9 months on from the initial discussion - there is some advice from a real expert. Important takeaways: -As Ed S has been pioneering and his work was referenced in Valdemar's video - if you are concerned about the ACCURACY of your amp sim plugin (or wondering why your AC30 sim sounds like a Dual Rectifier - there is great reason to calibrate your input -Valdemar is full in step with that (which was the whole thrust of our discussion in any case) -Valdemar's recommended steps - 1) Set your Input to 0db 2) CALIBRATE YOUR PLUGIN ACCORDING TO THE SPECS IN ED'S SPREADSHEET (to ensure accuracy) 3) Add Gain on your interface to below clipping (this may not be necessary for MANY people) 4) Reduce the input gain in the plugin by the amount that you've added on the knob
Yeah, this guy just got a really crappy interface that has loads of noise to create a video saying everyone was wrong to set gain to zero. A video that would have been better titled 'why is Behringer kit so cheap?' Gain on zero works and won't create loads of noise for many interfaces (the pod XT doesn't even have a gain knob and it's not noisy dunno about the newer line 6 boxes that double as audio interfaces whether they've added a gain knob - if not, well they are all on zero too) But (of course) putting interface input gain higher and lowering input gain on the plugin works too. The key thing is what the input gain for each plugin should be relative to the gain on zero for your particular interface that the spreadsheet provided.
@@johnnathancordy Fair enough. I have an alesis io 2 that I might have a play with later (because that does have a gain control) I figured the 'gain on zero, input gain value from this spreadsheet' was about keeping the instructions simple and only having to change the input gain on each plugin. One problem I see with setting the input gain 'just below clipping' is that can vary from guitar to guitar. So you might end up having to configure every time you swap guitars? If that were the case maybe getting your highest output guitar so it doesn't clip would work - even if the signal to noise is slightly higher for some lower output guitars?
@@michael1Yes, that's the misleading part of his video. He actually commented on his own video (in the comments section) "admitting that he cheated a bit" just to make a point. He used the example with crappiest interface and gain knobs on the plugin maxed out, just to prove someone can reduce the noise floor and have and audible proof. For most of decent interfaces in 2024, the reduction in noise floor would be inaudible. So, in my opinion, there's no point changing the setting (because you have to do this process for all the guitars some is using) to get a 1 dB reduction of noise floor.
@@michael1 +24dB increased gain to me sounds like he's plugged into line input, not hi-z. The instrument input should have its own gain circuit before hitting the preamp. that, or he's using a passive DI box. He's not technically wrong about these instructions but under most circumstances with even an entry-level interface, the noise floor is so low you wouldn't be able to hear it anyway. It's also just kind of a fiddly process, though in the end what matters is you liking what you hear coming out of your speakers.
Really awesome response video! 🙂 After all of this discussion, I will absolutely tell you that, sometimes setting your gain to its lowest settings will be "Just Fine"🙂 - The problem that had arisen was that people were taking this advice, which is a great shortcut to getting decent results, and claiming it was the *optimal* method of adjusting things. The factors that make the biggest difference are; 1. if you are using a weak output pickup (or if you roll off the volume, like you showed). 2. if your interface has an unusually high instrument input level (meaning you need a stronger signal to drive it to full peak value). 3. If your interface has a lot of self noise. 4. if you're using very high gain ampsims. In my demo, pretty much all of these elements were involved, and that shows the absolute worst case scenario that can occur. Just as an aside, I know Steve Atkinson, creator of Neural Amp Modeler, has just announced some new features in NAM that will make calibrating input gain much easier, he's got some really clever thoughts on the process that we should be seeing in a future update!
So to sum this up… joh is looking for real amp like response from the modelers and you were looking for highest dynamic range or deepest noise floor..they are two different things.. You video was irrelevant to what john was claiming
@@AgapeSignalNo, not at all! Both are after the "real amp like response", but the method RUclipsrs talked about half a year ago with setting the input to zero achieved that at the expense of a healthy signal to noise ratio. Now he presented a new method that achieved the same thing without sacrificing a clean and healthy signal.
@@AgapeSignalexactly. And even though I commented on his video pointing out this confusion he's creating I don't think he understands it. Nor the rest of people cheering his "new method". Fighting to lower the noise floor on bad interfaces it's one thing, compansating the input gain is a different topic.
@josuastangl7140 you misunderstood the point. Ghost Notes video is exactly about this. Lowering the noise floor. He's not taking in consideration any adjustments for the input level that would make an amp sim to sound and react like the real amp. EdS spreadsheet is made to correct the input gain for different plugins.
@@OrangeMicMusic GhostNotes is working off of EdS spreadsheet, keeping the overall input gain identical. He simply for example adds 6dB on the interface and reduces 6dB in software. This way the input gain is still identical and optimal for the real amp feel, but now you have a lower noise floor. His input gain values before the signal hits the amp sim are still identical to EdS's spreadsheet
It is really weird that a company like Neural DSP makes a 60 min comedy movie with an IMDB rating of 3.5/10 for a product they are about to release which is basically the same product they released before but smaller, but dont bother to clarify this argument with a 3 min video.
My brother is an audio engineer, and his response to everything going on was "expecting the manual to explain how to set your gain is like expecting the manual for your keyboard to teach you how to play piano." I think the video that is being referenced was great, but it's too succinct for a guitarist to understand what they're actually doing and all of the why and how to set everything up when the spreadsheet doesn't cover your software. You really couldn't explain it all without a three hour video, which I would be surprised if someone hasn't made already, but it's probably about gain staging broadly, and you'd have to figure out how to apply that to guitar. When I use a passive DI box though, a lot of my gain problems will go away.
@@JewettMusic The issue is that they all tell you to set the gain bellow peak but that doesn't give you an accurate representation of the amps. If you don't own any of the amps how do you know if it's accurate or not? Rhett Shull's video was great in the sense that he compared an ampsim to the real amp.
@@andresilvasophisma that is the big issue. Gain staging is an expert's skill that requires experience, it can't be boiled down to "do one thing on every situation" and the amp sim manufacturers don't know the output of your pickups, which matters too
@@JewettMusic Thank you for a voice of sanity in this page. Guitar players suddenly seem to think they are audio engineers, and then complain unknowingly about their own incompetence. Also, a good DI box is a useful recommendation for this particular forum, even a cheapish one like a Radial, and it will inevitably clean up a lot of noise, too.
@JasonSadites response is a great addition to this conversation. In addition to a lack of standards, the other issue is a lack of manufacturer documentation combined with differing approaches applied by third parties who sell captures. A consortium to develop some standards would be a very good thing.
There are standards for half of the problem (the calibration between input level and digital full-scale); see my comment on John's pinned comment for the refs. But the standards are really recommendations; there is no enforcement mechanism. So your point about "lack of manufacturer documentation" is the real issue (also addressed in my comment). The better manufacturers do tell you enough to settle this half of the problem. Alas, the datasheets that Ghost Note showed in the video were bad examples in this regard. -Tom
The problem is as old as musical instruments themselves…the engineer designing the tool and writing the instructions doesn’t necessarily know the goal of the artist using the tool/instrument. Really well said with humility and honesty. There’s no reason to apologize for getting an important (to us musicians anyway) discussion started. I always enjoy your video! Keep up the great work!
There are two different end goals. Engineer is a measurable answer Artist 100% subjective answer Seems weird that an artist's end goal would be to hear the plugin sound the same as the real thing. Most established artists are usually looking for a unique sound.
Dear John - a load of the noise from your guitar that I usually hear on your channel is just digital clock noise from your interface (and possibly your footpedals) because you are too close to your desk. Test this - try standing two meters back from your desk and record your guitar that way. You will hear a much more drastic difference in your noise floor, and much less of that "vacuum cleaner" sound.
@JuanDavidArtal Something to do with how the dimming works. Kinda like how a dimmer switch can introduce a bunch of line noise. It should also be noted that power supplies produce a ton of EMI..
@@benjaminashlin Right. I just was pointing out a variable that people can try to see if it makes a difference. Can't image its something that a lot of people would even consider. Like, I have a pair of active monitors that introduce a ton of noise because they create a ground loop. Easy enough fix by just cutting the ground connection on the XLR (since they don't have a ground lift). I can only imagine how many people struggle with that and don't realize what's going on.
I also saw that video the other day and it finally clicked for me. Originally I was like everyone else, wondering why this, for example, Bassman plug in sounded so darn dirty, then I saw your video and switched to leaving the Scarlet interface at 0 and the tones were right and proper and everything sounded so much better. However, the waveforms in my DAW were TINY, so darn small you could hardly see em, I play with a passive Strat and Jazz Bass, btw. So, I tried the new way the other day to get the interface up to near clipping, and it gives me a number for the increase, then I subtracted that number from the plug-in input, and it is finally OPTIMAL. I have the tones the way I want them, and I have big healthy waveforms, and better signal to noise ratio. I think we finally figured it out ya'll. Thanks for leading me on this journey. Especially as a bassist primarily, figuring out guitar tones is so much more difficult than on bass, like they aren't the same thing at all when trying to get great tones. my mind was blown by that when I started learning guitar a couple years ago.
Your point about the waveforms is an important one - not because of the waveforms themselves, of course, but because the signal level that these amp sims are "calibrated" for isn't consistent with the signal level most other plugins are calibrated for. Use the interface the way the ampsim creators tell you and, say, you want to run an LA2A emulation before a clean amp to get a more consistent funky compressed sound? Well, your level will be so low you'll need to push input gain on the LA2A a lot to get any compression at all, and then you'll have to lower output on the LA2A a ton to get it back to the level the amp is expecting. If the compressor adds saturation based on internal gain structure (like the LA2A often does), that can change your sound drastically.
I guess I was trying to communicate based on the idea that we wanted to achieve as close to the amp as possible. Not really focusing on the signal to noise ratio. As I tried to demonstrate in my video comparing the plugin to the amp with the input at zero. He’s completely correct in advising that in order to get the best noise and dynamic range figures from an ADC you need to set your levels so that the peak signal level equates to 0dBFS. That’s totally right. But I think he may have misunderstood what you and I were trying to explain. Which was trying to achieve the most accurate representation of the amp in the amp sim.
That, and at 0 gain on your Apollo, your signal to noise is already more than sufficiently optimised. There would be no benefit to recording any hotter. If you were recording on a Behringer, I can't imagine you'd be recording with your peaks at -35dBFS. You were using the gear you have as correctly as is possible.
Na man …I think he understood the point, he was saying it’s possible to get an identical result with a lower noise floor. The sim accuracy is only affected by its input level. Not the recording level. The 0db thing is still valid though. It’s a consistent target and easy to use - it’s just not ‘optimal’. If there’s no discernible difference in practice it’s not worth worrying about.
@@fastboy_guitarsthe noise floor is constrained by the background noise of the pickups and how loud the peak already is. UAD Apollos are already on the verge of clipping at minimum gain and humbuckers. They also have a noisefloor that is well below the background noise of the pickups. So on both counts, there is no way to reduce the noisefloor further, in this case. On a behringer with single coils, sure. But Rabbea’s rig is already optimal.
Anyone creating plugins knows that the optimal input signal will be peaking a little bit below 0 dBFS. If you noticed that the plugin reacts differently, like for example a clean preset sounds way to distorted, simply lower the input gain of the plugin but NOT of your pre-amp/interface.
Yes I agree. The main problem is that if we all got the best signal to noise ratio first by turning up the gain on the interface, there is no reliable way to know what that actual increase is to then do an equal decrease on the plugin. You could obviously use your ears, but the point of setting it to zero is to get the exact like for like.
1.) I saw his video - and subsequently his reference to you being 'wrong'. I knew that you would have a response to this - instead of steering away from it - because you are a credible person. That's why we watch you. 2.) Keeping my Scarlett interface at 0 when using Neural plugins just sounds better (and appears to be consistent with the chart). So - in my case at least - you were both right.....and my tone is better.
