account, but they shouldn't make a 12dB (in some cases!) difference.



Dropping the DI 6dB (as we worked out yesterday) for the FM-3 seemed very close to the real amp
 
Just checked my calibration again, the same sine wave signal triggers the gate to the nearest decimal in both HX stomp and Native so as far as I can tell I'm (hopefully) not doing anything stupid.

I'm actually finding the 0.707vRMS = 1vPeak = -12dBFS calibration a bit hot for how I play. The Avalon U5 I use has 3dB increments and -12dB is the 4th one. Thinking I'll go to -18 or -21 and boost internally (or not at all for the 2204).

Something I'm finding odd is there is still a 12dB difference between Helix and Amplitube still. Even in the video above, its nuts how the same signal needs such a variety of different input levels for each one (some want a lower signal, some much higher).
 
Thinking I'll go to -18 or -21 and boost internally

Ill Allow It GIF



I see you almost hitting 0dBFS when playing with -12dBFS calibration, what pickups?
 
I see you almost hitting 0dBFS when playing with -12dBFS calibration, what pickups?

Yeah, again much hotter than I'd have probably set if I was going off Line 6's visual guide. Seymour Duncan Custom. Same sort of volume as a JB (I've had EMG's and a Custom 5 in here and they're all fairly similar in terms of output)
 
Input 1 (unbalanced 500mV):

View attachment 5138

Input 2 (same unbalanced 500mV):

View attachment 5139

Input 2 (balanced 500mV signal):

View attachment 5140

I'm confused here, because it looks like I have them matched the same when using the line input with balanced cables. If I were to reamp like this, I'm pretty sure it would sound like I'm slamming the input way too hard. My default was to set the Input trim to around 0.5 and then thing would behave more like my amps.
Just seeing now this discussion about the FM3, I hope I can clear up some things with some known facts and what I've observed on my fm9.

1. Inputs have an automatic digital gain compensation, so that whatever input gain setting you have (what's called input pad on the fm3 iirc), the signal reaching the grid is always at the right level for the amp/fx.

2. Out 2 is designed for unity gain, so as you probably have already found out, that means that sending 500 mV to input 1 you should get 500 mV on out 2 as well, with the out 2 knob set all the way up.
Out 1 on the other hand boosts the signal by ~18.1dB to reach line levels.

3. Vu meters on the outputs don't display dBFS, so if you're above the 0 you're not actually clipping the output, 0 equals to -12 or -15 dBFS (can't recall exactly atm)
EDIT: yep, 0 is -12 dBFS, but the analog outs start clipping around -6 dBFS if you have the Out1 knob all the way up, on the fm9 at least

4. Shunts in the grid always represent 2 channels of a stereo signal, even when they come from input 1 which is mono. The signal is simply copied at the same level on both channels.

5. A mono block (like drive or amp) by default has the input select set to "sum L+R", with that the two stereo channels of the incoming signal are summed together but lowered by 6dB to keep a consistent level entering the block, so that it's the same as taking just one channel (left only or right only).


To go back to the purpose of this thread, finding out what level fractal devices expect at their instrument inputs (IN1) should be quite easy: send those same 500 mV at input 1 (without clipping the AD converter) and just check at what dBFS level the instrument input (should be usb input 3 on the fm3 iirc) is detected in your DAW.
 
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PS: when reamping thru a fractal device, always make sure to check the pan law in your daw cuz it might attenuate the signal going back to it.
And always send it as a stereo signal to the USB IN or IN1 block (unless you change the input select parameter on the first mono block)
 
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PS: when reamping thru a fractal device, always make sure to check the pan law in your daw cuz it might attenuate the signal going back to it.
And always send it as a stereo signal to the USB IN or IN1 block (unless you change the input select parameter on the first mono block)

The points you made above were definitely all what I encountered yesterday, so that all checks out.

