Optimal input level for highest accuracy when using NDSP plugins?

I’ve been posting online a bit recently about this subject, and it would be amazing to have some concrete numbers on what internal reference level the Neural DSP plugins are designed to work at.

I believe that the Quad Cortex DI input is 9.5dBu - if someone was to use this as a DI input for the Neural plugins, and with the plugin input at 0, is this the most 1:1 level for using them?

I’m not looking for responses like “adjust until it sounds good” or guessing - there will be a fixed value used to model/capture the amplifiers that gives the most accurate response (as in the same as plugging directly into the amp).

I believe it to be somewhere in the 9.5dBu-12dBu range but anything more concrete would be greatly appreciated!

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Just thought I’d update this.

The very helpful and friendly support guys gave a perfect answer that makes sense for me to share (in case anyone else wants to find this information).

TLDR is, if you use a UAD or (most) Focusrite interfaces, set your preamp gain to 0, and plugin input level at unity/0 and you have the perfect gain response (the same as plugging your guitar straight into the amp). If you use something else, check your specs sheet (it’ll give your value with gain at 0). If it’s a number bigger than 12.2, you’ll need to boost by the difference, if the number is smaller then you’ll need to subtract by the difference.

Our plugins are made with the following audio interface gain:

  • Analog domain sine waveform 1 Vp = 0.707 VRMS = -0.79 dBu equals -13 dBFS in the digital domain.
  • In other words, when feeding the audio interface input a sine 1 Vp from a signal generator, it results in -13 dBFS in the digital domain.

If you want to calibrate your interface in order to mimic the input gain our engineers use when creating and testing the plugins, I would advise you to feed a sine waveform 1 Vp = 0.707 VRMS = -0.79 dBu to the interface and set the interface gain to such level that the DAW peak meter shows -13 dBFS. Feeding a sine waveform on different interfaces will result in different values (again, this is the reason why we cannot provide a concrete value). Check these examples of feeding a sine waveform 1 Vp:

  • UAD Apollo x6: -13 dBFS
  • UAD Apollo Twin: -12.9 dBFS
  • Quad Cortex: -15.1 dBFS (1M impedance) Input level at 0 on QC.
  • Focusrite Scarlett: -13.1 dBFS
  • Focusrite Clarett: -14.6 dBFS
  • Apogee Duet: -14.5 dBFS

However, I have to tell you that can be achieved by connecting your guitar to the Hi-Z input of a UAD interface with the gain at minimum (to ease the pain of doing that with all your interfaces and electric guitar combinations). If your interface features a Hi-Z input, leaving the gain input by default (minimum) is more than enough. Add input gain if one of your guitars lacks output level (as our support team suggested, increase it as much as you can without clipping).

What if you use Quad Cortex as an audio interface?

If you use Quad Cortex as an audio interface and you want to get its input close to a UAD interface (-13dBFS), just connect your instrument with the QC’s instrument input at 0.0 dB. If you wanna match them exactly, you have to boost Quad Cortex USB output by approx +2.3 dB before reaching the plugin’s input.



I read your answer, your considerations and I also saw your video on Youtube where you explain the problem.

I find the topic very interesting, so since my sound card is a motu m2 I went to check my max Input level:
Maximum Input Level Mic Inputs: +10 dBu (Min Gain)
Line/Hi-Z Inputs: +16 dBu (Min Gain)

So being the level you advise me to set the gain knob of my sound card to zero and set to +3.8

Now my current setup for using the quad cortex is as follows:

The configuration I am using allows me to enter the quad cortex with the instrument from input 1. To be able to insert a series of effects exit with the USB 3 of the quad cortex and enter the input of a plugin from the plugin I exit and enter the cortex again with the USB 5/6 at this point I exit the Cortex again and enter a Motu M2 sound card

The strange thing is that I also asked support directly and the answer they gave me was completely different from the one you report and which I paste for clarity:

First email:

There’s no specific target for this parameter, but a good practice is to play as hard as possible and increase the input gain until the signal clips. Once it’s clipping, dial it back a bit so it no longer clips. This will give you the best signal-to-noise ratio.
Alternatively, you can just leave it at 0.0dB, which should be good for almost everyone unless your DI signal is too loud.

Second Mail

Indeed, the gain level of both inputs will mainly depend on the signal level, going out of the QC to your interface’s input.
In both cases, the same rule applies: to get the best signal/noise ratio, always make sure to keep the signal in the green range, so before any clipping starts to happen; there’s no other predefined rule.

What do you think? can you give me your point of view?

The answer they gave you there is essentially what customer support is trained to say, which is unfortunately quite vague advice (but potentially leads to less confusion at the cost of less accuracy). It took me a lot of work to get a direct answer from the developers with more specific information. Their advice of strumming hard and avoiding clipping would lead to single coil guitars being far too overgained, and you’d lose the nuance of different pickups between different guitars.

