Input levels spike to red with 0 input gain

...So there is no actual input gain staging ...that is now confusing me more...

In any case I now have all my levels within visually decent levels without pasting the red on the outputs and hitting the orange pretty steady on the input, still can't make a red happen on the input led's though ...but the sound is glorious ...so I a can live with the confusion.

Cheers,
Ken
I'm not quite sure what you think you would gain stage?

In a traditional guitar rig, what would you do? This is what the Axe Fx models.

The Instrument Level setting (not Input Trim, which is a setting within the Amp block - I think Cliff had a brain fart in his post ;)) is for optimizing the Analog to Digital converters for signal to noise. It does not affect the input gain (volume) or the tone.

You would typically increase gain before the amp with Drive or boost pedals, or EQs, etc. No different here...

You can then increase the gain in various ways in pretty much every block.
 
Let me see if I can clear this up.

The Instrument Level is to make it so the A/D converter hears the best signal it possibly can. So say you have a super low output Strat. You crank up the input level so that that Strat is hitting the converter at a level that is making sure it's well above the (very low) noise floor of the Axe-Fx. The A/D does its work and then brings down the signal it outputs to the processor by the same amount you gained up, so that what is coming in is going out, regardless. Conversely, if you have a super hot guitar, like an EBMM JP15, it's already hitting the A/D way above the noise floor, and you don't want to add unnecessary noise by having the input higher than it needs to be to convert the signal at an optimal level. So you turn the input level down a good bit. The converter then compensates for how much you turned down by bringing up the signal by an equal amount before it outputs to the processor. The signal hitting your grid (i.e. pedals, amps, whatever) is in theory unchanged in level from what is coming out of your guitar. You've just optimized the level at which it's being A/D converted.


You "can't" really "clip" the input given that it takes drive pedals and what-have-you just as well as an amp does.

Per Cliff: "The Axe-Fx III input was designed to mimic a typical tube amp input using an average of Marshall and Fender amps for the component values."
 
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Let me see if I can clear this up.

The Instrument Level is to make it so the A/D converter hears the best signal it possibly can. So say you have a super low output Strat. You crank up the input level so that that Strat is hitting the converter at a level that is making sure it's well above the (very low) noise floor of the Axe-Fx. The A/D does its work and then brings down the signal it outputs to the processor by the same amount you gained up, so that what is coming in is going out, regardless. Conversely, if you have a super hot guitar, like an EBMM JP15, it's already hitting the A/D way above the noise floor, and you don't want to add unnecessary noise by having the input higher than it needs to be to convert the signal at an optimal level. So you turn the input level down a good bit. The converter then compensates for how much you turned down by bringing up the signal by an equal amount before it outputs to the processor. The signal hitting your grid (i.e. pedals, amps, whatever) is in theory unchanged in level from what is coming out of your guitar. You've just optimized the level at which it's being A/D converted.

You "can't" really "clip" the input given that it takes drive pedals and what-have-you just as well as an amp does.

Per Cliff: "The Axe-Fx III input was designed to mimic a typical tube amp input using an average of Marshall and Fender amps for the component values."
"

That's an AWESOME description and AMAZING function.
 
Let me see if I can clear this up.

The Instrument Level is to make it so the A/D converter hears the best signal it possibly can. So say you have a super low output Strat. You crank up the input level so that that Strat is hitting the converter at a level that is making sure it's well above the (very low) noise floor of the Axe-Fx. The A/D does its work and then brings down the signal it outputs to the processor by the same amount you gained up, so that what is coming in is going out, regardless. Conversely, if you have a super hot guitar, like an EBMM JP15, it's already hitting the A/D way above the noise floor, and you don't want to add unnecessary noise by having the input higher than it needs to be to convert the signal at an optimal level. So you turn the input level down a good bit. The converter then compensates for how much you turned down by bringing up the signal by an equal amount before it outputs to the processor. The signal hitting your grid (i.e. pedals, amps, whatever) is in theory unchanged in level from what is coming out of your guitar. You've just optimized the level at which it's being A/D converted.


You "can't" really "clip" the input given that it takes drive pedals and what-have-you just as well as an amp does.

Per Cliff: "The Axe-Fx III input was designed to mimic a typical tube amp input using an average of Marshall and Fender amps for the component values."

That does indeed clear things up...the one thing that was messing me up was thinking the Layout screen meter was an input meter and the fact that it doesn't seem to jive with the input or output LED meter on the front panel...I was getting heavy red on the screen but no red on the LED's on the front panel output meter.
All in all it was hard to get to get to sound bad no matter what you do....and that is a good thing!

Again ...thanks to all that replied!
 
Let me see if I can clear this up.

The Instrument Level is to make it so the A/D converter hears the best signal it possibly can. So say you have a super low output Strat. You crank up the input level so that that Strat is hitting the converter at a level that is making sure it's well above the (very low) noise floor of the Axe-Fx. The A/D does its work and then brings down the signal it outputs to the processor by the same amount you gained up, so that what is coming in is going out, regardless. Conversely, if you have a super hot guitar, like an EBMM JP15, it's already hitting the A/D way above the noise floor, and you don't want to add unnecessary noise by having the input higher than it needs to be to convert the signal at an optimal level. So you turn the input level down a good bit. The converter then compensates for how much you turned down by bringing up the signal by an equal amount before it outputs to the processor. The signal hitting your grid (i.e. pedals, amps, whatever) is in theory unchanged in level from what is coming out of your guitar. You've just optimized the level at which it's being A/D converted.


You "can't" really "clip" the input given that it takes drive pedals and what-have-you just as well as an amp does.

Per Cliff: "The Axe-Fx III input was designed to mimic a typical tube amp input using an average of Marshall and Fender amps for the component values."


Is there a way to have presets based on what guitar you are using?

Most of my guitars have a similar output level so I don't really notice the difference. However, my father is constantly adjusting based on what one he picks up. It'd be cool to have this dial saved and recalled based on the patch or as a separate recall.
 
Is there a way to have presets based on what guitar you are using?

Most of my guitars have a similar output level so I don't really notice the difference. However, my father is constantly adjusting based on what one he picks up. It'd be cool to have this dial saved and recalled based on the patch or as a separate recall.
The Instrument Level is a global setting. Adjust it for your hottest guitar and leave it.

For presets, you can create as many of those as you need for different guitars.
 
Is there a way to have presets based on what guitar you are using?

Most of my guitars have a similar output level so I don't really notice the difference. However, my father is constantly adjusting based on what one he picks up. It'd be cool to have this dial saved and recalled based on the patch or as a separate recall.
Yup. Just make a preset for that guitar. Adjust the Amp like you would a real amp, or adjust the Input block as needed.

A preset can’t “know” what guitar is plugged in and change automatically, if that’s what you’re asking.
 
Yup. Just make a preset for that guitar. Adjust the Amp like you would a real amp, or adjust the Input block as needed.

A preset can’t “know” what guitar is plugged in and change automatically, if that’s what you’re asking.
Yeah that's what he's been doing. He's more of a tweaker than I am. Always throwing different pickups in all of his guitars.

It's probably more a wish list thing for me so I don't have to go down the rabbit hole conversation every time he tweaks something, but these conversations are always welcomed, especially over a pint .
 
Sorry to hijack this thread but I though this might be related to the OP... When recording USB Input 5 or 6 directly on the Axe III for reamping, I noticed that the Input 1 (DI) wave forms that are recorded in Studio One get chopped off on the top and bottom (kind of like compression) and levels for the DI channel would not go above -17 as is there was a hard limiter on the channel (which I checked, there was no software limiter). Is this because the input gain is too high or just the nature of my pickups? (Seymour Blackouts). I noticed that as I went down on the instrument level %, the DI levels in Studio One actually went up from -17 (as shown in pic) to about -6. Had to go to about 5 to 10% for this though and the waveforms were still being "chopped" in Studio One when I played hard.

Here's a pic of what I mean...



Top is processed signal, bottom is DI signal.

Is this common or am I missing something? I just want to make sure I'm reamping the best signal possible.
 
