Guide: Re-Amping with external sound card/audio interface

I still don't understand your issue with -12dB?
You add 3 tracks to this also peaking at -12 you hit 0 dB. even if it was harmonically a wasteland say like a kazoo, 808 or sine wave at the 10th track peaking at -12 you'd be over digital 0 in your master buss.

For the life of it I will never understand why anyone prints their tracks too hot, not like noise floor as any consideration.
And pulling the master fader is a crap recipe for bad gain staging.
Most plugs run happiest at -18 to -12.

Kinda like burning a steak and then tossing it on a bag of ice to cool it off. If you under cook it you can always slap it back o the grill f you over cook it it's done with,
 
  • Like
Reactions: Rex
What Ed said. Faders set to zero, average levels running a little under -12 dBfs...isn't that pretty much exactly the sweet spot?
 
Ed DeGenaro said:
I still don't understand your issue with -12dB?
You add 3 tracks to this also peaking at -12 you hit 0 dB. even if it was harmonically a wasteland say like a kazoo, 808 or sine wave at the 10th track peaking at -12 you'd be over digital 0 in your master buss.

For the life of it I will never understand why anyone prints their tracks too hot, not like noise floor as any consideration.
And pulling the master fader is a crap recipe for bad gain staging.
Most plugs run happiest at -18 to -12.

Kinda like burning a steak and then tossing it on a bag of ice to cool it off. If you under cook it you can always slap it back o the grill f you over cook it it's done with,
Click to expand...
Looks like I touched a nerve. I don't think your steak analogy is quite appropriate here. A healthy signal at -12 to -6dB is not "over cooking." To me that would be 0dB or above. Humoring your analogy: what is "slapping it back on the grill?" Boosting the signal with gain, EQ, compression, etc.? That adds more noise than a stronger gain stage up front. Having to redo a take with a stronger signal? That's not very efficient.

It seems you are forecasting issues based on assumption. Master fader attenuation?

What would be the problem with pulling a channel strip's fader down? A robust signal attenuated will sound better than weak signal that's been boosted.
 
Ed's point was that, when your average signal gets much above -12, you risk clipping your peaks—especially when you start mixing multiple signals together. And you can't undo clipping. There's no way to "slap it back on the grill" and remove the clipping.
 
CactusTone said:
Meant to say +4dB or Bypass in my previous post! So, I had it at +4dB with the preamp gain all the way down.
Click to expand...

It looks like that's a setting for mic/line combo inputs, where bypassing the preamp results in a fixed +4 dB reference. Did you try a 1/4" line-only input, switching the reference level to -10 dB?
 
CactusTone said:
Looks like I touched a nerve. I don't think your steak analogy is quite appropriate here. A healthy signal at -12 to -6dB is not "over cooking." To me that would be 0dB or above. Humoring your analogy: what is "slapping it back on the grill?" Boosting the signal with gain, EQ, compression, etc.? That adds more noise than a stronger gain stage up front. Having to redo a take with a stronger signal? That's not very efficient.

It seems you are forecasting issues based on assumption. Master fader attenuation?

What would be the problem with pulling a channel strip's fader down? A robust signal attenuated will sound better than weak signal that's been boosted.
Click to expand...
You haven't touched a nerve but usually I find guys with strong convictions gathered by YouTube.
What is the issue at pulling fader down? Look at fader resolution? There's a reason why the Majority of people I know mix with k scale.
Assume 20 tracks at -12 and your going to be over by about +18. it's an unreasonably hot signal. See how the Axe inputs go into Red at -6 why do you think this is.
 
Rex said:
Ed's point was that, when your average signal gets much above -12, you risk clipping your peaks—especially when you start mixing multiple signals together. And you can't undo clipping. There's no way to "slap it back on the grill" and remove the clipping.
Click to expand...
Even if it only peaks at -12 once you have four tracks at that it'll be over.
I've gotten so many projects to mix that were tracked too hot so I started trimming down to -18dB in my templates.
 
Ed DeGenaro said:
You haven't touched a nerve but usually I find guys with strong convictions gathered by YouTube.
What is the issue at pulling fader down? Look at fader resolution? There's a reason why the Majority of people I know mix with k scale.
Assume 20 tracks at -12 and your going to be over by about +18. it's an unreasonably hot signal. See how the Axe inputs go into Red at -6 why do you think this is.
Click to expand...
Don't know what your referring to regarding YouTube. I posted my video there to provide a visual of my front panel settings and meter readouts. I'll have to study up on fader resolution and how it effects a signal. New term to me. Resources you can point me towards? The Fractal manual suggests tickling the red. Do you avoid this? To confirm, you recommend multi track settings at -18dB? Thanks!
 