If you use your QC as a USB interface for NDSP plugins, the most accurate input level in the plugins will be +2.8dB.

If you use your MOTU M2 as an interface (with gain at 0), your value will be 3.8.

If you are doing more complex routing combining both bits of gear, I would recommend sending a sine wave through each chain and comparing the levels you get. Use M2+3.8dB or QC+2.8dB as your target.

Hi, thank you again for your reply. What you say makes sense, but I would like to be able to resolve my configuration with your help if you are available.

So I’ll try to be more clear.

First step quad cortex input instrument: here all input values ​​must be left at zero.

Second step from the quad cortex to the input of the neural dsp plugin here the output values ​​of the cortex USB must be left with the factory settings while the input on the plugin must be adjusted to 2.8.

Third step from the plugin to the quad cortex, in this case the plugin output is different for each scene so I imagine it shouldn’t be touched, while what values ​​should the inputs on USB 5/6 of the quad cortex have?

Fourth step, from the quad cortex to the motu m2 card, also in this case the gain of the card must be at 0 but the output of the quad cortex?

I hope I have been clearer and have your support in this regard

I don’t own a Quad Cortex so I can’t verify levels. You’d probably want to make sure that things are level when going in and out of the quad cortex. But also, what exactly are you trying to achieve?

If you want an accurate gain response, you might just find it easier using just plugins or just the quad cortex, or at most one round of conversion. Latency will probably be more annoying to have to deal with, and combining interfaces generally isn’t ideal for stability.

It’s certainly possible to do what you’re suggesting and have it perfectly calibrated level wise, but it seems like more hassle than it’s worth (unless i’m missing something?).

Maybe this video can you to understand better:

I follow the Tom suggestion, but i add 1 step before for insert the effect like pitch shifter or compressor before the plugin.


Apologies that this is a slightly older thread, but trying to get hold of the person in the know.

I have been following this topic closely, and have a 3rd gen Scarlett solo which should mean interface gain at 0, and 0.3 on the input - however I end up with just a really thin tone, and in some cases my neural plugin is showing hardly anything on the input meter (sometimes it shows nothing at all in the case of some legato) - I’m running BKP polymaths so not super low output pickups.

I therefore wanted to try and feed my interface a sinewave to see what a DAW is registering. If I generate a 400hz sinewave in audacity at 1 amplitude (which I understand is hotter than what Neural use), and feed the output from my interface back to the input, I was expecting to see -13 on my DAW (or a bit hotter in this case), but I don’t, it’s much less (-35 on my DAW). Have I made a total dog’s dinner of this, and is there a way I can test to make sure my interface is behaving as expected as it just doesn’t feel right. Any help is much appreciated


“If I generate a 400hz sinewave in audacity at 1 amplitude (which I understand is hotter than what Neural use), and feed the output from my interface back to the input, I was expecting to see -13 on my DAW (or a bit hotter in this case), but I don’t, it’s much less (-35 on my DAW)”

Are you measuring the analog voltage of this sine wave signal? To verify your interfaces headroom, you need to know the analog voltage of the signal you are testing it with.

Regarding the Solo 3rd gen. The specs state:

Instrument inputs

Frequency response20Hz - 20kHz ± 0.1dB
Dynamic range110dB (A-weighted)
Maximum input level12.5dBu (at minimum gain)
Gain range56dB

So, at minimum gain, you have 12.5dBu of headroom. NDSP is calibrated for 12.2dBu of headroom, so in theory boosting by 0.3dB will give the optimal level. Are you sure your input is set to instrument level and not line level? Line level will yield a signal that is too low, and itll also make the pickups sound weak as the impedance is very low for pickups.

Also check to make sure the interface is getting the correct power and try different USB ports (shouldn’t be an issue but just to rule things out).

I’m not sure how the focusrite software mixer behaves, but also check to make sure the signal isnt being altered by that in any way.

Thanks for your response. I’m not - is this as simple as chucking a multi-meter on the end of the cable sending the signal? 100% have the interface set with inst on - if I turn off the button, the signal drops even further

I’ve tried different USB ports, different cables to power the interface, and also different cables linking the output to input

Hi. Just FYI I found your video in an Axe FX thread on how to measure this - I’ll give this a go and report back!

OK so we have an update. It turns out, with my Scarlett solo, it doesn’t appear that the minimum gain dial is actually translating to -12.5 as it should. I produced a sign wave on my laptop, fed it out my interface, chucked a multi-meter on the end to get the voltage of the signal, noted it, and then fed that cable into my DAW where it gave me a dbfs reading. I found your calculator that if you feed in this information, it spits out your interfaces actual reading. My focusrite is spitting out a -14.45 on the lowest gain setting, so I slowly dialled up the gain on the interface to get to the magic -12.2 and bam, that’s everything feeling much better and where it should be. If you are interested, I was feeding a 0.286v sine wave, and in my daw it was producing -23.1 reading

Hi everyone - I am very much confused by this topic. I mostly compose orchestral music in the box. However, after a long hiatus from playing guitar, I decided to get back into play and would like to record. So, I bought a few NDSP plugins (Plini, Tim Henson).