Sorry to hijack this thread but I though this might be related to the OP... When recording USB Input 5 or 6 directly on the Axe III for reamping, I noticed that the Input 1 (DI) wave forms that are recorded in Studio One get chopped off on the top and bottom (kind of like compression) and levels for the DI channel would not go above -17 as is there was a hard limiter on the channel (which I checked, there was no software limiter). Is this because the input gain is too high or just the nature of my pickups? (Seymour Blackouts). I noticed that as I went down on the instrument level %, the DI levels in Studio One actually went up from -17 (as shown in pic) to about -6. Had to go to about 5 to 10% for this though and the waveforms were still being "chopped" in Studio One when I played hard.

Here's a pic of what I mean...



Top is processed signal, bottom is DI signal.

Is this common or am I missing something? I just want to make sure I'm reamping the best signal possible.
Maybe I'm not understanding, but the bottom signal just looks to be much quieter... Which would make sense for the DI because it has no amplification.
 
Sorry to hijack this thread but I though this might be related to the OP... When recording USB Input 5 or 6 directly on the Axe III for reamping, I noticed that the Input 1 (DI) wave forms that are recorded in Studio One get chopped off on the top and bottom (kind of like compression) and levels for the DI channel would not go above -17 as is there was a hard limiter on the channel (which I checked, there was no software limiter). Is this because the input gain is too high or just the nature of my pickups? (Seymour Blackouts). I noticed that as I went down on the instrument level %, the DI levels in Studio One actually went up from -17 (as shown in pic) to about -6. Had to go to about 5 to 10% for this though and the waveforms were still being "chopped" in Studio One when I played hard.

Here's a pic of what I mean...



Top is processed signal, bottom is DI signal.

Is this common or am I missing something? I just want to make sure I'm reamping the best signal possible.
what is your Input 1 Level set at in Settings > I/O? is the Red Input 1 light flashing on the front panel?
 
Active pickups shouldn’t clip Axe inputs, there’s plenty of headroom for these.

Some high output passive pickups on some guitars may produce more that 16 dBu on aggressive palm mutes, but actives clip internally way earlier than that.
 
Maybe I'm not understanding, but the bottom signal just looks to be much quieter... Which would make sense for the DI because it has no amplification.

Correct... it is quieter, but my concern is that the DI signal seems to be "limited"... as in the peaks are being cut off and the max levels I'm getting from the DI signal is -17db. If you look at the waveforms, they seem to be getting limited or compressed. Once I messed around with the Instrument level and brought that down, it seemed to cut off at -5db or -6db rather than -17db, but it was still being limited. Not sure if this explains it better...
 
what is your Input 1 Level set at in Settings > I/O? is the Red Input 1 light flashing on the front panel?

I tried anywhere between 50% and 5%... as I went down, the ceiling on the DI input in my DAW seemed to increase from -17db to around -6 or -5db. I left the instrument input at around 10% where it "tickled the red" as instructed in the manual.

When I play had through, the DI signal stills seems to be limited (or compressed) in the DAW. I'm thinking it might be my hot pickups... but was wondering if I was missing something on the AXE.
 
Active pickups shouldn’t clip Axe inputs, there’s plenty of headroom for these.

Some high output passive pickups on some guitars may produce more that 16 dBu on aggressive palm mutes, but actives clip internally way earlier than that.

Could it be the the pickups are compressing the signal? Is this possible?
 
Could it be the the pickups are compressing the signal? Is this possible?

Yes, active pickups clip internally, unless you mod them somehow with 18V power or something like that.

Can you zoom in the waveform to see the actual clipping area and post it here? If it’s digital clipping, there should be a strictly horizontal line on top. Digital signal cannot go above 0 under any circumstances. If it’s internal preamp clipping, it should be slanted.
 
Here’s what active pickup clipping looks like:

184906-343059.jpg


So yes, they compress! Also, they have lower peak output than high output passives.
 
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Yes, active pickups clip internally, unless you mod them somehow with 18V power or something like that.

Can you zoom in the waveform to see the actual clipping area and post it here? If it’s digital clipping, there should be a strictly horizontal line on top. Digital signal cannot go above 0 under any circumstances. If it’s internal preamp clipping, it should be slanted.

From what I have read, 18V mod doesn't really do anything for SD Blackouts, only for EMGs. I think what is being seen is compression that is inherent in the pickup design.
 
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