This is a really helpful video, thanks. I use my ears for judging what the settings should be and that's ok for amps that I know, but obviously doesn't work for amps I'm less familiar with which is a real pain. Because of the huge variation across different plugins I have often had times when I'm getting really frustrated with the results. It's good to finally get some sort of sensible answer to this problem from somewhere. Now it makes sense to me because if plugin creators can only baseline against one or two interfaces because there's no standard, then what we currently all have to do as users is to refer to the cross reference spreadsheet. EDIT: I've just read @LTMuse comment in reply to John's pinned comment. I recommend everyone read it because it clarifies a hell of a lot about what the actual standards are and also how you might expect your interface to behave depending on which country the plugin vendor and/or interface manufacturer is based in. Ultimately atm you'll definitely still need to use your ears to compare what the plugin manufacturers demo video sounds like with what you are getting through your interface.
I think the most important part in this discussion is not to change your input gain when changing guitars. I am feeding my input with a 0dBu signal and set the interface to my desired dBfs level. Then I just compensate in the digtal realm once. I really like how polite you are, not talking down to others...
Well compensating is what we do with tube amps as well. Any amp or any pedal behaves differently on single coils versus humbuckers and so we reset the gain for those guitar. Same as when recording guitar you need to set the pre-amp of the interface accordingly .
@@didtoknan8128I didn’t feel GhostNote was necessarily talking down to others. He was just irritated that RUclipsrs kept spreading misinformation that went against even very basic beginner audio engineering principles. I’m glad we now have a scientific method of setting input gain without sacrificing signal to noise ratio for those who don’t want to do it by ear.
So glad you’re talking about this! I started messing with digital recording in DAWs a few years ago and everything is confusing, requiring hours of tweaks and experiments to get things sounding good. Appreciate the clarity here, now I can solidly blame the lack of views my videos get on my playing, not the tone. 😂
It just makes much more sense to me that way. The baseline is crucial to know for accuracy, but we also need to bring the cleanest signal possible first.. otherwise we're just degrading it for no reason. For some equipment, it may not make much difference, but calibrating your own interface by calculating the difference with a Level Meter and then do the maths to compensate negatively in post is definitely the way to go for optimal results. The thing that was cool about Ed's way was the convenience of doing the maths once and then just remember the value to enter in the plugin.. and then simply make sure the knob is truly at 0 before starting tracking guitar on a project (no risk of level inconsistencies in a song). Not really a problem, but I would really love to see audio interface developpers start the make interfaces that can lock its input gain setting for consistencies/ease of use when you're using the same instruments/mics regularly.
Ultimately what we need is an audio interfaces input to be identical to the input jack on a real amplifier and everything after that point to be the software plugin.
You get that, if you set the input level of the A/D-Interface so that at maximum level you stay in the green/yellow level range and after you ONCE set in plug-in input level correctly, so that at maximum input signal, you still have quite a bit of input headroom. Around 12-16dB should usually work fine.
@@Andreas_Straub no, you're misunderstanding, currently audio interfaces introduce another pre-amp to the signal chain. its like adding another pre-amp in front of your pre-amp on your 'real' guitar amp. so if this was a real guitar amp you have input jack, gain knob to control the gain of the input jack, then your normal pre-amp, then the master volume (power amp stage). What I'm talking is an audio interface that is literally the same as a guitar amps input jack and literally 'everything' after that is done in the amp sim plugin/software (so prem-amp and effects etc) I understand that might not be possible, but currently we have one too many pre-amps, either the pre-amp needs to go from the interface or it needs to go from the plugin, or there needs to be some sort of intelligence between the two to gain match correctly without all this manual wizardly and spreadsheets. As John has mentioned, if you want your amp sim model of your real amp to sound exactly like the real thing then this matters and is currently a mess of differing options and input values for each manufacture and no standards.
@@myturningpoint the amplifier in a reasonably good A/D system is usually so good that it can be treated as non-relevant for the signal. So treat it like a piece of wire in a real amp.
on a "real" amplifier you need to compensate the input as well. a strat sounds less hot than a humbucker guitar on any amp. so setting the input is something you always have to do for any guitar you use. same for microphones.
Greetings, I congratulate you, you did not fall into the ego trap, I know of other people who were offended because they were mentioned on another channel. Thank you for the information you give us and continue to be humble.
I'm glad the topic is being discussed from a plugin/hardware designers point of view because ultimately they're the ones that are best placed to engineer the solution!
The Audio Assault Amp Locker platform has a calibration routine built into it, and the base version is free. If you're trying to measure gain changes based on the audio interface volume, making use of that would be even easier than loading volume measurement plugins etc
Was going to mention that, would be trivial for plugin makers to add a calibration routine to their ampsims. Press a button, strum a few seconds, it determines if the signal it is getting is too hot, adjusts plugin input gain accordingly. Problem solved.
What I've started doing recently is adjust my interface input gain until I'm hovering around -20 dB RMS with my clean di. Then lower the plugin input gain so I'm in the green entirely (Helix native) and once my tones are set up, lower the output gain until I'm back at -20db RMS. This tends to work better for me. Feels and sounds better, helps with the noise floor, etc. Not sure how scientific this is but got this tip from a friend who has run front of house for some really massive artists. But once he shared that its sounded way better.
I watched both videos and am an engineer ;) The idea is that your audio interface translates analog into digital and it has a fixed number of bits to do it. You want to use as much of those bits and fill them with signal as possible. That will lower the signal to noise ratio. The noise will be filling the 'least significant bits', so the more your signal sticks out into the 'most significant bits' the better of you are. I tried it with a 3rd gen Focusrite 18i20 and found the same as you describe: with humbuckers you won't be able to add too much gain before clipping if you really dig into your strings hard. So setting the input gain to zero is pretty close to ideal / good enough. If you play (say) a strat exclusively, then you can improve your signal to noise quite a bit. Checking with a level meter in your DAW, say you add 8 dB of boost from your interface before AD conversion, then scale down the (digital) input gain with the same amount and you should be done. Now if you interchange humbuckers with single coils regularly while using the same presets, this becomes unpractical (if your humbuckers clip). I find that leaving the interface at zero works well enough.
That's actually just one part of the video, the other much more impactful part of the discussion was what Valdemar was totally agreeing on, which is that there is a necessity (and thanks to Ed S a method) to calibrate your audio interface/amp sim if you're trying to get anywhere close to "accuracy". That's the part that is getting left behind in the discussion of signal to noise
@@johnnathancordy That should not be a problem, you got 5 stages to adjust the input level, if you use the plugin in your DAW. If you set the input on your interface as low as possible, you gain all the noise of the preamp in the interface, so it is better to set this level to a decent volume without clipping the preamp. I set the volume in the plugin using the input level and then I set the gain and the output of the "amp" to create the sound I want. After that you can correct the output level again in the plugin without influencing the sound. Conclusion: trust your ears.
Input gain on the interface is for setting the level at which your audio will be presented to the converters. Input gain on the plugin is the level at which the plugin will see your audio. Input gain on the interface is not supposed to be your handy guitar boost. If you want more or less signal going into the plugin, use the plugin's input level.
Instead of making gain adjustments in the plugin, I think I would find it easier to just insert a gain plugin before the amp sim with presets for the pickup type and amp sim. So if I was recording a single coil into helix native I could slap my single-coil-helix-native gain preset before it and not worry about changing anything within Native itself.
I've been using a combination of these methods for a while, mostly because I got fed up of having to adjust Neurals input gain everytime I switch between presets. With the specs of my interface I need to add 6.9dbu to get in the ball park so I just add that at the interface instead and leave it. It's not "optimum" re noise floor but it works and I'm far enough away from clipping I don't have to worry. Now I can flick between presets without worrying about input gain on the plugin.
Same here… then i saw the discussion and set input gain to zero.. adjusted ALL of my PreSet i commonly use and was annoyed with some of the preset noisefloor levels and set ALL Gates to minimum to counter that BuT still noise will come through while playing… Overall i think i need to readjust everything without a gate, with level meter and wait for neural to come out w a tool for dummy guitarist like i am. 🤣
Once us guitar players started using audio interfaces and DAWs monitoring via studio monitors, we may have brought some bad habits moving from the in the room amp world, to the amp sim plug-in world. The LUFS spec was developed for a reason once music moved from analog to being digitized going back to the CD days. Simply put, our guitar player ears want to hear that nice saturated loud amp feel. But it doesn't translate well once you have a A to D conversion signal chain. His gain staging technique is on the right track if we want our plugins to have good amp sim reproduction😅
John, your playing remains me a bit Mateo Mancuso’s flow and musicality. Maybe without the pyrotechnics of him but with that melodic phrasing and the mix of legato and picking style. Very very nice.
I think that in this increasingly more digital effects era manufacturers should embrace 32-bit float interfaces as a standard. No input gain setting and that's it! Also HUGE dynamic range from the multiple ADCs and excellent SNR! The icing on the cake would be if there would be a standard for dBu voltage at 0dBFS. Then all of the instruments in the world would translate in the same exact way in the digital world.
The best solution would, that the plug-in manufacturers mark an input level in the plug-in which is used as a reference for "normal" chord playing. Also not precise, but a good starting point.
I use the Volt 2 by UA and it has an instrument switch, and I leave the gain alone unless im using a guitar with really low output pickups, I find that going this way is the best to keep the natural sound of the guitar
The software control panel for my interface tells me the exact amount of gain being added when I turn the knob. So pretty simple to see that value and subtract it in the plugin.
I think a good pooint is: JB? Just leave it be with Hi-Ω. Single coil? Gain it up to a comfortable level and reduce it by the same amount or whatever works in terms of getting the sim to respond as the producer of it intended. Idk
I think this is one of those situations where youtube clings to an idea and it starts getting misrepresented as more and more people talk about it. Yes, it was initially about getting the amp to sound right, but viewers often look at this stuff as how to actually set things up "correctly". Perhaps there's a point where being careful is important. I can only imagine how difficult it is to balance between getting a youtube video out and accidently spreading the wrong message. But, of course, this all comes down to the fact that there's no solid documentation on how to run levels, which you would think would be among the most important aspects of recording guitar.
I think we finally got this one solved boys! I really like the approach of using the loudness meter. Also since you guys were in Reaper you could have just used its loudness meter. This solves the issue I was conversing with Ed about regarding people using di boxes, since you're going through a mic level input, gain is very different. If I'm not mistaken this would also make this method plug-in agnostic.
@jonathonawarnberg maybe a misunderstanding but it sounds like you don't understand the difference between l u f s, dbfs and rms. Clipping has everything to do with the volume. In the video they are specifically measuring with lufs.
I tried this dude's method and it immediately made everything sound better. Less hiss on sustained notes, strong signal, not blowing out the amp sim. No complaints from me.
I have a Player Plus Strat (noiseless pickups) and a UA VOLT2, although i embrace the advice from the engineer guy, i find that adding 6 to 9db of gain on the interface (without compensating back on the plugin input) i get better response and tone
Glad this guy made this video. Setting the gain to zero is a bad idea. Your best SNR ratio is at maximum gain. I think all plugin vendors should mimic Overloud THU. In their plugin they provide a color graded meter indicating their expected input level and a knob to increase/reduce this level at plugin input. With this, you only need two steps: increase the gain on the interface just below clipping level, and then increase/decrease the input level in the plugin according to the meter "good" input zone.
If you work through the logic of that, the advice would lead to this conclusion: Stratocaster with low output pickups and a headless Strandberg with Fishman Fluences hitting the same peak. Signal to Noise wasn't what the initial discussion was about; calibrating input levels according to various audio interface specifications and the calibrated input level that various plugins are designed to be used at is actually far more impactful and the whole reason that for so many of us plugins never sounded anything like what they should have.
@@Andreas_Straub Incorrect. The best SNR is at maximum gain. In lower gain settings the SNR behaves non-linearly as you get closer to your ADC noise floor. You want instead to use the full range of your ADC. If you don't believe me, just listen to the vid that John is referring to. Pretty obvious the difference in noise from 0 gain versus almost clipping gain.
