Calibrating Input Level for Plugins

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Anybody up for trying to crack the IK input levels once and for all?

I spent some time today with SJ50 and compared to my 5150 Block Letter, as well as a few other emulations that I know the internal operating headroom of. I found on the real amps, having the red channel's gain at 6 is where it noticeably flubs out and gets overly squishy and pretty useless.

Boosting the AT5 input to around 1V=0dBFS seemed to get this behaviour (sometimes it feels like it may be lower than this though). I could try and use my Mark III to determine gain levels, and some Marshalls. Maybe some others here have some real amps they can reference that are part of AT5? Such as Fender, Orange, VH4, TC100, JVM Satriani JMP-1. The manual barely even mentions input level besides saying "just make it clip and then back off a little". Following their advice, I have my headroom at about 4.6dBu - playing as hard as I can clips the AT input LED but generally its hovering near the top. This at least seems to match the behaviour of the 5150 quite close. Some other models it seems closer to 12dBu, and some closer to 0dBu. Having to guess, and have doubts is a workflow killer that doesn't exist when using real amps.

@IK Multimedia - Pete you'd win a ton of respect from me if you can give us some concrete numbers from the devs? Or maybe there is the opportunity to invite someone to discuss it here?
 
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If you really think about it, it's mind numbing that the programmer that developed the plugin don't know what their reference level is.

A fixed 16/24bit PCM data from the ADC is used as a signed variable, this variable is fed into the virtual 12ax7 model.


For example:

ltspice_wav_3.png



In the code the signed 16bit PCM is equal to a virtual 1v Peak, for us it means 1v Peak = 0dBFS.
Maybe the programmer doesn't understand what dBu is, but he surely must understand how he uses variables in his code.

Anyway, this is getting really boring, "play hard, hit green, don't clip" is major fucking bullshit.
The dBu-to-dBFS info should be a part of every plugin manual.
 
IMG_2495.jpeg

I mean, it’s an advertised feature of V5 which is hardly a new product.

I’d also love to know more information on their poweramp->speaker interactions (knowing what loads are used for what amps would be great). Immensely frustrating to see these important features get pushed as something they value, and then there is no mention in the software or manual on how to take advantage of it.





SO, after the thread about the AFD100 with the awesome prototype and gut shots I thought I'd compare a few emulations.

I'm able to set accurate input levels for Mercuriall and Fractal's emulations.

They all sound like they have different bright cap values, but even still if I boost my 11.4dBu input by 15dB, Amplitube's model still sounds too undergained. How can I have any confidence at all that I'm getting a correct response when the input is fully maxed and it still seems different to other emulations?
 
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Comparing STL's JMP-1 to Amplitube. This model in Amplitube seems to be close at my 11.4dBu settings. What's confusing is that it has more gain than STL's model - it just leads me to think IK are using different calibrations in different models, as some models (like the Slash AFD) want seriously hot levels to sound correct. Even if we don't know the exact internal operating level, it would be nice to know whether all models follow the same calibration or not. It really muddies the waters in trying to dial something in if your constantly doubting whether it's sounding as it should. It stops being an emulation at that point.
 
James Freeman said:
The most important parameter in a guitar amplifier, GAIN, is the most random and ignored one in plugins.
Click to expand...
"USE YOUR EARS BRO" is the biggest cop out too.

"My emulation is perfect, you just aren't using it right" gives a hell of a lot of wiggle room. Leaving out optimal internal headroom levels is instantly just hobby level BS that needs to be left in the past. Other plugins manage it, and often their primary goal isn't to gain up a signal. The entire purpose of an amplifier is to AMPLIFY a signal - therefore the levels going in, and how they're processed are critical. Amp's don't have wide ranging volume controls before the preamp, with peak meters that make you normalise your signal before you start.
 