My setup is:

  1. Guitar (passive humbuckers)

  2. Grace M101 preamp (with Hi-Z input) – the minimum gain for the Hi-Z input is -10dB (fully counterclockwise and increases in 5dB increments up to +45dB) – additionally, according to the manual, the Maximum output Level of the Grace M101 is +25dBu

  3. Lynx Hilo mk1 – the following specs relate to the Hilo Line In:

  4. Mac Studio Logic Pro with NDSP plugins

My question is, being new to recording guitar, how would I set the Input within the NDSP plugin to correspond with the 12.2 number set forth in this thread?

I hope someone can help me as I am at my wit’s end

Thank you!

These values are the available gain from the preamp circuit, it’s not the spec we are interested in here. Unfortunately it doesn’t look like the Grace manual provides the specs you need. We need to know the available headroom which involves the headroom of the converter and its relationship between analog signal level and digital dBFS level. This spec is saying “the preamp can add up to 45dB of gain”.

In this case, I’d recommend running a 0.775V (or any other known value) sine wave into your instrument input on the Grace (and Hilo if you like), and noting down the dBFS level that the sine wave shows as. Then you can determine your input headroom and adjust accordingly.

This video should help:

Thank you for the response - I do not have a DI box, Reamp box, or a multimeter though

In order to set the proper levels, I will have to either get the above equipment or another interface with known values

Perhaps I can email Grace Designs and ask for:

The Grace M101 is for all intents and purposes a DI box in this instance, so unless they know the converter they are using, they can’t tell you this value.

You know that the Lynx Hilo has 22dBu of headroom on the line inputs, so it would just depend how the Grace is set. It could be that with the preamp gain at minimum (-10dB), and the trim at maximum (+10dB), then you’d have 22dBu of headroom on the Lynx. This would be a bit too much headroom IMO, as you’d need to boost another 10dB to hit 12.2dBu. Ideally you’d have the gain control set to +10dB and the trim at 0, and then you SHOULD have 12dBu of headroom on the Lynx remaining.

But you’d need a reamp box and/or a multimeter to verify your levels.

You can buy a fairly cheap multimeter off Amazon to calibrate, they’re quite handy to have around the studio. Same goes for a reamp box - something like a Lehle P-Split can be used for a multitude of things (including as a line isolator) which can be handy for a lot of things.

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Thank you again

The Lynx has 24dBu of headroom (that’s the highest setting)

So, if I set the Grace to +10dB and trim at 0 and then the Lynx at +22dBu line input trim, then, theoretically, would I only need to boost the NDSP plugin Input by 0.2? or would I decrease it by 0.2? Damn this out of the box stuff is confusing

Hmmm, maybe I should just get a Quad Cortex to plug directly into my computer and record my guitars directly through that, bypassing the Grace and Lynx altogether? Can I record a guitar DI track via the QC or will it only record with amp/cab/FX?

You could use a QC as an interface to record guitars, I believe the DI level is 15dBu over USB. Or something like a Helix or Fractal has SPDIF which can plug directly into your main converters and would work well (less hassle changing monitors and plugging things in).

Honestly, I’d try and make things work with your Grace and Lynx as they’re both really nice high end gear. You’d just need a way to run a known voltage sine wave so you can measure the headroom of both pieces of gear in circuit.

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Once again, thank you @MirrorProfiles

I set the Grace at +10 and the Lynx Line Input at +22 and got a much better sound than I was initially getting when I started trying to piece this together

On another note, since I am new to trying to record guitars into my DAW, should I try to have the fader level in my DAW (Logic Pro) stay below a certain level (I keep reading (and probably misunderstanding) that when recording guitar you should try to record at -18)


This is something else that gets thrown around online without people really understanding why.

As far as recording the DI’s, as long as you leave your Grace and Lynx how they are you’re good. Once the signal has gone through the amp sim, if you are running into other analog modelling plugins, you may want to make sure you aren’t running them too loud or quiet.

Generally its not as critical as it is for a guitar amp as most studio processors are designed for a wide range of levels.

The -18dBFS thing is also not entirely straightforward. -18dBFS refers to a sine wave when calibrating. Its basically saying when a sine wave is -18dBFS, the meters will say its equivalent to an analog signal level being a certain loudness in the plugin. It doesn’t really have anything to do with your actual signal, its relative to a reference level.

Again I have some videos on this that might explain it better: