Setting the input gain on audio interfaces

Gear
#12

I’m not sure the instrument level inputs on the focusrite interfaces are particularly well thought out or implemented - if you can take care of your first stage of clipping before you get to the interface (i.e., stick a pedal infront) I think you might get better results. Otherwise a fet-based di like a countryman 85 will also be good.

I think problems arise because the dynamic swing between your most powerful output (usually a muted E powerchord but maybe a pinch harmonic beats that), your sort of middle volume regular playing and your quietest playing is so great that there’s just no way to set up the interface and the amp sim that you get good results across the whole range. Better to compress your signal first in some way whether that’s a clean boost, light drive, compressor whatever.

Even even better in my view is something like the AMT legends series preamp pedals - run the guitar into that, let your main tone generation be done in the analog world and then run your cabinet and room IRs in the DAW along with whatever post effects you fancy.

If you do want to stay purely in the box you could try ensuring that even the highest level out of the guitar doesn’t clip the interface then use a gain plugin (like Utility in Ableton) to make up the gain before reaching the amp sim.

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#13

TLDR;avoid A-D clipping and D-A clipping at the master. Look if there is a software mixer between analog input and the amp sim.

Is there some sort of software mixer “in-the-box” before the guitar sim software, that is lowering the gain your amp sim sees?

The reason I ask, is because that is how it is with my RME Babyface pro. So my setup is, never clip the analog input, if needed I can boost the input in the software mixer or in my plugin, S-gear. Inside of s-gear I can turn up the input gain all the way up, if I want to, and drive the “front end” of the amp as hard as needed. I do take care not to clip the output, if I am using S-gear as a stand alone app.

Inside a DAW clipping it is sort of the same, avoid analog clipping and clipping the master output, before D-A.

Bonus question : does it sound good if you are not clipping the analog in and simply raise the gain on your “amp” inside the sim? Or is it horrible? No reason to fight a problem just because the manual said so, if it sounds good.

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#14

Your number one goal is just to get the signal into your computer without clipping. Do some loud strumming, or loud rhythm playing that represents about as loud as you might realistically play. Set your input gain so that this loud stuff doesn’t clip and those loud peaks end up somewhere between -20 and -10 in your software. You’re done.

Longer answer: There is no need to squeeze super high up against zero on modern gear with decent or better preamps and converters. In fact, pro-level digital gear was originally designed so that approximately -18 (give or take) in your software meters was the equivalent of zero on an analog mixing desk. Meaning, your interface would be designed so that when you set the gain in the sweet spot of what the preamps and converters can handle, the digital signal that the device outputs to your software ends up around -18. The idea was that everything above that, between -18 and 0, is basically insurance that equates to the “headroom” you have in analog gear where you can come in a little hot and not clip. The exact numbers will be slightly different depending on the gear you use, but I try to end up somewhere between -20 and -10 just to keep the analog gear running in the range where it was designed to sound best.

Once you’re in software, you now have a clean non-clipped signal. If your amp sim software is expecting something louder, then you can just chuck a compressor on it before the amp sim, with gentle settings to take down some of the highest peaks, so that you can turn the makeup gain (i.e. compressor output) higher and still not clip over 0. However a good amp sim shouldn’t really require this. Amps are (among other things), compressors, and the amp model you load up should take care of all of that without requiring you to slam its input with a signal that is kissing zero on your hardest strums. If it does, I’d do the compressor method and listen to the output of your amp sim to make sure it sounds good.

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#15

Yeah, Troy has the right of it - in the digital world, your #1 priority is not clipping on input. You have 100% transparent noise-free gain staging available to you “in the box” once you’ve turned your analog signal into digital, so bring that in as cleanly as you possibly can. If necessary if the DAW is still showing you have ample headroom after the .wav file is in a track in your DAW, you can adjust the volume of the raw track there either through normalizing or some sort of variable make-up gain - Reaper lets you do this by right clicking and selecting some sort of properties menu which has a number of gain options. The Overloud plugin might also have some sort of trim control you can use to boost the signal as it’s coming into the plugin.