I’m not using the FM-3 as an interface, and AFAIK, In2 is better for balanced line inputs and In1 for Unbalanced Guitar?

I’m not sure about the FM-9 but In1 on the FM-3 is mono, In2 has L and R. There are various options in the Setup page for whether you want to Sum L and R or only use one input etc.

Why send a stereo signal in over a mono signal?
 
I’m not using the FM-3 as an interface, and AFAIK, In2 is better for balanced line inputs and In1 for Unbalanced Guitar?
Never checked that, will do asap

I’m not using the FM-3 as an interface
You could use it just to take this measurement (I'll do it as weel) or, in alternative, you can do it via spdif too by setting spdif out source to "input 1".

and AFAIK, In2 is better for balanced line inputs and In1 for Unbalanced Guitar?
Seems so from your measurements, and it is normal that a balanced connection is 6 dB louder than unbalanced. You might just boost the signal by 6dB if you want to use IN2 with an unbalanced signal.

I’m not sure about the FM-9 but In1 on the FM-3 is mono, In2 has L and R. There are various options in the Setup page for whether you want to Sum L and R or only use one input etc.
Yes, it's the same on the fm9 and axe fx III as well, input 1 is mono on all of those, but beware that they're mono only on the analog side, once the signal becomes digital it also becomes stereo.
All those settings on the other inputs don't change the level though, cuz that's managed on the blocks.
Sum L+R is available only on the outputs and works exactly as the same setting found on the blocks (sum plus 6dB attenuation)

Why send a stereo signal in over a mono signal?
Cuz that's how the grid works in Fractal land (and the signal routing in most daws too).
All digital inputs in the fm3 (USB IN, In1 set to digital and spdif IN) are stereo, so if you send the signal only to, for example, usb out 3 you'll get the signal only on the left channel on those inputs, and when that signal reaches the amp block, by default it will sum L+R and reduce the gain by 6dB, but since the R channel has no signal all you get is a 6 dB attenuation. So the only options to get it right are sending a stereo signal (thru usb outs 3 and 4) so that the amp will sum and attenuate two identical channels and will get the right level, or setting input select to "left only" on the amp block (or, to be precise, in the first mono block on the grid).
 


I think Jason may have had an eye on this forum. A lot of what he says is correct, but there's definitely stuff I really don't agree with - most of all disregarding IK's suggested calibration, and capturing the amps with test signals much louder than what they are being played with.

The main problems for me being:

- Most people will calibrate ToneX for the loud DI input level that their meters suggest. If you disregard this, your captures are going to have a different calibration to the vast majority of ToneX users. I'd imagine their pedal is following a similar calibration (or should be). Perhaps this is why Jason is having to set his ToneX pedal to -15dB (I'm pretty sure if I did something similar the ToneX calibration would be 12dB higher than the DI signal)?

- The test signal's have specific dynamic qualities that are there to give an accurate representation of the tone. If you're capturing at a level that is much louder, the amp may not exhibit the dynamic properties at the gain levels its intended to be used at.

- I think there should have been more of a focus on interface's 0dBFS reference levels for maximum dBU (in and out). Then also demonstrate how to set this so the signals in and out of the computer are accurate for any interface, DI box or mic preamp.

- he doesn't really mention that different software needs different input levels. He's found levels that work for him but other users may not have the same luck if they are using other gear.

- he should have mentioned and included a link to this thread and @James Freeman 's hard work.

- youtube videos that are educational in format and that offer lots of good advice with some bad advice can end up causing more problems. It could all be avoided if manufacturers explained this properly. they all have R&D teams and social media teams. no excuses in 2022 - help us use our gear properly.

- if I use my HX Stomp as a USB interface to record a DI signal through its instrument input, my (passive pickup) signal clips when I pick really hard. If I follow Line 6's input level guide, then I'm losing the correct behaviour of the modelling. If I use a Line 6 device as my DI, I won't have 12.5dB of headroom, I will have 0 room. Therefore, its better for me to go in at a lower level, and boost back up accordingly afterwards.
 