@@johnnathancordy I get that you can set 0 gain and then use the spreadsheet to calibrate, You will then need the other 2 steps the guys is explaining to be able to bring back the gain near clipping to reduce the noise. But if your plugin gives you an indication of what it expects as a proper input level, you don't need the table and the 0 gain step. If you don't want your Strat to come in as hard as your humbucker guitar, you still need to adjust a bit and reduce the plugin input level... But it's easier if the plugin provides a good starting point.
agree SNR at audio interface related to analog source before DAC and at highest gain we can achieve is the sweetspot for SNR, I think the most problematic here is the ampsim developer that not include meter indicator and input level in the input stage of the ampsim, so whenever we set the gain wrong we still can achieve the emulation that mimic real amp, because some big manufactured plugins like amplitube that have input signal indicator and level adjustment need to set the input at ampsim near clipping to get sound near real amps, that means the input level in amplitube was not correct, because some other plugins like bx rocrack that doesn't have input level, can emulate real amp closely with max achieveable gain before clipping in audio interface
The whole 'set your interface gain at 0' thing made a massive difference to the way the sims sounded in a good way, (I use S-gear and L6 Native), the downside was you ended up with a non existant waveform in the DAW, I think this guy's point of upping input gain on the interface and reducing the gain at the sim lets you get the right sound but you still end up with a healthy waveform you can edit.
The turning to 0 thing was the big breakthrough for me too! ...As soon as i did that, it made everything sound 1000x times better ! I've been banging my head against the wall for years, trying to figure out why all these amp sims i tried sounded like muddy ass with my input set for "right before clipping." I never thought about lowering the input gain in the amp sims because they have uv meters built in and weren't clipping or anything . This new way makes sense, same results, better noise floor, and being able to see you waveforms for editing again !
DAWs have the functionality to increase the audio waveform level. Cubase, Studio One, Reaper, Logic, Ableton Live (since V12) all do this. No need to gain up just to combat poor visibility.
I kind of think this idea of “accuracy” only really matters for a few kind of specific situations. For most of us, all that really matters is that we can get the sound that works for the part. In that case we set interface gain to just below clipping and set plugin input where it sounds good.
When I tried Plugins long ago, they sounded horrible, overdriven, distorted, and unusable. I tried different things, but I was a novice, and documentation was poor. I thought maybe the Plugins, my computer, Audio Interface, DAW, or Drivers, weren't good enough. I gave up trying Plugins. It's nice to get a better explanation. Thanks
You are recording a clean DI signal so makes sense to set your interface level just below clipping and get the best possible recording, because you can always change settings in the plugin or use a completely different plugin later. You cannot change the recording itself.
The best way to measure levels and gain staging is to use UV meter to get this accurate measurement. Guitar must seat in mix this noise you are referring probably will disappear after good gain staging your track. Magical 18db difference between stock gain meter and any UV meter. I saw this other video and Im totaly agree with.
Just start with gain knob at 0. Then palm mute your lowest string and chug hard. Your loudest chug should go up halfway on the input bar. If you're still above even at 0 gain on your interface TAKE AWAY more input on the amp sim. Contrary ADD gain slowly on your interface until you're at the middle. This will account for pickup types and interface types. But merely setting up either at 0 and hard coding the input knob at a number won't take all variables into play
@@johnnathancordy this is absolutely true but if at the end of the day are we to be more concerned with the accurate representation per the plugin manufacturer or do we just want it to sound how we want our preset to sound with the least amount of noise floor? I agree with you and will probably use the gain meter pre the plugin to match unity but I still really just want it to sound good to my ears rather than what the manufacturer considers it should like at any particular gain (input) level.
That's a fine approach to take, but I think it would be better to be making those decisions from an informed position rather than just because plugin manufacturers haven't tried to help their consumer base
The sim manufactures are rolling their eyes at these discussions. The problem is the incompetence of the users. But since their user base is incompetent, they should probably find a way to address it. It's a hilarious debate. Some creators have even framed it as a conspiracy by the sim manufacturers! My god, the humanity!
THE number one issue for me, I need to be able to INSTANTLY save all amp settings with the DAW project. If something like the Soldano X88-R Rackmount Preamp could do that, I'd record hardware guitar amp direct instead of using amp sims. Even my AXFX requires additional steps exporting and importing presets to do that.
Watched ghostnotes video several times last night. Watched Rhett's the night before. I stumbled into these videos searching for feedback on who has the best software... I did try ghostnotes method earlier this morning. He includes to say you should increase the output of the plug-in by ED S worksheet, before you reduce it. So ultimately you do reduce it, but not before utilizing ED S worksheet. If I followed that correctly.
So, @johnnathancordy, if I want to record the DI track for later use with some other plug-in, how do I accurately adjust the pre-recorded DI track for that plug-in?
I see 2 things being mixed up in these videos: recording input gain and plugin's "gain". Record clean signal without clipping, then pass the signal through any plugin you like and tune the plugin's "gain". If the recorded signal is good, you can tweak it later however you want. You can increase the input level in the plugin ("plugin gain"), or you can lower it, it's up to you. If you want to be on a safe side, just record with the guitar peaking at -6db. The modern analog-to-digital converters are good enough, and unless it's dirt cheap, it will give you a good quality sound with low noise floor. As mentioned in the video, the noise from the single-coil pickups will be much higher than the noise from the audio interface preamps.
This is the best advice on this video. Just hit -6 to -10 on your loudest playing and use the plugins input knob to tune the input to around -12db as recommended by plugin manufacturers.
I get what you are saying, I've just got a new computer, and all my Reaper files for guitar sound completely different? I'm going to have to start again, all due to impedance and input. I don't use specs though, never have, I always just use my eras and my experimentation. Some sims are better with the PC recording volume up nearly full, for a great sustain, but if you try that with some other sims it will sound far too processed and awful.
I'm using my Line 6 UX1 Interface that actually doesn't happen to have a gain knob for its instrument input lol How do I deal with this? Do I assume it's already at 0 since it's specifically made for guitar input?
Nobody mentioned guitar pickups. I have SD Black Winter in my Schecter which is hot AF. You can compare the signal coming from this one with let’s say single coil in Strat
We need plugins to have a Vu meter on the front and tell us how much gain compensation is required :) it's not that hard. Hope someone takes the lead on this
@@MarcoRaaphorst because I have no idea whether a plugin needs 0dbfs or something else? Different plugins react completely different to the same level of input gain (for the exact same amp) Moreover, I might use compressors or something in front of the amp plugin.. or a pedal plugin
@@pb25193 you are compensating in the wrong way. If the plugin or preset is faulty you need to reset the input level of the plugin and not of the pre-amp/interface
@@pb25193 you can change the input gain on the plugin. That parameter is meant for that, to compensate different guitar sand different levels of input (many people record at a way too low level so plugin manufacturers add an input gain so you can change it). The signal of the interface/pre-amp need to be optimal so you are recording with the least amount of noise and getting a good recording signal. This how you should do it for mics and for guitar as well.
@@MarcoRaaphorst having a vu meter will help me to decide exactly how much I need to set it to. If I go from single coil to humbucker for example, a vu meter can tell me how much I need to adjust it. I already do everything you are saying, I just load a separate vu plug-in. They could start including one with the plug-in.
The only option I have to reduce levels into the interface is using a DI as the guitar is already close to clipping at 0db on the interface. That or use a buffered pedal which is strange as interface claims to match impedance.
Avid's Eleven Rack, though a hardware unit, had automatic impedance matching on the guitar input and achieved the "feel". Sadly the cab modelling wasn't as good. Blackstar has an audio interface that is better matched for this sort of thing. Personally I don't use amp modeled plugins unless I'm on the go and writing/demoing but I still prefer my hybrid amp/pedal/multieffects floor unit to get the proper amp tone, with the convenience and wide selection of effects. Maybe now the info is out there, I might consider giving plugin s a try.
Overall, Ed S's method was partially correct. It's a good basis to start off of before we start to add gain on our interface, and then reduce the input gain on the plugin to get as close as we can. I was totally cool with leaving it at 0 db, but at least this will help with the extra noise when not using a gate.
It made the presets made by other people sound better - they always sounded awful when I turned the gain up on the audio interface which I had previously been used to. Helix Native sounds better now since all this zero on the interface stuff.
For me Ed S's work has been crucial in actually making plugins (especially Helix Native) actually useful. Now, if I'm also concerned about SNR I can apply what Valdemer has suggested too
Oh... once again they're talking about completely different things. This discussion is split between three completely different subjects but because of common terminology it's going wrong every time. Input level of your DAC audio interface, Input level in PLUGIN to make amp sim behave close to original physical amp and max volume levels in different audio interfaces. You don't have ANY audio interface in front of the real amp and _amp_ takes care about staging you guitar up. You control this with amp's GAIN knob. And this gain is somehow _calibrated_ to boost your signal from 0 to 10. Right ? Some plugins just _made_ to mimic this behaviour. To have input signal straight from your guitar without any gain staging. And analogue amps literally have some built-in calibration how your voltage "stages" from 0 to 10 on gain knob. This is the same dial as on your _audio interface_ input knob. But in DIGITAL domain you have such thing as signal/noise ratio on DAC. And _plugin_ is NOT a wired valve amp. It already deals with signal from DAC, not from your guitar so to speak. And it literally doesn't matter where to level your signal up : on your audio interface, in your DAW by gain utility _in front_ of plugin or in plugin by itself. You just have to KNOW what your particular plugin _expects_ as an initial input level. And for DAC the most optimal input level to reduce NOISE is just below clipping. So it's _better_ to do this on your audio interface because this is THE ONLY physical thing having its physical limitations. The rest in your computer is just a math. And EdS video is NOT about input level. It's about maximum input level of your audio interface and its corresponding range on plugin's audio interface. It has nothing to do with your input level. And discussions about real/virtual amp correspondence is here for years. I.e. with amps profiling (in tonex i.e.) you have no idea if your amp behaves as expected. You have no reference point. Here you have REAL Morgan amp and its virtual twin. And you might compare. But when you have Dumble profile on Tonex what would you compare with ? Or of you're trying some exotic amps you can't even find video on RUclips for ? The only way you might find out how it supposed to sound and behave is to listen to a records. But it's never only amp ! Player might have some boost in front or use load box or have weak/overclocked pickups. I.e. I have no idea why Mr. Cordy is talking about -18db input. My _standard_ Gibson 335 with neck _humbucker_ barely hits -24db. And if I crank my Volt1 just below clipping I have -6db -12db in my daw (what is declared as _normal_ for audio recording standards). And there's no such thing as "correct" input level in plugin. There's an input level when gain knob INSIDE plugin gives you similar results with _real amp_. That's why you have huge input dial in all NDSP plugins i.e. And you have input gain trim in Tonex etc etc. There's a video showing that for tonex levelling up input and gain is LITERALLY the same. If you make it louder before plugin - you just make it lower inside. It doesn't matter what your input level is if it's adjusted correctly to make amp sound just like real. You just have no reference point here except demo videos from plugin creators or real amps if you have one. But for _computers_ it's definitely better to set your input level on physical audio interface to make your audio interface working in _optimal_ conditions and reduce noise level. It's just how DAC's are supposed to be used. Once again - your audio interface is _the only_ physical thing you can't affect from software later. It's obviously correct to adjust your audio interface for best result. And later you do whatever you wish in your DAW/plugin. And EdS video is about different RANGES of gain for different sound cards. If your audio interface's max level is 12 and NeuralDSP's max level is 15 so your 50% of gain is 6db but NDSP's "noon" is 7.5. And their plugin on max output volume will hit the headroom of YOUR audio-interface. So your DAW is mapping your audio interface accordingly but it shifts initial meant volume level from creator's perspective. And Ed's video just tells you how to adjust input level in your plugin to achieve optimal re-mapping. It''s NOT about input level or about amp sound. It's about _mapping_ different max volume ranges between different audio interfaces. It tells you which interface plugin creators have been using to give you a reference point. Both Ed's and Valdemar's videos about _technical engineering stuff_ but plugin input gain staging is about amp sim behaviour from artistic perspective and its similarity to the real amp. Three completely different subjects ! Ranges, noise/signal ratio, amp-sim "similarity".
Thank you very much. However, what should I do with Marshall Plexi emulation (Softube) for UAD or Tweed Delux. They do not have the function of lowering/raising the level to the amplifier. In addition, in the UAD Arrow it is impossible to set the gain to zero on Hi-Z. I've already lost my mind with the levels. Help me figure it out, please.
I think i can help with one of these. The Softube Marshall Plexi emulation does have an input gain, it's just hidden. If you look in the top right corner of the UI you'll see three icons next to the 'cog' icon. These expand the UI. Click on the one that expands the UI in the left direction and a meter will appear that shows you the input level. Just grab the horizontal bar and move up or down to adjust. Hope this helps.
No idea how big the differences really are... I own 6 audio interfaces (not counting devices which are 'also' an audio interface), 2 x Apogee, 2 x MOTU, 2 x RME. I have connected the same guitar with the neck pickup wide open, same cable, same signal chain to each of them with all of them having input gain set to zero and preamps (if any) turned off. The maximum difference I get between these 6 interfaces is between 2 and 3 dB when I dig in like a mad person, playing 'normally' the differences hardly exceed 1 dB... I do not consider that a problem. The basic principle is simple - if you just use a plug-in for practicing, the signal to noise ratio is pretty irrelevant at room levels, just keep the interface at zero boost and call it a day. Works with HX Native, Neural plugins, Genome, Amplitube/Tonex, NAM Player - dynamics are there and the amps do sound as intended. For recording or higher output levels you will want to optimize the signal to noise ratio. In this case obtain the highest peak with the audio interface set to zero boost (lets call that 'g1'), then increase the gain on the audio interface so that the loudest signal stays just below clipping territory, then reduce the now digital input signal back to g1 before it hits any drive pedals or the amp within the plugin. If you do that using the plugin's input gain parameter, or any block that has a level control does not really matter - bits do not have a volume, you are just adjusting a number. Works all the time, and minor adjustments within the plugin do not instantly kill the tone or dynamics, within limits you can absolutely adjust to taste. The differences are smaller than a tube amp that has just been turned on vs the same amp running for three hours.
I think all this is telling us is….. there’s more than 1 way to set things up and a lot depends on what equipment you have, what you want to do and how you want it all to work together. I think it’s also worth watching the Jason Sadites video on this topic.
If you are clipping the clean signal, does it make sense? We're simply destroying the signal. Especially if we play very dynamic, clipping will be audible at higher strokes.
since ive had neural dsp, and some other free sims in the past, ive assumed you just leave the input gain at 0 unless youre having weird issues or something. turns out i was overdriving it by a good 7 or 8 db. i followed ghost notes instructions to figure out what i needed and i lost about 6db of noise (with the volume on the guitar up)
Didn't know there was a problem? always use the audio interface like a tape meter so adjust to just under clipping ish then check input on selected Chanel to make sure there no clipping there? so max input less noise just like with a tape machine? maybe I'm just old school but never had a problem with pluggins Cheers
I wonder if the guy from Scuffham s gear plug in would have some insight into this topic. That is a great plugin if you have never heard of it. I’d like to see you reach out to him. Thanks
The method of inputting 0 gain works well for me, and then adjusting the amp sim input as needed. The amps sims are very accurate sounding to my ears. I tried the method described by the electrical engineer and found it problematic. That was my experience anyway. I don’t think this discussion is over…
In video, gain is used to increase a signal to effectively see into dark corners. The cost is degradation of the image primarily by increased grain or noise or what have you. The ideal situation is to have a fast enough sensor so you don’t need to add gain. Guitarists who want distortion have the choice of a hot pickup, a hot sim, a hot input, or some mix of same. To my mind, somewhere in the chain there should be an accurate sounding clean which is then modified. My thinking is that you always want to be able to mix in clean some way. But I’m also a dickhead in a bedroom. So buyer beware.
Everyone is still wrong in my opinion unfortunately (not that I have a solution). He’s right that there’s no standardized measurement which is a problem, but there can’t be one since output levels of pickups also aren’t standardized; there can’t be a universal solution that wouldn’t cause clipping in some application. Null tests help paint a picture but in my opinion the only true way to dial in proper input levels is to set the real amp to as low as possible where you still have a desired tone, set the controls on the simulated amp to look identical to the real amp and the adjust the input control until the tone is similar. But this also defeats the purpose of amp sims helping you avoid buying the real thing
Isn’t this entire topic focused on eliminating the output level of the pickups from the equation? If you set the input level like explained in the original video referenced here, then every guitar still hits the amp sim differently with the exact volume it would hit a real amp and the controls will work the same (if modeled the same). Only now you get less noise than if you were to just set your interface to zero.
@ if you stick to one plugin manufacturer yes, they’re not standardized between plugin manufacturers is what I’m talking about. Like if you use Mixwave Mike Stringer and Neural DSP Gojira, despite sounding similar; they’ve been modeled at different input levels and impedance so you have to compensate
This whole thing is a massive turn off for me personally when it comes to amp sims. I just want to play. I want the amp sim software I paid for to sound like the amp it's simulating, and I don't want to have to analyse wave forms and dB meters and play with half a dozen levels to make that happen. I don't think it's unreasonable from a consumer standpoint to wish we didn't have to deal with this. The fact that Neural themselves had to update their long standing, incorrect advice on how to gain stage makes me wonder if even they know what the hell is going on. Even their revised advice of 'set to zero' is now in question. I write this as an owner of NDSP Parallax, Gojira, Tone King, Borgen BassKnob and AT5. Amp sim software is SO convenient because I need basically no physical gear other than an interface. It's great. But all this shit just to get them to sound like they're supposed to? It's ridiculous. It shouldn't be this dark art, spreadsheet rubbish.
You'd readjust the input gain on the plugin to account for the amount of gain you're adding from the volume knob (in many cases with many interfaces this step might not be necessary at all)
@@johnnathancordy humbuckers output more than single coils in general for example. if you maxed SNR on the interface, then two guitars with these different pickups would end up with equal output going into the plugin. the plugin wouldn't be responding like a real amp would to different guitars
My favorite thing about the dBu trend of 2024 is the confusion over such a simple concept. Dudes buying their first multimeter on Amazon and making spreadsheets instead of turning a knob and saying "nah not enough, eh thats better, whoa too much". Its almost as funny as audiophiles taking a bit of science and extrapolating it into a weird elaborate elitism gymnastics routine. Yes, accuracy is important to you but it is not important to the listener. 20+ years of amp sims and nobody has ever said an album didnt sound accurate. Sure, there are several "um ackshully" retorts and you are actually correct. Its just not that big of a deal, bro. Knowing what you want (and knowing how to get it) is much more important than knowing what the builder intended.
"Dickheads like me".... While this gets your point across.... we all have things we are good at or bad at. I have had trouble understanding how people find things I find easy, so hard. I find I get treated the same about other things. The number of times I have gotten told "figure it out" or "it's obvious" by people I can't even start to explain things I find easy to. Anyway, guitar amps VS FOH amps which is very much the same as amp input vs digital IF input. A guitar amp is mostly run at 105% for clean sounds and 150% is about where it runs out. A FOH amp should run at 10% to 20% max and never get close to peak. So a 100watt clean amp (I use a class D amp for example) with a digital amp modeler in front of it is the same as a 10 to 20watt guitar amp. For a digital IF, the input signal from the guitar with the loudest pickup selected and the volume at max with the hardest playing should still be -10db or less. bringing it up will not have any noticeable affect on S/N with the normal 24bit ADCs everyone has now. inside the computer, where the signal is now 32bit float, gain can be adjusted to whatever level a plugin needs to work right. The spreadsheet is a great place to start. Most guitar players are not sound engineers and should not be expected to have the know how an engineer has. Still, we can all learn more. I have used an analog modeler for years but it was a real learning point the first time I ran it into a side on a stereo amp into a 12inch speaker.... and went "Ah, thats the sound".
@@eds4754 thanks, but how do we know how much to reduce it after we're calibrated the audio interface input? It doesn't give any numerical values as you turn it up or down.
@@kisschicken ah annoyingly in the plugin the input gain doesn't show values. You just have to use a trim plugin before it. Make sure to submit a feedback request to UA to add a numerical readout for input levels as its important!
When I play my guitar for business or pleasure I do what I can to minimize my signal to noise ratio and then I make my amp sound the way I want it to sound by turning the knobs
Create a controversy, then get views trying solve the problem that really doesn't exist. Every piece of hardware has instructions of how to set input levels to optimize the signal. The software tells you how to set levels. I've never had issues and get great results.
I use both analogue setups and modelling setups. I kind get it, but also don’t get it about people’s obsession with nitpicking about noise with digital modelling setups. I guess I’d understand why someone who has only ever used modelling might expect some kind of perfect, final-production, processed quality in their real-time playing sound, but if someone has ever used and dealt with all the imperfections inherent in tube gear, why should the sound of digital plugins *have to be* flawless? Not taking a poke at Johnathan, just don’t get the attitude of people in general on the issue.
I have like 20 valve amps, and I WANT the plugins to behave like those do. The plugins are models of specific amps, and I want them to sound and behave like what I’m familiar with. Given that it takes me about 1 second to adjust the input knob, I don’t see what the downside is?
@@eds4754 I currently have 5 valve amps and I’ve owned a half dozen or more other ones over the past forty years, and none of them were any less noisy than an “uncalibrated” digital setup.
@@darwinsaye calibration is about achieving the same (as in, desired) gain response from the amp model as the reference amp. The discussion about noise is regarding A/D converters and noise floors, something different to calibration.
I think that the intent of this practice is being lost trying to compare this to making the plug-in sound accurate, because that’s not the case. The intent of this practice is to get the most detailed digital signal of your guitar into your plugin at 0db gain. We want to have the most accurate translation of the GUITAR, and this is why the plugins respond best this way. The boosted guitar signal is why the amp doesn’t sound right, not fault of the plug-in. Gain staging the analog to digital converter is correct, and reducing the gain at the input of the plugin corrects the signal of the guitar back to the proper input level.
9 months on from the initial discussion - there is some advice from a real expert. Important takeaways:
-As Ed S has been pioneering and his work was referenced in Valdemar's video - if you are concerned about the ACCURACY of your amp sim plugin (or wondering why your AC30 sim sounds like a Dual Rectifier - there is great reason to calibrate your input
-Valdemar is full in step with that (which was the whole thrust of our discussion in any case)
-Valdemar's recommended steps - 1) Set your Input to 0db 2) CALIBRATE YOUR PLUGIN ACCORDING TO THE SPECS IN ED'S SPREADSHEET (to ensure accuracy) 3) Add Gain on your interface to below clipping (this may not be necessary for MANY people) 4) Reduce the input gain in the plugin by the amount that you've added on the knob
Yeah, this guy just got a really crappy interface that has loads of noise to create a video saying everyone was wrong to set gain to zero. A video that would have been better titled 'why is Behringer kit so cheap?' Gain on zero works and won't create loads of noise for many interfaces (the pod XT doesn't even have a gain knob and it's not noisy dunno about the newer line 6 boxes that double as audio interfaces whether they've added a gain knob - if not, well they are all on zero too)
But (of course) putting interface input gain higher and lowering input gain on the plugin works too. The key thing is what the input gain for each plugin should be relative to the gain on zero for your particular interface that the spreadsheet provided.
To be fair, my interface has a shit noisefloor too, so maybe he was literally making the video for me haha
@@johnnathancordy Fair enough. I have an alesis io 2 that I might have a play with later (because that does have a gain control) I figured the 'gain on zero, input gain value from this spreadsheet' was about keeping the instructions simple and only having to change the input gain on each plugin.
One problem I see with setting the input gain 'just below clipping' is that can vary from guitar to guitar. So you might end up having to configure every time you swap guitars? If that were the case maybe getting your highest output guitar so it doesn't clip would work - even if the signal to noise is slightly higher for some lower output guitars?
@@michael1Yes, that's the misleading part of his video. He actually commented on his own video (in the comments section) "admitting that he cheated a bit" just to make a point.
He used the example with crappiest interface and gain knobs on the plugin maxed out, just to prove someone can reduce the noise floor and have and audible proof.
For most of decent interfaces in 2024, the reduction in noise floor would be inaudible.
So, in my opinion, there's no point changing the setting (because you have to do this process for all the guitars some is using) to get a 1 dB reduction of noise floor.
@@michael1 +24dB increased gain to me sounds like he's plugged into line input, not hi-z. The instrument input should have its own gain circuit before hitting the preamp. that, or he's using a passive DI box.
He's not technically wrong about these instructions but under most circumstances with even an entry-level interface, the noise floor is so low you wouldn't be able to hear it anyway. It's also just kind of a fiddly process, though in the end what matters is you liking what you hear coming out of your speakers.
Really awesome response video! 🙂 After all of this discussion, I will absolutely tell you that, sometimes setting your gain to its lowest settings will be "Just Fine"🙂 - The problem that had arisen was that people were taking this advice, which is a great shortcut to getting decent results, and claiming it was the *optimal* method of adjusting things.
The factors that make the biggest difference are;
1. if you are using a weak output pickup (or if you roll off the volume, like you showed).
2. if your interface has an unusually high instrument input level (meaning you need a stronger signal to drive it to full peak value).
3. If your interface has a lot of self noise.
4. if you're using very high gain ampsims.
In my demo, pretty much all of these elements were involved, and that shows the absolute worst case scenario that can occur.
Just as an aside, I know Steve Atkinson, creator of Neural Amp Modeler, has just announced some new features in NAM that will make calibrating input gain much easier, he's got some really clever thoughts on the process that we should be seeing in a future update!
So to sum this up… joh is looking for real amp like response from the modelers and you were looking for highest dynamic range or deepest noise floor..they are two different things..
You video was irrelevant to what john was claiming
@@AgapeSignalNo, not at all!
Both are after the "real amp like response", but the method RUclipsrs talked about half a year ago with setting the input to zero achieved that at the expense of a healthy signal to noise ratio.
Now he presented a new method that achieved the same thing without sacrificing a clean and healthy signal.
@@AgapeSignalexactly. And even though I commented on his video pointing out this confusion he's creating I don't think he understands it. Nor the rest of people cheering his "new method".
Fighting to lower the noise floor on bad interfaces it's one thing, compansating the input gain is a different topic.
@josuastangl7140
you misunderstood the point. Ghost Notes video is exactly about this. Lowering the noise floor. He's not taking in consideration any adjustments for the input level that would make an amp sim to sound and react like the real amp.
EdS spreadsheet is made to correct the input gain for different plugins.
@@OrangeMicMusic GhostNotes is working off of EdS spreadsheet, keeping the overall input gain identical.
He simply for example adds 6dB on the interface and reduces 6dB in software.
This way the input gain is still identical and optimal for the real amp feel, but now you have a lower noise floor.
His input gain values before the signal hits the amp sim are still identical to EdS's spreadsheet
It is really weird that a company like Neural DSP makes a 60 min comedy movie with an IMDB rating of 3.5/10 for a product they are about to release which is basically the same product they released before but smaller, but dont bother to clarify this argument with a 3 min video.
My brother is an audio engineer, and his response to everything going on was "expecting the manual to explain how to set your gain is like expecting the manual for your keyboard to teach you how to play piano."
I think the video that is being referenced was great, but it's too succinct for a guitarist to understand what they're actually doing and all of the why and how to set everything up when the spreadsheet doesn't cover your software. You really couldn't explain it all without a three hour video, which I would be surprised if someone hasn't made already, but it's probably about gain staging broadly, and you'd have to figure out how to apply that to guitar.
When I use a passive DI box though, a lot of my gain problems will go away.
Tbh, they're not the worst ones.
@@JewettMusic The issue is that they all tell you to set the gain bellow peak but that doesn't give you an accurate representation of the amps.
If you don't own any of the amps how do you know if it's accurate or not?
Rhett Shull's video was great in the sense that he compared an ampsim to the real amp.
@@andresilvasophisma that is the big issue. Gain staging is an expert's skill that requires experience, it can't be boiled down to "do one thing on every situation" and the amp sim manufacturers don't know the output of your pickups, which matters too
@@JewettMusic Thank you for a voice of sanity in this page. Guitar players suddenly seem to think they are audio engineers, and then complain unknowingly about their own incompetence. Also, a good DI box is a useful recommendation for this particular forum, even a cheapish one like a Radial, and it will inevitably clean up a lot of noise, too.
@JasonSadites response is a great addition to this conversation. In addition to a lack of standards, the other issue is a lack of manufacturer documentation combined with differing approaches applied by third parties who sell captures. A consortium to develop some standards would be a very good thing.
There are standards for half of the problem (the calibration between input level and digital full-scale); see my comment on John's pinned comment for the refs. But the standards are really recommendations; there is no enforcement mechanism. So your point about "lack of manufacturer documentation" is the real issue (also addressed in my comment). The better manufacturers do tell you enough to settle this half of the problem. Alas, the datasheets that Ghost Note showed in the video were bad examples in this regard. -Tom
The problem is as old as musical instruments themselves…the engineer designing the tool and writing the instructions doesn’t necessarily know the goal of the artist using the tool/instrument.
Really well said with humility and honesty. There’s no reason to apologize for getting an important (to us musicians anyway) discussion started. I always enjoy your video! Keep up the great work!
There are two different end goals.
Engineer is a measurable answer
Artist 100% subjective answer
Seems weird that an artist's end goal would be to hear the plugin sound the same as the real thing. Most established artists are usually looking for a unique sound.
Dear John - a load of the noise from your guitar that I usually hear on your channel is just digital clock noise from your interface (and possibly your footpedals) because you are too close to your desk. Test this - try standing two meters back from your desk and record your guitar that way. You will hear a much more drastic difference in your noise floor, and much less of that "vacuum cleaner" sound.
You can also get noise when your monitor isn't set to 100% brightness.
@@RyRyTheBassGuywhat? Why?
@JuanDavidArtal Something to do with how the dimming works. Kinda like how a dimmer switch can introduce a bunch of line noise.
It should also be noted that power supplies produce a ton of EMI..
@@RyRyTheBassGuydepends on the monitor. Some don't use PWM for dimming and others have better noise leakage.
@@benjaminashlin Right. I just was pointing out a variable that people can try to see if it makes a difference. Can't image its something that a lot of people would even consider.
Like, I have a pair of active monitors that introduce a ton of noise because they create a ground loop. Easy enough fix by just cutting the ground connection on the XLR (since they don't have a ground lift). I can only imagine how many people struggle with that and don't realize what's going on.
"a dickhead in the bedroom", "my wife will tell you just as much" - this killed me haha
moment of appreciation for our beloved and patient wifes/gfs coping with another GAS syndrome effects
I also saw that video the other day and it finally clicked for me. Originally I was like everyone else, wondering why this, for example, Bassman plug in sounded so darn dirty, then I saw your video and switched to leaving the Scarlet interface at 0 and the tones were right and proper and everything sounded so much better. However, the waveforms in my DAW were TINY, so darn small you could hardly see em, I play with a passive Strat and Jazz Bass, btw. So, I tried the new way the other day to get the interface up to near clipping, and it gives me a number for the increase, then I subtracted that number from the plug-in input, and it is finally OPTIMAL. I have the tones the way I want them, and I have big healthy waveforms, and better signal to noise ratio. I think we finally figured it out ya'll.
Thanks for leading me on this journey. Especially as a bassist primarily, figuring out guitar tones is so much more difficult than on bass, like they aren't the same thing at all when trying to get great tones. my mind was blown by that when I started learning guitar a couple years ago.
Your point about the waveforms is an important one - not because of the waveforms themselves, of course, but because the signal level that these amp sims are "calibrated" for isn't consistent with the signal level most other plugins are calibrated for.
Use the interface the way the ampsim creators tell you and, say, you want to run an LA2A emulation before a clean amp to get a more consistent funky compressed sound? Well, your level will be so low you'll need to push input gain on the LA2A a lot to get any compression at all, and then you'll have to lower output on the LA2A a ton to get it back to the level the amp is expecting. If the compressor adds saturation based on internal gain structure (like the LA2A often does), that can change your sound drastically.
@ this was enlightening, thank you
I guess I was trying to communicate based on the idea that we wanted to achieve as close to the amp as possible. Not really focusing on the signal to noise ratio. As I tried to demonstrate in my video comparing the plugin to the amp with the input at zero.
He’s completely correct in advising that in order to get the best noise and dynamic range figures from an ADC you need to set your levels so that the peak signal level equates to 0dBFS. That’s totally right.
But I think he may have misunderstood what you and I were trying to explain. Which was trying to achieve the most accurate representation of the amp in the amp sim.
That, and at 0 gain on your Apollo, your signal to noise is already more than sufficiently optimised. There would be no benefit to recording any hotter.
If you were recording on a Behringer, I can't imagine you'd be recording with your peaks at -35dBFS. You were using the gear you have as correctly as is possible.
Na man …I think he understood the point, he was saying it’s possible to get an identical result with a lower noise floor.
The sim accuracy is only affected by its input level. Not the recording level.
The 0db thing is still valid though. It’s a consistent target and easy to use - it’s just not ‘optimal’.
If there’s no discernible difference in practice it’s not worth worrying about.
@@fastboy_guitarsthe noise floor is constrained by the background noise of the pickups and how loud the peak already is. UAD Apollos are already on the verge of clipping at minimum gain and humbuckers.
They also have a noisefloor that is well below the background noise of the pickups. So on both counts, there is no way to reduce the noisefloor further, in this case.
On a behringer with single coils, sure. But Rabbea’s rig is already optimal.
Anyone creating plugins knows that the optimal input signal will be peaking a little bit below 0 dBFS. If you noticed that the plugin reacts differently, like for example a clean preset sounds way to distorted, simply lower the input gain of the plugin but NOT of your pre-amp/interface.
Yes I agree. The main problem is that if we all got the best signal to noise ratio first by turning up the gain on the interface, there is no reliable way to know what that actual increase is to then do an equal decrease on the plugin. You could obviously use your ears, but the point of setting it to zero is to get the exact like for like.
1.) I saw his video - and subsequently his reference to you being 'wrong'. I knew that you would have a response to this - instead of steering away from it - because you are a credible person. That's why we watch you.
2.) Keeping my Scarlett interface at 0 when using Neural plugins just sounds better (and appears to be consistent with the chart). So - in my case at least - you were both right.....and my tone is better.
This is a really helpful video, thanks. I use my ears for judging what the settings should be and that's ok for amps that I know, but obviously doesn't work for amps I'm less familiar with which is a real pain. Because of the huge variation across different plugins I have often had times when I'm getting really frustrated with the results. It's good to finally get some sort of sensible answer to this problem from somewhere. Now it makes sense to me because if plugin creators can only baseline against one or two interfaces because there's no standard, then what we currently all have to do as users is to refer to the cross reference spreadsheet.
EDIT: I've just read @LTMuse comment in reply to John's pinned comment. I recommend everyone read it because it clarifies a hell of a lot about what the actual standards are and also how you might expect your interface to behave depending on which country the plugin vendor and/or interface manufacturer is based in. Ultimately atm you'll definitely still need to use your ears to compare what the plugin manufacturers demo video sounds like with what you are getting through your interface.
I think the most important part in this discussion is not to change your input gain when changing guitars. I am feeding my input with a 0dBu signal and set the interface to my desired dBfs level. Then I just compensate in the digtal realm once.
I really like how polite you are, not talking down to others...
Well compensating is what we do with tube amps as well. Any amp or any pedal behaves differently on single coils versus humbuckers and so we reset the gain for those guitar. Same as when recording guitar you need to set the pre-amp of the interface accordingly .
Totally agree about your last sentence. GhostNoteAudio should learn from John.
@@didtoknan8128I didn’t feel GhostNote was necessarily talking down to others.
He was just irritated that RUclipsrs kept spreading misinformation that went against even very basic beginner audio engineering principles.
I’m glad we now have a scientific method of setting input gain without sacrificing signal to noise ratio for those who don’t want to do it by ear.
@@josuastangl7140 Yes. Like the myth that the output stage of Vox AC30 is a class A amplifier.
So glad you’re talking about this! I started messing with digital recording in DAWs a few years ago and everything is confusing, requiring hours of tweaks and experiments to get things sounding good. Appreciate the clarity here, now I can solidly blame the lack of views my videos get on my playing, not the tone. 😂
It just makes much more sense to me that way. The baseline is crucial to know for accuracy, but we also need to bring the cleanest signal possible first.. otherwise we're just degrading it for no reason. For some equipment, it may not make much difference, but calibrating your own interface by calculating the difference with a Level Meter and then do the maths to compensate negatively in post is definitely the way to go for optimal results. The thing that was cool about Ed's way was the convenience of doing the maths once and then just remember the value to enter in the plugin.. and then simply make sure the knob is truly at 0 before starting tracking guitar on a project (no risk of level inconsistencies in a song). Not really a problem, but I would really love to see audio interface developpers start the make interfaces that can lock its input gain setting for consistencies/ease of use when you're using the same instruments/mics regularly.
