Sascha Franck
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No offense taken, but seriously, in case I need to record things quickly, I don't hesitate to cut and trim a lot - and a decent waveform display defenitely helps quite a bit.
Calibrating Input Level for Plugins
- Thread starter James Freeman
- Start date
No offense taken, but seriously, in case I need to record things quickly, I don't hesitate to cut and trim a lot - and a decent waveform display defenitely helps quite a bit.
2dor
Roadie
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You have a couple advantages here.
First one is that your gear is great. Your Motu is fantastic, so noise will not be an issue for you.
Second is that you don´t give such an importance to accuracy. As you said several times, you prefer getting the tone you like rather than sticking to the most accuracy setting as possible. This gives you freedom to tweak as you like.
![[Nathan]](/data/avatars/m/0/613.jpg?1726885163){kind=avatar}
[Nathan]
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Yeah I seem to have modes now. If I’m practicing or riffhunting I’ll just pull up a neuraldsp or STL plugin cause the default sounds do the job
I don’t do any NAM hunting but I’ll try out 2dors packs or something that catches my interest. That mode is also the kind of mode I’ll test ampsims out (new and old)…. Not really trying to achieve anything just mucking around and enjoying.
Doing captures is another mode and finally mixing. I know it sounds like a lot but I seem to have more distinct goals. So many times in the past I’d start testing out ampsims only to be up at 1am trying to figure out a riff…. Or vice versa trying ti write a song only to have 6 ampsims up auditioning things haha.
The main point is that if I’m writing/recording then I’ll literally use anything cause I know it’s a DI and will get replaced later. When I get into mixing mode I’ll care about NAM profiles or sims
I don’t do any NAM hunting but I’ll try out 2dors packs or something that catches my interest. That mode is also the kind of mode I’ll test ampsims out (new and old)…. Not really trying to achieve anything just mucking around and enjoying.
Doing captures is another mode and finally mixing. I know it sounds like a lot but I seem to have more distinct goals. So many times in the past I’d start testing out ampsims only to be up at 1am trying to figure out a riff…. Or vice versa trying ti write a song only to have 6 ampsims up auditioning things haha.
The main point is that if I’m writing/recording then I’ll literally use anything cause I know it’s a DI and will get replaced later. When I get into mixing mode I’ll care about NAM profiles or sims
Sascha Franck
Rock Star
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Btw, on a sidenote (not sure whether I mentioned it in this thread already), perhaps important to some folks (namely those using Logic): Logic 11 comes with a most (!) horrible bug related to waveform displaying.
Once a file is lower than something around -40dB (still need to measure the exact value), it won't display anything. Even if you non destructively raise the region gain as much as possible (+30dB), this will not change anything (region gain usually is reflected in the arrangement page's waveform display). Note: -40dB (or even lower) is a pretty common value as soon as you use your guitar volume a lot (*check* in my case).
And as if that wasn't enough, when you open Logic's audio editor and try to just normalize the file, it'll tell you that the file would contain absolute silence and simply refuse to do anything. No, I'm not kidding you! Interestingly enough, when you have it search for a peak level, it'll tell you the right value. And it'll also allow you to raise the level destructively in the editor. And at one point, the waveform will start to show up magically, too (it's pretty much like a switch). Hence, it's also using different methods to find a file's level for the normalization and peak searching functions - which is completely absurd, really.
Apple must be aware of that issue and refuses to fix it for several months by now. Which is just ridiculous. To put it mildly.
At some times I really wish I wasn't a Logic user. This is one of these times.
Once a file is lower than something around -40dB (still need to measure the exact value), it won't display anything. Even if you non destructively raise the region gain as much as possible (+30dB), this will not change anything (region gain usually is reflected in the arrangement page's waveform display). Note: -40dB (or even lower) is a pretty common value as soon as you use your guitar volume a lot (*check* in my case).
And as if that wasn't enough, when you open Logic's audio editor and try to just normalize the file, it'll tell you that the file would contain absolute silence and simply refuse to do anything. No, I'm not kidding you! Interestingly enough, when you have it search for a peak level, it'll tell you the right value. And it'll also allow you to raise the level destructively in the editor. And at one point, the waveform will start to show up magically, too (it's pretty much like a switch). Hence, it's also using different methods to find a file's level for the normalization and peak searching functions - which is completely absurd, really.
Apple must be aware of that issue and refuses to fix it for several months by now. Which is just ridiculous. To put it mildly.
At some times I really wish I wasn't a Logic user. This is one of these times.
TheTrueZoltan!
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BenIfin
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Holy shit balls.
I *finally* took the advice from this thread after having given up on making Helix Native sound any-good with my RME UCX ... everything I did it either souned weak or hyper-gained.
Long story short ..... I gained the RME Input to +8db to just avoid clipping ... looked up the spreadsheet which suggested +1.5db for RME and Native ..... then set the Native Input Trim to -6.5db being +1.5 minus 8db ... holy f*ck ... it sounds freakin amazing.
Why did I wait so long !!!!!!!!!!!!!!!!!
I *finally* took the advice from this thread after having given up on making Helix Native sound any-good with my RME UCX ... everything I did it either souned weak or hyper-gained.
Long story short ..... I gained the RME Input to +8db to just avoid clipping ... looked up the spreadsheet which suggested +1.5db for RME and Native ..... then set the Native Input Trim to -6.5db being +1.5 minus 8db ... holy f*ck ... it sounds freakin amazing.
Why did I wait so long !!!!!!!!!!!!!!!!!
[Nathan]
Roadie
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Good you got there!
Does it sound any different to leaving the gain at 0 and having helix on +1.5? Obviously the SnR offset works in your position but I’d be interested to know if you really hear any difference when the gains at 0.
Helix obviously has gold inside it, but some of the UI clunk and sliders still get to me…. But I still use it especially for pedals and fx
James Freeman
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Good good!
Some plugins sound worse than others with incorrect input gain, a lot of people are not aware of that fact.
Very important if you are using a real SD-1 (or other drive pedals) into plugins.
BenIfin
Roadie
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Thanks lads [and ladies] !
Yeah ... I had literally given up on Native and just thought my RME didn't "play well with it :) ... as my RME UCX only has a 470k Input Impedance [not 1M Ohm] .... it was only ever thin and weaker or super-gainey-squashed.
After all my previous totally wrong settings etc..... I wiped and re-set everything in my RME Totalmix settings ..... I first did the Gain on 0 db and Helix Trim on +1.5db ... and it sounded really really really good.
I then did A/B the 0db / +1.5db settings -vs +8db / -6.5db settings and whilst quite close .... the latter settings did sound clearer both in terms of the total sound and the gain the louder I played them.
As I cranked the playback through my home monitors, when I cracked the studio Volume way up *loud* ..... the later settings [+8db / -6.5db] stayed "in-tact" and "pleasant" to my ears ..... whereas the prior [0db / +1.5db] settings sounded weaker and thinner the louder they got.
If my RME UCX had a 1M Ohm Input Impedance I suspect it would sound even tighter and clearer.
Its obviously all subjective ..... but to me ... my ears weren't lying.
Apologies for all the fluffy / non-technical / non-musical descriptors.
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2dor
Roadie
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No need to apologize - it's a real eye opener when you nail it & it clicks. Everything sounds much, much better. EoB stuff that should clean up nicely with the pot now really does so etc.
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BenIfin
Roadie
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Yep - totally :)
Now I've got it "working" as it is supposed to .... it is utterly astonishing how great and versatile Native is and sounds !
laxu
Rock Star
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I want to try making a few NAM captures of my BluGuitar Amp 1. Can someone help me run through setting my levels so I get it right from the start? I plan to try the Tonezone3000 for actually creating the capture.
The signal chain will be Audient EOV 8 line out -> BluGuitar Amp 1 ME -> Bluetone Loadbox -> Audient EVO 8 line in.
The EVO 8 specs say max input level is +10 dBu for DI, +16 dBu for mic/line. How should I set the outgoing and incoming levels on the audio interface?
Also, if I'm playing through other people's NAM captures and the captures say they're done with a +13 dBu signal, is the way to calibrate to that as simple as adding 3 dB gain from my audio interface or raising the NAM plugin input knob by +3 dB?
The signal chain will be Audient EOV 8 line out -> BluGuitar Amp 1 ME -> Bluetone Loadbox -> Audient EVO 8 line in.
The EVO 8 specs say max input level is +10 dBu for DI, +16 dBu for mic/line. How should I set the outgoing and incoming levels on the audio interface?
Also, if I'm playing through other people's NAM captures and the captures say they're done with a +13 dBu signal, is the way to calibrate to that as simple as adding 3 dB gain from my audio interface or raising the NAM plugin input knob by +3 dB?
MirrorProfiles
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your outgoing level should be with the fader at 0dBFS, basically the loudest signal you can output through your reamp chain. If you can run a sine wave out of the reamp chain and note the dBFS level and the voltage (RMS AC) you can calculate the level hitting the amp.
For your DI level (when using the model), with gain at 0dB, you would need to lower your signal 3dB, not raise it. 10dBu means your signal is closer to 0 than 13dBu of headroom.
MirrorProfiles
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@laxu
You can also work out your reamp level without the need for a multimeter by running the reamp signal into your DI with input gain at minimum (10dBu of headroom) but its a bit more fiddly.
You can run a sine wave at 0dBFS (i.e. maximum level) from your output, run that into your input chain. If that is clipping, then your reamp level is higher than 10dBu - from there you could keep lowering the sine wave output level until it gets to -0.1dBFS on the input level, and then add whatever level in dB you had to reduce the output by to 10dBu to get the reamp level.
If your reamp level at 0dBFS is below clipping when coming back in to your instrument input, it means your reamp signal is less than 10dBu. So whatever level it is below 0dBFS, take that from 10dBu (so if it comes back in at -7dBFS, it means your reamp level would be 3dBu).
You can also set your reamp send to match your 10dBu input level by adjusting the knob on the reamp box so your reamp level going out matches the DI level going in.
You can also work out your reamp level without the need for a multimeter by running the reamp signal into your DI with input gain at minimum (10dBu of headroom) but its a bit more fiddly.
You can run a sine wave at 0dBFS (i.e. maximum level) from your output, run that into your input chain. If that is clipping, then your reamp level is higher than 10dBu - from there you could keep lowering the sine wave output level until it gets to -0.1dBFS on the input level, and then add whatever level in dB you had to reduce the output by to 10dBu to get the reamp level.
If your reamp level at 0dBFS is below clipping when coming back in to your instrument input, it means your reamp signal is less than 10dBu. So whatever level it is below 0dBFS, take that from 10dBu (so if it comes back in at -7dBFS, it means your reamp level would be 3dBu).
You can also set your reamp send to match your 10dBu input level by adjusting the knob on the reamp box so your reamp level going out matches the DI level going in.
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laxu
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So I have it routed like this for testing, using Logic Pro:
Track 1 - DI:
- Guitar -> Audient EVO 8 Input 1 instrument.
- Track: Input 1 -> Output 3.
- EVO 8 Output3 -> Lehle P-Split.
- Lehle P-Split ISO out with ground lift ON -> BluGuitar Amp 1 input.
- BluGuitar Amp 1 speaker out -> Bluetone Loadbox. Loadbox line-out level is maxed.
- Bluetone Loadbox line-out -> EVO 8 Input 3. As minimal gain I need from the preamp as it takes to peak at 0 dB.
- Track: Input 3 -> ML Sound Lab MIKKO 2 cab sim plugin -> Stereo output.
I can see Output 3 is sending a signal just below clipping, but the incoming signal through the Loadbox is still something like 15 dB less than what I get with guitar -> Amp 1. It is also audibly way less overdriven.
MirrorProfiles
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So what you need to do is:
- put a signal generator on track 1, route it to output 3. (probably want to turn your monitors off just to be safe), set the level of the sine wave so it is at 0dBFS.
- run a line level cable from output 3 to your Lehle, as you are doing.
- rather than plugging your Lehle to the amp, plug this cable straight to your audient instrument input.
- follow my steps above
This is to determine the level hitting the amp, it’s the one that matters. The level coming from the loadbox isn’t really important for capturing an amp, if you’re doing a pedal capture and want to retain accurate output level modelling then you’d consider it.
Once you know the level hitting your amp, you can determine how much you (or anyone else) needs to adjust to reach unity.
For the level coming from the loadbox, just avoid clipping
laxu
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For testing, I hooked up my Gigrig Wetter Box to use as an A/B box. Turning it on (signal through audio interface -> amp) vs off (guitar straight to amp), there's just so much less gain. Surely this won't result in accurate captures, if the input level into the amp is always wrong?
I found some Amp 1 NAM captures on Tone Hunt and those have the same issue - I had to basically crank the input gain knob on the plugin to make those captures behave anything like the amp with guitar connected straight.
MirrorProfiles
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What is the maximum dBu level hitting the amp? People using IK reamp boxes are making models with it as low as -6dBu which is way lower than a guitar and they turn out ok still. Ideally it would be around 12.
But it’s just easier if you measure it and have a definitive value to shoot for
Have you thought about approximating your output levels by using the instrument input as reference? It looks like the Evo 8 DI at minimum gain is 10dBu. You could connect the output via the P-Split into your instrument input, and pull up a tone generator plugin at something like -12dBFS.
It looks like the EVO 8’s outputs are 11dBu. Try to see how much you need to boost the unattenuated output get to unity gain (the -12dBFS reference tone level). If the P-Split doesn’t attenuate the signal at all, you’d need to boost around 1dB to get to unity based on the specifications. If there is some attenuation from the P-Split then you’d have to boost a bit more.