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The randomness of all this is mind numbing, now we have Output Level involved for capturing too.
So not only the capture itself has a random baked in gain but also the user sets random input gain afterwards.

Because vintage single coils are just as hot as a 18v EMGs and a JCM800 has more gain than a 5150III.

Play hard, don't clip!
 
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But the question is: Does is really matter?
If people accept they don't know what at what levels their gear work and even when specs are published they don't read it?
Couple of year ago I was really curious what Helix levels were. I had to measure them.
https://line6.com/support/topic/48167-helix-input-and-output-levels-research-almost-solved/
So the instrument level is 11dBu max (11dBu=0dBFS)
To get 11dBu with an audio interface instrument input you just have to know what your max line output is, make a loop back connection, set signal generator level to negative difference (max out level minus 11dBu) and set your input gain to get 0dBFS on your DAW meter.
Still, if your level is 6dB louder or 6dB quiter, can't it be compensated with an amp gain knob?
Where is the evidence it sounds completly different?
I am big fan of +11dBu max instrument level "recommendation" but am also interested at what line level different modellers/profiler manufacturers propose for their models/captures.
Please keep sharing your findings.
 
I see you almost hitting 0dBFS when playing with -12dBFS calibration, what pickups?

figured I'd try a few other guitars just to see where they stand. Used HX Stomp as an interface (which I'd never normally do) but rules out anything dumb on my end. Played as hard as possible to gauge what it can take....

498T in a Les Paul clips the input like a motherfucker. Loudest peak from the Telecaster was -4.8 (generally the peaks were between -7 and -8, so plenty of headroom there. My Eastman with 57 Classics peaked at -1.2dB

Screenshot 2023-03-21 at 19.33.35.png


What's nice is the Avalon has stepped gain controls that measure exactly 3dB, so if I wanted I could easily adapt to any guitar (or just keep it set at minimum and leave Helix Native/ToneX boosted by 9dB
 
I do but the wiring is buggered. Pretty sure the internal preamp clips pretty early (unless it’s modded for 18V or really far from the strings), can’t remember what that would produce going into a DI though
 
Okay tested my EMG 81.
Strumming hard the internal EMG81 Opamp is clipping but not Helix Input, without the pad I max at -5.9dBFS.

Helix Guitar Input is 11.5 dBu = 4.117v Peak. dBu Calculator
4.117*10^(-5.9/20) = 2.08v Peak
20*log(2.08/4.117) = -5.9dB

Battery is fresh enough.
Apparently the classic EMG 81 is just not a very hot pickup.

EMG81 no pad.png
 
I suppose the distance from the strings would only affect the RMS value, once the preamp clips internally it ain’t going any higher.

I think a lot of the perceived “hotness” of an EMG is from the preamp clipping the loud notes, and the softer ones being closer in volume for a more even sound. I used to play EMG’s all the time, but rarely play them now. Whenever I do, I always enjoy (the novelty of) how even everything sounds.

It was also kind of the reason I switched to passives, I wanted to have more control over my dynamics and EMG’s clipping puts a stop to that. I like them for a particular sound though, nothing does THAT THING better
 
EMG 81 vs SD SH-4.png


It was also kind of the reason I switched to passives, I wanted to have more control over my dynamics and EMG’s clipping puts a stop to that. I like them for a particular sound though, nothing does THAT THING better
100%
I love my EMG's 81/85 they are in my main super strat guitar (Cort X-TH) which I use daily, but I prefer how the Seymour JB/59 sound and feel better but they are in my Edwards LP which is not as comfortable so I use it less.
 
EMG’s waveforms look so hilarious in a thread about headroom/gainstaging….

Sort of like a 5150’s overly cold bias, when you “fix” it you lose its magic. and it’s important to preserve how that signal hits the amp, clipped or not, it’s just as important
 
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