Ultimately what we need is an audio interfaces input to be identical to the input jack on a real amplifier and everything after that point to be the software plugin.
You get that, if you set the input level of the A/D-Interface so that at maximum level you stay in the green/yellow level range and after you ONCE set in plug-in input level correctly, so that at maximum input signal, you still have quite a bit of input headroom. Around 12-16dB should usually work fine.
@@Andreas_Straub no, you're misunderstanding, currently audio interfaces introduce another pre-amp to the signal chain. its like adding another pre-amp in front of your pre-amp on your 'real' guitar amp. so if this was a real guitar amp you have input jack, gain knob to control the gain of the input jack, then your normal pre-amp, then the master volume (power amp stage).
What I'm talking is an audio interface that is literally the same as a guitar amps input jack and literally 'everything' after that is done in the amp sim plugin/software (so prem-amp and effects etc)
I understand that might not be possible, but currently we have one too many pre-amps, either the pre-amp needs to go from the interface or it needs to go from the plugin, or there needs to be some sort of intelligence between the two to gain match correctly without all this manual wizardly and spreadsheets.
As John has mentioned, if you want your amp sim model of your real amp to sound exactly like the real thing then this matters and is currently a mess of differing options and input values for each manufacture and no standards.
@@myturningpoint the amplifier in a reasonably good A/D system is usually so good that it can be treated as non-relevant for the signal. So treat it like a piece of wire in a real amp.
on a "real" amplifier you need to compensate the input as well. a strat sounds less hot than a humbucker guitar on any amp. so setting the input is something you always have to do for any guitar you use. same for microphones.
Well UAD Unison plugins kinda do that, unfortunately they seem to have stopped developing as many Unison stuff...
Playing at the end is 🔥 !!!
This conversation is so helpful and has changed the way I use plugins for the better.
Greetings, I congratulate you, you did not fall into the ego trap, I know of other people who were offended because they were mentioned on another channel. Thank you for the information you give us and continue to be humble.
I'm glad the topic is being discussed from a plugin/hardware designers point of view because ultimately they're the ones that are best placed to engineer the solution!
THIS
The Audio Assault Amp Locker platform has a calibration routine built into it, and the base version is free. If you're trying to measure gain changes based on the audio interface volume, making use of that would be even easier than loading volume measurement plugins etc
Was going to mention that, would be trivial for plugin makers to add a calibration routine to their ampsims. Press a button, strum a few seconds, it determines if the signal it is getting is too hot, adjusts plugin input gain accordingly. Problem solved.
So do the Waves PRS sims
What I've started doing recently is adjust my interface input gain until I'm hovering around -20 dB RMS with my clean di. Then lower the plugin input gain so I'm in the green entirely (Helix native) and once my tones are set up, lower the output gain until I'm back at -20db RMS. This tends to work better for me. Feels and sounds better, helps with the noise floor, etc.
Not sure how scientific this is but got this tip from a friend who has run front of house for some really massive artists. But once he shared that its sounded way better.
I watched both videos and am an engineer ;) The idea is that your audio interface translates analog into digital and it has a fixed number of bits to do it. You want to use as much of those bits and fill them with signal as possible. That will lower the signal to noise ratio. The noise will be filling the 'least significant bits', so the more your signal sticks out into the 'most significant bits' the better of you are.
I tried it with a 3rd gen Focusrite 18i20 and found the same as you describe: with humbuckers you won't be able to add too much gain before clipping if you really dig into your strings hard. So setting the input gain to zero is pretty close to ideal / good enough.
If you play (say) a strat exclusively, then you can improve your signal to noise quite a bit. Checking with a level meter in your DAW, say you add 8 dB of boost from your interface before AD conversion, then scale down the (digital) input gain with the same amount and you should be done.
Now if you interchange humbuckers with single coils regularly while using the same presets, this becomes unpractical (if your humbuckers clip). I find that leaving the interface at zero works well enough.
That's actually just one part of the video, the other much more impactful part of the discussion was what Valdemar was totally agreeing on, which is that there is a necessity (and thanks to Ed S a method) to calibrate your audio interface/amp sim if you're trying to get anywhere close to "accuracy".
That's the part that is getting left behind in the discussion of signal to noise
@@johnnathancordy That should not be a problem, you got 5 stages to adjust the input level, if you use the plugin in your DAW. If you set the input on your interface as low as possible, you gain all the noise of the preamp in the interface, so it is better to set this level to a decent volume without clipping the preamp. I set the volume in the plugin using the input level and then I set the gain and the output of the "amp" to create the sound I want. After that you can correct the output level again in the plugin without influencing the sound. Conclusion: trust your ears.
@@NoCats-on-Guitars Conclusion: zero accuracy.
@@teatime6414 Don't panic, it's enough "accuracy" for a proper recording of the sound you want.
Input gain on the interface is for setting the level at which your audio will be presented to the converters. Input gain on the plugin is the level at which the plugin will see your audio. Input gain on the interface is not supposed to be your handy guitar boost. If you want more or less signal going into the plugin, use the plugin's input level.
Exactly
Instead of making gain adjustments in the plugin, I think I would find it easier to just insert a gain plugin before the amp sim with presets for the pickup type and amp sim. So if I was recording a single coil into helix native I could slap my single-coil-helix-native gain preset before it and not worry about changing anything within Native itself.
To me, it seems like an unnecessary step when you can just adjust the input on the plugin?
Very good idea!
Love the conversation. Thanks for putting the video up for the community.
I've been using a combination of these methods for a while, mostly because I got fed up of having to adjust Neurals input gain everytime I switch between presets. With the specs of my interface I need to add 6.9dbu to get in the ball park so I just add that at the interface instead and leave it. It's not "optimum" re noise floor but it works and I'm far enough away from clipping I don't have to worry. Now I can flick between presets without worrying about input gain on the plugin.
Same here… then i saw the discussion and set input gain to zero.. adjusted ALL of my PreSet i commonly use and was annoyed with some of the preset noisefloor levels and set ALL Gates to minimum to counter that BuT still noise will come through while playing…
Overall i think i need to readjust everything without a gate, with level meter and wait for neural to come out w a tool for dummy guitarist like i am.
🤣
Once us guitar players started using audio interfaces and DAWs monitoring via studio monitors, we may have brought some bad habits moving from the in the room amp world, to the amp sim plug-in world. The LUFS spec was developed for a reason once music moved from analog to being digitized going back to the CD days. Simply put, our guitar player ears want to hear that nice saturated loud amp feel. But it doesn't translate well once you have a A to D conversion signal chain. His gain staging technique is on the right track if we want our plugins to have good amp sim reproduction😅
John, your playing remains me a bit Mateo Mancuso’s flow and musicality. Maybe without the pyrotechnics of him but with that melodic phrasing and the mix of legato and picking style. Very very nice.
I think that in this increasingly more digital effects era manufacturers should embrace 32-bit float interfaces as a standard. No input gain setting and that's it! Also HUGE dynamic range from the multiple ADCs and excellent SNR!
The icing on the cake would be if there would be a standard for dBu voltage at 0dBFS. Then all of the instruments in the world would translate in the same exact way in the digital world.
The best solution would, that the plug-in manufacturers mark an input level in the plug-in which is used as a reference for "normal" chord playing. Also not precise, but a good starting point.
agree, and actually that's the best solution rather we debate about gain in audio interface
I use the Volt 2 by UA and it has an instrument switch, and I leave the gain alone unless im using a guitar with really low output pickups, I find that going this way is the best to keep the natural sound of the guitar
The software control panel for my interface tells me the exact amount of gain being added when I turn the knob. So pretty simple to see that value and subtract it in the plugin.
I think a good pooint is: JB? Just leave it be with Hi-Ω. Single coil? Gain it up to a comfortable level and reduce it by the same amount or whatever works in terms of getting the sim to respond as the producer of it intended. Idk
I think this is one of those situations where youtube clings to an idea and it starts getting misrepresented as more and more people talk about it. Yes, it was initially about getting the amp to sound right, but viewers often look at this stuff as how to actually set things up "correctly". Perhaps there's a point where being careful is important. I can only imagine how difficult it is to balance between getting a youtube video out and accidently spreading the wrong message.
But, of course, this all comes down to the fact that there's no solid documentation on how to run levels, which you would think would be among the most important aspects of recording guitar.
I think we finally got this one solved boys! I really like the approach of using the loudness meter. Also since you guys were in Reaper you could have just used its loudness meter.
This solves the issue I was conversing with Ed about regarding people using di boxes, since you're going through a mic level input, gain is very different. If I'm not mistaken this would also make this method plug-in agnostic.
@jonathonawarnberg maybe a misunderstanding but it sounds like you don't understand the difference between l u f s, dbfs and rms. Clipping has everything to do with the volume.
In the video they are specifically measuring with lufs.
Hey John just use the way you like, all your presets are AMAZING, thank you for everything you have done!
I tried this dude's method and it immediately made everything sound better. Less hiss on sustained notes, strong signal, not blowing out the amp sim. No complaints from me.
I have a Player Plus Strat (noiseless pickups) and a UA VOLT2, although i embrace the advice from the engineer guy, i find that adding 6 to 9db of gain on the interface (without compensating back on the plugin input) i get better response and tone
Glad this guy made this video. Setting the gain to zero is a bad idea. Your best SNR ratio is at maximum gain.
I think all plugin vendors should mimic Overloud THU. In their plugin they provide a color graded meter indicating their expected input level and a knob to increase/reduce this level at plugin input. With this, you only need two steps: increase the gain on the interface just below clipping level, and then increase/decrease the input level in the plugin according to the meter "good" input zone.
If you work through the logic of that, the advice would lead to this conclusion:
Stratocaster with low output pickups and a headless Strandberg with Fishman Fluences hitting the same peak.
Signal to Noise wasn't what the initial discussion was about; calibrating input levels according to various audio interface specifications and the calibrated input level that various plugins are designed to be used at is actually far more impactful and the whole reason that for so many of us plugins never sounded anything like what they should have.
The best signal to noise is at MINIMUM gain still giving you a near full range digital input signal.
@@Andreas_Straub Incorrect. The best SNR is at maximum gain. In lower gain settings the SNR behaves non-linearly as you get closer to your ADC noise floor. You want instead to use the full range of your ADC.
If you don't believe me, just listen to the vid that John is referring to. Pretty obvious the difference in noise from 0 gain versus almost clipping gain.
@@johnnathancordy
I get that you can set 0 gain and then use the spreadsheet to calibrate, You will then need the other 2 steps the guys is explaining to be able to bring back the gain near clipping to reduce the noise. But if your plugin gives you an indication of what it expects as a proper input level, you don't need the table and the 0 gain step.
If you don't want your Strat to come in as hard as your humbucker guitar, you still need to adjust a bit and reduce the plugin input level... But it's easier if the plugin provides a good starting point.
agree SNR at audio interface related to analog source before DAC and at highest gain we can achieve is the sweetspot for SNR, I think the most problematic here is the ampsim developer that not include meter indicator and input level in the input stage of the ampsim, so whenever we set the gain wrong we still can achieve the emulation that mimic real amp, because some big manufactured plugins like amplitube that have input signal indicator and level adjustment need to set the input at ampsim near clipping to get sound near real amps, that means the input level in amplitube was not correct, because some other plugins like bx rocrack that doesn't have input level, can emulate real amp closely with max achieveable gain before clipping in audio interface
The whole 'set your interface gain at 0' thing made a massive difference to the way the sims sounded in a good way, (I use S-gear and L6 Native), the downside was you ended up with a non existant waveform in the DAW, I think this guy's point of upping input gain on the interface and reducing the gain at the sim lets you get the right sound but you still end up with a healthy waveform you can edit.