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#16

Thanks everyone, this has been instructive - and saved me money :slight_smile:

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#17

On a real amp actual voltages matter (to enter nonlinear operating regions), but on digital systems there is often little bearing: For example, my wireless pads my guitar by -12dB (likely because I have active pickups), then cranks it by +24dB, and then I have to drop it down at the input of my Axe FX to “20%,” etc., so how does the software “know” if I’m playing loudly [non-linear], or not [linear]? I believe that what they do is normalize compared to a reference that is specific to one’s guitar (somehow set by heuristics with the “yellow light”) and then reason relative to that (so they’ll likely multiply by the reciprocal of the reference). I wish they were more transparent about the process, but I think that they don’t want to scare customers.

I’m not even concerned with clipping as long as it’s rare: I love distortion and likely won’t tell the difference anyway. I think the real reason not to clip is for purposes of re-amping, e.g., one decides to turn of the (fake) “Mesa Boogie Dual Rectifier” and go “acoustic.”

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#18

This is an old thread but I don’t think the original question was really addressed. The TH-U amp sim has a feature to test the input signal to the sim. If the signal is too low that doesn’t allow full use of dynamic range. The input test asks to set analog input volumes to max before clipping and to strum guitar hard. The input level should reach a certain digital level indicated by the level indicator reach the green region. There is apparently an issue with this test in that, as the original post described, the input digital signal is registered as very weak - never reaches the green. Note this also occurs with non-plugin version of TH-U stand-alone so its not a DAW issue.

I have a help request submitted - not expecting much.

#19

Brother @tommo

From my experience the input levels should be set according to what your DAW sees.
Keep it between -24 and -12, don’t ask me why I know this.

What the interface sees, is unimportant.

#20

You’re an s-gear dude like me. I’m not sure if they are a different scenario than other plugins, but their docs have a lot to say about input level and caution that anything even near clipping can cause undesired behavior. Not sure what that equates to in your your magic range of -24 and -12 :slight_smile:

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#21

They probably correlate, but, well, if you want to be clipping at the preamp level, you should be damned sure you know what it’s doing to the signal. There are a lot of high end mic preamps that people intentionally push a little for the slight saturation you get out of them. They usually cost more per channel than even a fairly high end 8-input interface today will set you back, and there’s a good reason for that.

I don’t think there are too many situations where you have to worry about clipping at the preamp level but not at the converter level, in the “normal” home recording range. But I’d also generally make sure that, if you have any sort of preamp clip indicator, it’s staying out of the red, and very likely out of the yellow if you have that level of visibility, unless you’re absolutely sure that intentionally overdriving it a little is adding something very musical you want in the recording. And you definitely DON’T want to clip at the analog-to-digital conversion point, even if something else is later reducing the signal down to -12-24 when it hits the DAW channel.

There’s a lot more to gain staging than “don’t accidentally overdrive something along the way even if your finished signal isn’t clipping” but that’s the single highest-yield step in the process, making sure nothing is clipping earlier in the chain than the track VU meter you see at the end. :+1:

#22

I’ve had good results when the input your DAW sees is -24 to -12, can allow peeks to breach that but best to keep it in the middle of that range for the most part.

Adjust your hardware gain to meet that range. @Drew they do correlate, but I’m I’m sure there is some hardware or the other out there that does not, in my case it’s been RME always and it’s been reliable. I don’t need to look at the hardware monitor levels.

It’s a headroom game, and for my workflow this is the truest signal setup for my needs.

Sure there are some things you might want to overdrive at the source, it’s a flavour thing and that’s cool too if you know what your going for.