Just look at the faders, it's a long scale so the lower you the actually physical space is smaller for moves.

I strongly suggest peaking -12dB in the DAW.
RMS at -15 to -18.
Since in digital you don't have the ability to go "Over" this will get you to analog 0 dBVU. Meaning just like in analog you now can go 20 db into Red. I'm guessing that is why Slate plugs one running there.

As for tickling the red in the Axe, I don't I'm just below but that's a matter of taste. Red in the axe is -6dB. We're talking what prints on DAW though.

Resources? Read Bob Katz site http://www.digido.com/
 
CactusTone said:
The Fractal manual suggests tickling the red. Do you avoid this?
Click to expand...
The manual suggests tickling the red at the Axe's input, not its output. If you do that at the output, you're in danger of clipping the Axe's output.
 
Last edited:
i just got some coax->lightpipe converters from monoprice the other day and got this to work and it's great! an added bonus that i don't know if possible with regular USB reamping is being able to reamp two separate tracks simultaneously, which is great! the only thing i noticed though is there seem to be intermittent dropouts for like the first 30 seconds to a minute or so and then after that everything is fine, is that a normal thing? i'm not sure what i should change where to help that out.
 
Hey guys, trying this out based on Sasha's instructions. When I go into the I/O menu of the Axe FX XL plus FW6.03, into the Audio tab and change the Word Clock from Auto to SPDIF/AES In I get a message that reads "No Input Clock". Any ideas? I'm using the M-Audio Profire 2626 audio interface.
Attached is the screenshot of the settings on my M-Audio.
 

Attachments

  • Screenshot.jpg
    Screenshot.jpg
    37.4 KB · Views: 12
Errr difficult to say because interfaces are very different but for a start you want 48khz 24 bit and also external sync?

On my particular interface I have to make sure Cubase is outputting to my interfaces "DAW MIX" output.
 
Hey guys, I've had 2 highly respected mixing / sound engineers tell me that it's best to record in 44.1. I suppose one of the only reasons to record in 48 is if your re-amping the Axe digitally that's locked in at 48?

What disadvantages are there using a re-amp box like the Radial Re-amp JCR-1 so you could have the project at 44.1 vs the digital method that Sasha describes?
 
derrylgabel said:
Hey guys, I've had 2 highly respected mixing / sound engineers tell me that it's best to record in 44.1. I suppose one of the only reasons to record in 48 is if your re-amping the Axe digitally that's locked in at 48?

What disadvantages are there using a re-amp box like the Radial Re-amp JCR-1 so you could have the project at 44.1 vs the digital method that Sasha describes?
Click to expand...
And their reasoning flawed as it may be was?
I used to do everything at 44.1 when CDs actually still were a used medium for music delivery but subsequently worked on enough sessions that were for Film and that all was 48 k or HD and I haven't gone back to 44.1 since.

Used every way to re-amp over the years from flipped DI with pedals, to radial xamp, jdv, jdx,jd7, re-amp, eleven rack.
And the one thing that even when I use miked Amps comes in handy is that with the AFII RE-AMPING in digital is less prone to ground loops and gremlins then re-amp boxes.
 
  • Like
Reactions: Rex
The closer you get to a 40 KHz sample rate, the steeper the anti-aliasing filters have to be, and steep filters can have undesirable artifacts. I think @FractalAudio said that the theoretically-ideal sampling rate would be around 60 KHz. If you go any higher, you just add noise and CPU overhead, without introducing any audible improvements.
 
Ed DeGenaro said:
And their reasoning flawed as it may be was?
I used to do everything at 44.1 when CDs actually still were a used medium for music delivery but subsequently worked on enough sessions that were for Film and that all was 48 k or HD and I haven't gone back to 44.1 since.

Used every way to re-amp over the years from flipped DI with pedals, to radial xamp, jdv, jdx,jd7, re-amp, eleven rack.
And the one thing that even when I use miked Amps comes in handy is that with the AFII RE-AMPING in digital is less prone to ground loops and gremlins then re-amp boxes.
Click to expand...
Thanks Ed
 
You must log in or register to reply here.
Back
Top Bottom