Question about recording level

horror

horror

Inspired
Hi all, I'm currently running output 2 (set to copy output 1) at +4dbu into the line level inputs on my Roland Studio Capture (also set to +4dbu). I'm recording into Logic Pro. My I/O levels in the Axe are all balanced per a write-up that Yek had done (very helpful). No other hardware involved.

When I record, in order to get a good signal level (peaking between -18 and -6 on the Roland and Logic level meters), I need to crank the Out 2 knob on front of the Axe all the way to max. I figured this is fine until reading some information about unity gain and this setup only being useful for 4CM.

Since 4CM is not a factor for me, I'm wondering if I'm doing something wrong here?
 
Since you are in the digital realm I don't think it matters where you crank up volume as long as you don't clip (anywhere). FWIW I keep my front output at noon and use the volume in the amp block to attain my desired record level. I don't save the preset, so that it returns to where it was, but rather make a note in the Logic track as to the preset and amp volume and any other particulars in case it needs a revisit. Just an idea to try. It's hard to get a cookie cutter process when everyone uses different gear and methods.
 
Depending on the level coming out of the grid, you may need to adjust the output level, so, no, there’s nothing necessarily wrong there, but check the grid output level just to make sure it’s not extremely low. In any case, you might want to try to record using digital output instead of analog . That removes some of the worries about levels and improves audio fidelity.
 
I also use a studio capture - I use the line inputs on the back set to +4dBu (just like you it seems)

If I run my presets such that the peak levels are dancing around the zero line in the output block meter (or in the preset levelling tool), it will be close to peaking, if I max out Out 1 on my AxeIII. So, I have reduced Out 1 to around 50%, and now my peaks are around -18- -12 dBFS, when I record.
 
Hey all - thanks so much for the replies. Based on Smittefar's response and similar setup I'm thinking that GlennO may be on to something (grid output level might be low). I'm also interested in playing around with your idea, mwd. Easy enough to tweak the amp volume and let it roll back by not saving the preset.
Some ideas for me to try now. Thanks again. Very helpful to hear what others are doing.
 
@horror if you open up the Grid Layout view on Axe FX III (the Layout button), then hit Zoom to zoom out, you'll see your global output level for your preset. I usually aim for right around 0 dB (flickering into the red, but hovering around 0 otherwise). If you're significantly lower than that it could explain why your output level is low to your interface.
 
Thanks IronSean - this is a great tip as I am so often working just in AxeEdit that I forget about the screens on the Axe itself
 
The output blocks in AxeEdit will show you the same output meter. Like IronSean said, aim for that 0dB line on the meter. There's also the Preset Leveling tool in AxeEdit that shows the same meter and give you direct access to the amp block output levels for tweaking.
 
Got it, thanks. Thought maybe there was some additional info available on the axe screen that I wasn't seeing in AxeEdit.

With that said, my output level appears to be healthy and hovering around 0 dB but for whatever reason, the Output 2 knob at noon sends a lower than optimal recording level (around -30 dBFS).

My initial worry was that the recorded tracks with Out 2 at max would sound drastically different than what I was comfortably monitoring and tweaking via Out 1 to speakers (with knob at 10-11).

I should have just listened to the recorded tracks in the first place. They sound fine. Nothing's clipping in the path and thinking of mwd's point above, I'll keep the Out 2 knob dimed and forget about it.
 
That doesn’t sound right, but it’s not clear what role the audio interface is playing here. One way to avoid mysteries like this is to connect your AxeFX directly to your computer.
 
I hear you and love the thought of keeping things in the digital domain. However, I passed over this initially because I was collaborating with others in a 44.1 project. To avoid any sample rate conversion, I set up for analog out to interface.
On a completely unrelated note, I was looking at your signature and saw the reference to Spectrasonics. Listened to some clips - great stuff! I have had Omnisphere on my wish list for so long but yet to pull the trigger.
 
Thanks.

FYI, you are doing sample rate conversion. There's no way around that. Connecting directly to your computer will do automatic sample rate conversion to 44.1, but an analog connection to an audio interface will also do sample rate conversion. The disadvantage to the audio interface is the extra D/A/D conversions. That and the complexity you've run into the volume levels. So I think you're making things more difficult for yourself for no good reason :).
 
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