The turning to 0 thing was the big breakthrough for me too! ...As soon as i did that, it made everything sound 1000x times better ! I've been banging my head against the wall for years, trying to figure out why all these amp sims i tried sounded like muddy ass with my input set for "right before clipping." I never thought about lowering the input gain in the amp sims because they have uv meters built in and weren't clipping or anything . This new way makes sense, same results, better noise floor, and being able to see you waveforms for editing again !
And as I mentioned in another comment, now your signal is around the level _basically every other plugin_ expects.
Spot on man
DAWs have the functionality to increase the audio waveform level. Cubase, Studio One, Reaper, Logic, Ableton Live (since V12) all do this. No need to gain up just to combat poor visibility.
Sorry, meant to say "audio waveform SIZE" not "level"
I kind of think this idea of “accuracy” only really matters for a few kind of specific situations. For most of us, all that really matters is that we can get the sound that works for the part. In that case we set interface gain to just below clipping and set plugin input where it sounds good.
When I tried Plugins long ago, they sounded horrible, overdriven, distorted, and unusable. I tried different things, but I was a novice, and documentation was poor. I thought maybe the Plugins, my computer, Audio Interface, DAW, or Drivers, weren't good enough. I gave up trying Plugins. It's nice to get a better explanation. Thanks
You are recording a clean DI signal so makes sense to set your interface level just below clipping and get the best possible recording, because you can always change settings in the plugin or use a completely different plugin later. You cannot change the recording itself.
The best way to measure levels and gain staging is to use UV meter to get this accurate measurement. Guitar must seat in mix this noise you are referring probably will disappear after good gain staging your track. Magical 18db difference between stock gain meter and any UV meter. I saw this other video and Im totaly agree with.
If you are using logic, just stick a gain utility plugin before the amp sim, easy to correct the signal there for amp input.
Just start with gain knob at 0. Then palm mute your lowest string and chug hard. Your loudest chug should go up halfway on the input bar. If you're still above even at 0 gain on your interface TAKE AWAY more input on the amp sim. Contrary ADD gain slowly on your interface until you're at the middle. This will account for pickup types and interface types. But merely setting up either at 0 and hard coding the input knob at a number won't take all variables into play
Kudos for the update 🤙
I saw his video yesterday. At this point I think we all need to just follow our ears - which is kind of where we all started. Idk
Following our ears doesn't really answer anything around accuracy?
@@johnnathancordy this is absolutely true but if at the end of the day are we to be more concerned with the accurate representation per the plugin manufacturer or do we just want it to sound how we want our preset to sound with the least amount of noise floor? I agree with you and will probably use the gain meter pre the plugin to match unity but I still really just want it to sound good to my ears rather than what the manufacturer considers it should like at any particular gain (input) level.
That's a fine approach to take, but I think it would be better to be making those decisions from an informed position rather than just because plugin manufacturers haven't tried to help their consumer base
The sim manufactures are rolling their eyes at these discussions. The problem is the incompetence of the users. But since their user base is incompetent, they should probably find a way to address it. It's a hilarious debate. Some creators have even framed it as a conspiracy by the sim manufacturers! My god, the humanity!
THE number one issue for me, I need to be able to INSTANTLY save all amp settings with the DAW project. If something like the Soldano X88-R Rackmount Preamp could do that, I'd record hardware guitar amp direct instead of using amp sims. Even my AXFX requires additional steps exporting and importing presets to do that.
Watched ghostnotes video several times last night. Watched Rhett's the night before. I stumbled into these videos searching for feedback on who has the best software... I did try ghostnotes method earlier this morning. He includes to say you should increase the output of the plug-in by ED S worksheet, before you reduce it. So ultimately you do reduce it, but not before utilizing ED S worksheet. If I followed that correctly.
Exactly, Ed S work is the ALL important first 2 steps
So, @johnnathancordy, if I want to record the DI track for later use with some other plug-in, how do I accurately adjust the pre-recorded DI track for that plug-in?
I see 2 things being mixed up in these videos: recording input gain and plugin's "gain".
Record clean signal without clipping, then pass the signal through any plugin you like and tune the plugin's "gain". If the recorded signal is good, you can tweak it later however you want. You can increase the input level in the plugin ("plugin gain"), or you can lower it, it's up to you.
If you want to be on a safe side, just record with the guitar peaking at -6db. The modern analog-to-digital converters are good enough, and unless it's dirt cheap, it will give you a good quality sound with low noise floor. As mentioned in the video, the noise from the single-coil pickups will be much higher than the noise from the audio interface preamps.
This is the best advice on this video. Just hit -6 to -10 on your loudest playing and use the plugins input knob to tune the input to around -12db as recommended by plugin manufacturers.
I get what you are saying, I've just got a new computer, and all my Reaper files for guitar sound completely different? I'm going to have to start again, all due to impedance and input. I don't use specs though, never have, I always just use my eras and my experimentation. Some sims are better with the PC recording volume up nearly full, for a great sustain, but if you try that with some other sims it will sound far too processed and awful.
I'm using my Line 6 UX1 Interface that actually doesn't happen to have a gain knob for its instrument input lol How do I deal with this? Do I assume it's already at 0 since it's specifically made for guitar input?
The only thing that significantly lowered my noise floor was a damn active di box. I still have to face away from computer for electric emi
Aren't you supposed to use an audio interface with specifically a high Z input?
i run amplitube 4/5 with my interface on 3/4 but its a zoom interface specifically for guitar , even looks like guitar head
Nobody mentioned guitar pickups. I have SD Black Winter in my Schecter which is hot AF. You can compare the signal coming from this one with let’s say single coil in Strat
At this point, I just prepare popcorn and watch the show of how every week a new concept is released to be "the one"
We need plugins to have a Vu meter on the front and tell us how much gain compensation is required :) it's not that hard. Hope someone takes the lead on this
Why? 0 dBFS is 0 dBFS on the computer/plugin. Unity gain.
@@MarcoRaaphorst because I have no idea whether a plugin needs 0dbfs or something else? Different plugins react completely different to the same level of input gain (for the exact same amp)
Moreover, I might use compressors or something in front of the amp plugin.. or a pedal plugin
@@pb25193 you are compensating in the wrong way. If the plugin or preset is faulty you need to reset the input level of the plugin and not of the pre-amp/interface
@@pb25193 you can change the input gain on the plugin. That parameter is meant for that, to compensate different guitar sand different levels of input (many people record at a way too low level so plugin manufacturers add an input gain so you can change it). The signal of the interface/pre-amp need to be optimal so you are recording with the least amount of noise and getting a good recording signal. This how you should do it for mics and for guitar as well.
@@MarcoRaaphorst having a vu meter will help me to decide exactly how much I need to set it to. If I go from single coil to humbucker for example, a vu meter can tell me how much I need to adjust it. I already do everything you are saying, I just load a separate vu plug-in. They could start including one with the plug-in.
The only option I have to reduce levels into the interface is using a DI as the guitar is already close to clipping at 0db on the interface. That or use a buffered pedal which is strange as interface claims to match impedance.
Avid's Eleven Rack, though a hardware unit, had automatic impedance matching on the guitar input and achieved the "feel". Sadly the cab modelling wasn't as good. Blackstar has an audio interface that is better matched for this sort of thing. Personally I don't use amp modeled plugins unless I'm on the go and writing/demoing but I still prefer my hybrid amp/pedal/multieffects floor unit to get the proper amp tone, with the convenience and wide selection of effects. Maybe now the info is out there, I might consider giving plugin s a try.
Apollo twin does this
Overall, Ed S's method was partially correct. It's a good basis to start off of before we start to add gain on our interface, and then reduce the input gain on the plugin to get as close as we can. I was totally cool with leaving it at 0 db, but at least this will help with the extra noise when not using a gate.
It made the presets made by other people sound better - they always sounded awful when I turned the gain up on the audio interface which I had previously been used to. Helix Native sounds better now since all this zero on the interface stuff.
For me Ed S's work has been crucial in actually making plugins (especially Helix Native) actually useful. Now, if I'm also concerned about SNR I can apply what Valdemer has suggested too
Oh... once again they're talking about completely different things. This discussion is split between three completely different subjects but because of common terminology it's going wrong every time. Input level of your DAC audio interface, Input level in PLUGIN to make amp sim behave close to original physical amp and max volume levels in different audio interfaces. You don't have ANY audio interface in front of the real amp and _amp_ takes care about staging you guitar up. You control this with amp's GAIN knob. And this gain is somehow _calibrated_ to boost your signal from 0 to 10. Right ? Some plugins just _made_ to mimic this behaviour. To have input signal straight from your guitar without any gain staging. And analogue amps literally have some built-in calibration how your voltage "stages" from 0 to 10 on gain knob. This is the same dial as on your _audio interface_ input knob. But in DIGITAL domain you have such thing as signal/noise ratio on DAC. And _plugin_ is NOT a wired valve amp. It already deals with signal from DAC, not from your guitar so to speak. And it literally doesn't matter where to level your signal up : on your audio interface, in your DAW by gain utility _in front_ of plugin or in plugin by itself. You just have to KNOW what your particular plugin _expects_ as an initial input level. And for DAC the most optimal input level to reduce NOISE is just below clipping. So it's _better_ to do this on your audio interface because this is THE ONLY physical thing having its physical limitations. The rest in your computer is just a math. And EdS video is NOT about input level. It's about maximum input level of your audio interface and its corresponding range on plugin's audio interface. It has nothing to do with your input level. And discussions about real/virtual amp correspondence is here for years. I.e. with amps profiling (in tonex i.e.) you have no idea if your amp behaves as expected. You have no reference point. Here you have REAL Morgan amp and its virtual twin. And you might compare. But when you have Dumble profile on Tonex what would you compare with ? Or of you're trying some exotic amps you can't even find video on RUclips for ? The only way you might find out how it supposed to sound and behave is to listen to a records. But it's never only amp ! Player might have some boost in front or use load box or have weak/overclocked pickups. I.e. I have no idea why Mr. Cordy is talking about -18db input. My _standard_ Gibson 335 with neck _humbucker_ barely hits -24db. And if I crank my Volt1 just below clipping I have -6db -12db in my daw (what is declared as _normal_ for audio recording standards). And there's no such thing as "correct" input level in plugin. There's an input level when gain knob INSIDE plugin gives you similar results with _real amp_. That's why you have huge input dial in all NDSP plugins i.e. And you have input gain trim in Tonex etc etc. There's a video showing that for tonex levelling up input and gain is LITERALLY the same. If you make it louder before plugin - you just make it lower inside. It doesn't matter what your input level is if it's adjusted correctly to make amp sound just like real. You just have no reference point here except demo videos from plugin creators or real amps if you have one. But for _computers_ it's definitely better to set your input level on physical audio interface to make your audio interface working in _optimal_ conditions and reduce noise level. It's just how DAC's are supposed to be used. Once again - your audio interface is _the only_ physical thing you can't affect from software later. It's obviously correct to adjust your audio interface for best result. And later you do whatever you wish in your DAW/plugin. And EdS video is about different RANGES of gain for different sound cards. If your audio interface's max level is 12 and NeuralDSP's max level is 15 so your 50% of gain is 6db but NDSP's "noon" is 7.5. And their plugin on max output volume will hit the headroom of YOUR audio-interface. So your DAW is mapping your audio interface accordingly but it shifts initial meant volume level from creator's perspective. And Ed's video just tells you how to adjust input level in your plugin to achieve optimal re-mapping. It''s NOT about input level or about amp sound. It's about _mapping_ different max volume ranges between different audio interfaces. It tells you which interface plugin creators have been using to give you a reference point. Both Ed's and Valdemar's videos about _technical engineering stuff_ but plugin input gain staging is about amp sim behaviour from artistic perspective and its similarity to the real amp. Three completely different subjects ! Ranges, noise/signal ratio, amp-sim "similarity".
Thank you very much. However, what should I do with Marshall Plexi emulation (Softube) for UAD or Tweed Delux. They do not have the function of lowering/raising the level to the amplifier. In addition, in the UAD Arrow it is impossible to set the gain to zero on Hi-Z. I've already lost my mind with the levels. Help me figure it out, please.