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#23

This pretty much means that nothing in your chain is likely adding and then reducing gain before you hit the DAW, then, which IMO means this is pretty safe. But, I guess the main reason I’m just warranting a little caution is if you’re tracking based on DAW volume, and you have anything active on the channel strip while tracking, that can get into trouble - if you’re monitoring output on the channel bt the channel has a few VSTs running and you’ve got like a tube screamer model and then an amp model, then the interplay of their outputs and gain stages means you absolutely could be be inducing clipping, including at the interface, that you’re just not seeing or able to hear clearly downstream. In that scenario I’d want to be monitoring the DI signal to make sure THAT’s not clipping. Otherwise, even if the modeled output is falling at -12db or so, you could be lopping off the peaks of your signal in ways that may not be overly obvious through a Marshall amp model or something, but will be robbing it of some of it’s impact and “punch” and will cause it to get a bit indistinct in the mix.

This probably isn’t a problem if you’re micing an amp, and monitoring the mic’d signal as it hits the DAW with no FX processing on the tack and the track fader at unity. And if I remember correctly that’s how you record, micing up a real amp. But if someone IS slamming the front of their interface, I wouldn’t want to have them come away from this thread thinking “cool, cool, I just gotta slide down the track slider until it falls to -12db and I’ll be golden here,” and expect that to really work.

#24

If you’re slamming the front of your interface, the DAW will reflect that as well. Though you make a fair point, for example if I was using a UAD and post processing between the interface input and DAW input, it’s not 1 to 1. But we are digressing from my point. In the DAW -24 to -12 is where I keep the signal regardless.

#25

As long as the DAW is monitoring input, not output, and there’s nothing else in the signal chain, that’s true. But it definitely doesn’t hurt to double check since clipping on input isn’t something you can fix downstream.

#26

sorry if this has been mentioned but I would always recommend a dedicated DI if you can afford it. Many interfaces (even with instrument inputs) are not that great and will clip with higher output pickups.

The countryman type 85 is a classic for a reason.

Other than that, aim for about -6db for your peaks (loud strum etc.) and you should be golden :slight_smile:

#27

Yeah, one of my guitars has Black Winters and will clip if I look at it funny with the input gain minimized. I’m probably just going to replace them, but I’ve been wondering what was wrong.

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#28

Yeah, just a weird thing with many interfaces (it even happens on my relatively expensive audient unit) so I always just use a D.I.

nothing wrong with your pickups :slight_smile:

#29

Little update that I have been using recently:

Provided the input DI is not clipping, you can always fix things after the fact by “normalizing” the DI waveform before it gets re-amped by the various plugins. Different DAWs will probably have different options and methods to do it, but in Reaper I found that normalizing by setting an overall target LUFS value (a measure of perceived loudness) gives fairly consistent results.

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#30

Yeah - one of the ways digital recording is transformative that’s maybe a little underappreciated is that in STARK contrast to analog, where every gain stage also adds self-noise, you have access to transparent, silent, noise-free gain control. That plus the much higher useable headroom (off memory the tape noise floor is something like -50 to -60db, while 24-bit .wav files have a dynamic range all the way down to like -144db) means that you really don’t have any reason at all to try to record “hot.” You can peak at -30db in digital, which in the analog world would leave you dynamic range of about -20db in practice, and STILL have more than 110db of usable dynamic range from your peaks to the floor. That’s insane and in practice your limiting factor isn’t going to be equipment self-noise so much as whatever ambient noise is also captured with your recording signal.

It’s honestly pretty amazing. And, arguably, a lot of the “analog warmth” that people miss in digital is probably related to the amount of compression you just had to use on everything to get everything smoothly into the dyamic range that analog could support, plus things like maybe rolling off the highs and lows a little more than CD allowed because of the difficulty in geting a vinyl record to reporoduce that smoothly without the needle jumping out on bigger bass notes. But, the evolution of recording technology in the last 20 years even that I’ve been doing it, is mind-blowing.

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#31

Absolutely, in my mind one merely needs to capture the signal appropriately (no clipping and minimal background noise), and then everything can be repeatedly changed to one’s heart’s content.