MirrorProfiles said:
"USE YOUR EARS BRO" is the biggest cop out too.
Click to expand...
This one is really close to LMGTFY to me. I so often see noobs asking about compressor basics and some blowhard pulls the "use your ears bro!" IF they knew what to be listening for they wouldn't be asking you peckerneck! With LMGTFY its usually after they get super frustrated googling down 50 different rabbit holes, then google points them to a forum of alleged experts so they finally get to say "yay! Now I know who to ask". And when they do, they get LMGTFY

Regarding calibration, even very few interface manufacturers spell anything out. And then the piles and piles of mic pre's out there without any real info on how much gain they can actually add.....PITA
 
People with AxeFx3, FM9, FM3, can anyone please give us solid input dBu numbers and noise measurement (RMS) in DAW when used as a USB interface or SPDIF?

Edit:
In the wiki (link) for the Axe-Fx 3 and FM9 I see:
INSTR (front) – 1/4” phone jack, unbalanced, conditioned for guitar use, auto-switching, 1 Megaohm (adjustable), +16dBu instrument level, Secret Sauce IV
Click to expand...
"The instrument input max voltage is about +/- 5.9V (11.8Vpp = about 17.5 dBu). Inputs 2/3 can handle up to around 11V (about 23 dBu)."
Click to expand...
So is it 16 or 17.5 dBu?
No dBu info for FM3?

Noise level reading in RMS in DAW would be great.

EDIT2:
BTW, the wiki is wrong.
11.8Vpp/2 = 5.9Vp which is 14.6 dBu, seems they confused RMS and Peak.

Hence, we need real measurements.
 
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Using a UAD Volt (powered off my pedalboard PSU), an iPad Pro+Logic, Lehle P-Split as a reamp box (just mad really). Set it up for 0.514V RMS
FM-3 as a USB interface on my main recording computer, the level fluctuates very slightly between -21.4dBFS and -21.1dBFS. Not sure if this is a result of the signal generator "rig" or something the FM-3 is doing (the voltage seemed stable when I measured though). Return input, it doesnt fluctuate at all, fixed at -21.3 (Ill use that as a value). That's giving me 17.74dBu for the USB interface.

Screenshot 2023-05-27 at 13.23.42.png


Nothing plugged into the input, getting these levels of noise. -120 Short Term LUFS

Screenshot 2023-05-27 at 13.33.22.png
 
Unless I'm missing something, the only options I can record via SPDIF are:

Screenshot 2023-05-27 at 13.41.29.png

So if I want to record a DI via Input 1 and SPDIF, I have to route the input to Output 1 or 2. If I route Input 1 directly to Output 2 like this:

Screenshot 2023-05-27 at 13.43.51.png

Then my 0.514V RMS signal comes in my DAW fluctuating between -3.4 and -3.1dBFS. If I go into input 2, directly outputting to Out 2 and then via SPDIF, it comes in at -3.2dBFS without fluctuating at all. -0.36dBu (presumably there is some signal boosting going on between input and output).

Lastly, if I route Input 2 as the SPDIF source, I get -26.1dBFS=22.5dBu
 
Thank you.
Very interesting, so FM3 is 17.74 dBu and -114dBFS RMS.
That's -108dBFS RMS at 12dBu... is that like 19dB noisier than a Helix? :unsure:
EDIT: Try to make it 12dBu with the Input Trim in the FM3 and read the RMS noise floor.

PS. It is better to measure noise floor with a cable plugged in and a guitar with volume at zero.
For example, Helix completely mutes (-Inf) the signal when there's no cable in the guitar input.

MirrorProfiles said:
Lastly, if I route Input 2 as the SPDIF source, I get -26.1dBFS=22.5dBu
Click to expand...
Yes, that's probably the correct way to measure the ADC value.
The Outputs are trying to stay in the -10dBV and +4dBu standards and not directly feed the Input to SPDIF.
 
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a bit baffled as to why the USB and SPDIF signals are 5dB different when using input 2 as the source. Surely these should match?
 
James Freeman said:
Wait.... why do you use Return/Input 2?
Only Input 1/ Instrument has the super quiet "Secret Sauce" input.
Click to expand...
I'm typically using an Avalon U5 as my DI, and then routing it afterwards depending on what I'm doing. I find the FM-3's I/O kind of limiting for my workflow so its easier to use the analog in's and out's on it. Secret Sauce sounds like something I want to avoid :ROFLMAO:

These results being so varied confuses me a ton, why are USB and SPDIF giving different results for the same input source?
 
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