I think i can help with one of these. The Softube Marshall Plexi emulation does have an input gain, it's just hidden. If you look in the top right corner of the UI you'll see three icons next to the 'cog' icon. These expand the UI. Click on the one that expands the UI in the left direction and a meter will appear that shows you the input level. Just grab the horizontal bar and move up or down to adjust. Hope this helps.
UAD was smart with their Unison plugins. The instrument gain actually controls the amp sim gain, and not the input level.
Have you dived into sample rate though? At 88.2 khz it sounds a lot more natural and analogue. Try it! It was a game changer for me
No idea how big the differences really are... I own 6 audio interfaces (not counting devices which are 'also' an audio interface), 2 x Apogee, 2 x MOTU, 2 x RME. I have connected the same guitar with the neck pickup wide open, same cable, same signal chain to each of them with all of them having input gain set to zero and preamps (if any) turned off. The maximum difference I get between these 6 interfaces is between 2 and 3 dB when I dig in like a mad person, playing 'normally' the differences hardly exceed 1 dB... I do not consider that a problem. The basic principle is simple - if you just use a plug-in for practicing, the signal to noise ratio is pretty irrelevant at room levels, just keep the interface at zero boost and call it a day. Works with HX Native, Neural plugins, Genome, Amplitube/Tonex, NAM Player - dynamics are there and the amps do sound as intended. For recording or higher output levels you will want to optimize the signal to noise ratio. In this case obtain the highest peak with the audio interface set to zero boost (lets call that 'g1'), then increase the gain on the audio interface so that the loudest signal stays just below clipping territory, then reduce the now digital input signal back to g1 before it hits any drive pedals or the amp within the plugin. If you do that using the plugin's input gain parameter, or any block that has a level control does not really matter - bits do not have a volume, you are just adjusting a number. Works all the time, and minor adjustments within the plugin do not instantly kill the tone or dynamics, within limits you can absolutely adjust to taste. The differences are smaller than a tube amp that has just been turned on vs the same amp running for three hours.
The difference between you becoming a audio engineer in the details
@@Jerry-x2z Is this even a sentence?
@@uwedasler425 guess the truth hurts lol good luck bro
I think all this is telling us is….. there’s more than 1 way to set things up and a lot depends on what equipment you have, what you want to do and how you want it all to work together.
I think it’s also worth watching the Jason Sadites video on this topic.
Yep Jason had some good points!
Interestingly Fractal Audio devices do this gain into the converters and gain after the converters automatically to maintain unity gain.
If you are clipping the clean signal, does it make sense? We're simply destroying the signal. Especially if we play very dynamic, clipping will be audible at higher strokes.
A Dh who is my Favorite Uk player today. All from the Broom.
since ive had neural dsp, and some other free sims in the past, ive assumed you just leave the input gain at 0 unless youre having weird issues or something. turns out i was overdriving it by a good 7 or 8 db. i followed ghost notes instructions to figure out what i needed and i lost about 6db of noise (with the volume on the guitar up)
Didn't know there was a problem? always use the audio interface like a tape meter so adjust to just under clipping ish then check input on selected Chanel to make sure there no clipping there? so max input less noise just like with a tape machine? maybe I'm just old school but never had a problem with pluggins Cheers
I wonder if the guy from Scuffham s gear plug in would have some insight into this topic. That is a great plugin if you have never heard of it. I’d like to see you reach out to him. Thanks
Scuffham S Gear is great!
There’s a recent post up on this topic on the Scuffham forum
i just watched 5 of these videos and i think im just going to keep “turning knobs” til it sounds good like a normal person
Good Night.
Well and what is the correct method?.
The method of inputting 0 gain works well for me, and then adjusting the amp sim input as needed. The amps sims are very accurate sounding to my ears. I tried the method described by the electrical engineer and found it problematic. That was my experience anyway. I don’t think this discussion is over…
In video, gain is used to increase a signal to effectively see into dark corners. The cost is degradation of the image primarily by increased grain or noise or what have you. The ideal situation is to have a fast enough sensor so you don’t need to add gain. Guitarists who want distortion have the choice of a hot pickup, a hot sim, a hot input, or some mix of same. To my mind, somewhere in the chain there should be an accurate sounding clean which is then modified. My thinking is that you always want to be able to mix in clean some way. But I’m also a dickhead in a bedroom. So buyer beware.
When you say the interface at “0” do you mean at “0” or at noon? When I set my Focusrite at “0” (literally) I get no signal whatsoever.
I can never get helix native to sound as good as my hardware or any other modeler device .
Inside you are two wolves: one wants everything to be exactly like it should be in the real world, the other just wants it to sound good. Fight.
Everyone is still wrong in my opinion unfortunately (not that I have a solution). He’s right that there’s no standardized measurement which is a problem, but there can’t be one since output levels of pickups also aren’t standardized; there can’t be a universal solution that wouldn’t cause clipping in some application. Null tests help paint a picture but in my opinion the only true way to dial in proper input levels is to set the real amp to as low as possible where you still have a desired tone, set the controls on the simulated amp to look identical to the real amp and the adjust the input control until the tone is similar. But this also defeats the purpose of amp sims helping you avoid buying the real thing
Isn’t this entire topic focused on eliminating the output level of the pickups from the equation?
If you set the input level like explained in the original video referenced here, then every guitar still hits the amp sim differently with the exact volume it would hit a real amp and the controls will work the same (if modeled the same).
Only now you get less noise than if you were to just set your interface to zero.
@ if you stick to one plugin manufacturer yes, they’re not standardized between plugin manufacturers is what I’m talking about. Like if you use Mixwave Mike Stringer and Neural DSP Gojira, despite sounding similar; they’ve been modeled at different input levels and impedance so you have to compensate
This whole thing is a massive turn off for me personally when it comes to amp sims.
I just want to play. I want the amp sim software I paid for to sound like the amp it's simulating, and I don't want to have to analyse wave forms and dB meters and play with half a dozen levels to make that happen.
I don't think it's unreasonable from a consumer standpoint to wish we didn't have to deal with this.
The fact that Neural themselves had to update their long standing, incorrect advice on how to gain stage makes me wonder if even they know what the hell is going on. Even their revised advice of 'set to zero' is now in question.
I write this as an owner of NDSP Parallax, Gojira, Tone King, Borgen BassKnob and AT5.
Amp sim software is SO convenient because I need basically no physical gear other than an interface. It's great. But all this shit just to get them to sound like they're supposed to? It's ridiculous. It shouldn't be this dark art, spreadsheet rubbish.
As soon as I saw that guys video and he named you and Rhett etc etc I thought here we go 🍿
You didn’t bring up A/D conversion bit depth at all. I’m pretty sure that was the whole point
humbucker chads stay winning
If you're maximizing SNR for every guitar doesn't that negate the output differences between different pickup types?
You'd readjust the input gain on the plugin to account for the amount of gain you're adding from the volume knob (in many cases with many interfaces this step might not be necessary at all)
@@johnnathancordy humbuckers output more than single coils in general for example. if you maxed SNR on the interface, then two guitars with these different pickups would end up with equal output going into the plugin. the plugin wouldn't be responding like a real amp would to different guitars
My favorite thing about the dBu trend of 2024 is the confusion over such a simple concept. Dudes buying their first multimeter on Amazon and making spreadsheets instead of turning a knob and saying "nah not enough, eh thats better, whoa too much". Its almost as funny as audiophiles taking a bit of science and extrapolating it into a weird elaborate elitism gymnastics routine.
Yes, accuracy is important to you but it is not important to the listener. 20+ years of amp sims and nobody has ever said an album didnt sound accurate. Sure, there are several "um ackshully" retorts and you are actually correct. Its just not that big of a deal, bro. Knowing what you want (and knowing how to get it) is much more important than knowing what the builder intended.
"Dickheads like me".... While this gets your point across.... we all have things we are good at or bad at. I have had trouble understanding how people find things I find easy, so hard. I find I get treated the same about other things. The number of times I have gotten told "figure it out" or "it's obvious" by people I can't even start to explain things I find easy to. Anyway, guitar amps VS FOH amps which is very much the same as amp input vs digital IF input. A guitar amp is mostly run at 105% for clean sounds and 150% is about where it runs out. A FOH amp should run at 10% to 20% max and never get close to peak. So a 100watt clean amp (I use a class D amp for example) with a digital amp modeler in front of it is the same as a 10 to 20watt guitar amp. For a digital IF, the input signal from the guitar with the loudest pickup selected and the volume at max with the hardest playing should still be -10db or less. bringing it up will not have any noticeable affect on S/N with the normal 24bit ADCs everyone has now.
inside the computer, where the signal is now 32bit float, gain can be adjusted to whatever level a plugin needs to work right. The spreadsheet is a great place to start. Most guitar players are not sound engineers and should not be expected to have the know how an engineer has. Still, we can all learn more. I have used an analog modeler for years but it was a real learning point the first time I ran it into a side on a stereo amp into a 12inch speaker.... and went "Ah, thats the sound".
Does anyone know the max input volume for Amplitube 5?
What do we do about the new UA Audio plugins? They don't give you a number on input gain in the plugin.
12.2dBu, they did it to match their interfaces at 0 gain
@@kisschicken I put a volume trim plugin before the UA plugin to do the offset. Very weird how they have knobs and no values
@@eds4754 thanks, but how do we know how much to reduce it after we're calibrated the audio interface input? It doesn't give any numerical values as you turn it up or down.
@@kisschicken ah annoyingly in the plugin the input gain doesn't show values. You just have to use a trim plugin before it.
Make sure to submit a feedback request to UA to add a numerical readout for input levels as its important!
@@eds4754 thank you. any recommendations on a trim plugin? i'm in ableton if that matters.
I think all you people are overthinking this, start at zero, use your ears adjust according to taste. Period.
This is why you will never be an audio engineer
@sidvicious332 not trying to be, just a regular guitar player. Most of us are just playing guitar for pleasure.
@@Neuroanalisispeople who play for more than just pleasure might not want to follow your advice
When I play my guitar for business or pleasure I do what I can to minimize my signal to noise ratio and then I make my amp sound the way I want it to sound by turning the knobs
Create a controversy, then get views trying solve the problem that really doesn't exist.
Every piece of hardware has instructions of how to set input levels to optimize the signal. The software tells you how to set levels.
I've never had issues and get great results.
“Input trim” has been the most maddening thing with ToneX.
on The Studio Rats channel, they talk about input trim on ToneX. Maybe that can help you!
Incorrectly*
Adverbs 💪🏻😎
i don't trust companies that just pushing out new plugins constantly. i like how DSP have gone and upgraded their stuff instead.
I use both analogue setups and modelling setups. I kind get it, but also don’t get it about people’s obsession with nitpicking about noise with digital modelling setups. I guess I’d understand why someone who has only ever used modelling might expect some kind of perfect, final-production, processed quality in their real-time playing sound, but if someone has ever used and dealt with all the imperfections inherent in tube gear, why should the sound of digital plugins *have to be* flawless? Not taking a poke at Johnathan, just don’t get the attitude of people in general on the issue.
I have like 20 valve amps, and I WANT the plugins to behave like those do. The plugins are models of specific amps, and I want them to sound and behave like what I’m familiar with. Given that it takes me about 1 second to adjust the input knob, I don’t see what the downside is?
Bragging
@@johnnathancordy 19 of them are spider valves
@@eds4754 I currently have 5 valve amps and I’ve owned a half dozen or more other ones over the past forty years, and none of them were any less noisy than an “uncalibrated” digital setup.
@@darwinsaye calibration is about achieving the same (as in, desired) gain response from the amp model as the reference amp.
The discussion about noise is regarding A/D converters and noise floors, something different to calibration.
Step 1,000,000: wonder why your audio interface is not listed in the chart you have to use to make this work; get frustrated; use your ears.
great
I think that the intent of this practice is being lost trying to compare this to making the plug-in sound accurate, because that’s not the case. The intent of this practice is to get the most detailed digital signal of your guitar into your plugin at 0db gain. We want to have the most accurate translation of the GUITAR, and this is why the plugins respond best this way. The boosted guitar signal is why the amp doesn’t sound right, not fault of the plug-in. Gain staging the analog to digital converter is correct, and reducing the gain at the input of the plugin corrects the signal of the guitar back to the proper input level.