I assume the OT’s normalize is just doing what any normalize function does, that is, make the signal as loud as it can be without changing the dynamics.
For me, unless I learn something I am doing wrong, the easiest and most consistent thing to do is:
We need some type of GainAgainAgainInternal thread to determine how much precisely sample gain should be dropped after normalization to allow reasonable headroom in the AMP.
Conclusions like this would be good. Just an example:
That is just an example of the conclusion I am after. I have not tested those settings yet.
Just occurred to me that if the OT normalizes the samples to 0dBFS, you could experience intersample peaks and therefore clip the master out with certain program material…? Maybe this is what’s happening, hmm
Intersample clipping does not show on the meters as it actually clips at the DAC
If that is indeed the case then some headroom must be left at the track mix level
This is all about clipping individual track AMPs. Not the master.
okay then, so that rules out intersample peaks. We back at the internal summing engine, its headroom and all that.
Just made another test with a new project, a Flex playing a 0db normalized sine wave.
VOL=+63
LEVEL=127
No clipping
Main Level 0 to 2> clipping
Fx 1 and 2 set to None :
Main Level 0 to 1> clipping
Thanks so much for doing that. I need to drill into my case some more. I will report back.
Would you mind posting that sine wave file?
What if a normalized sinewave will not clip but some other sample will? As in, the sinewave is a constant amplitude signal with no dynamics, right? What about something like a really fast transient?
Dead laptop screen unfortunately. Smartphone only for now.
It is quite easy to generate with a DAW.
I set it at 256hz, 0db, looping perfectly at 120 bpm, so it can be used to check clicks and clipping.
0db is 0db, clipping is clipping, whatever the signal is IMHO. I think it’s harder to detect fast transients clipping by ear. Totally obvious with sine waves.
1000 Hz sine is a professional reference for tests, but so boring to hear!
Now I am doing more tests, and I might have had a temporary lapse of insanity about all this. Sigh.
I will report back when I know more.
From test below :
VOL = +63 > +12db
VOL = -32 > -12db
New project
Flex playing at 0db sine wave
Fx 1 and 2 set to none
VOL = +63
No clipping
VOL = +63
Attributes Gain = +0.5db > clipping
VOL=0
Attributes = +12db > clipping
VOL=-32
Attributes = +24db > clipping
Sample gain staging order seems to be :
Attributes Gain > VOL > Fx1 > Fx2 >Track Level > Main Level
From other similar tests with Attributes Gain :
Main Level 0 to +63 > +12 db
Track Level 127 to 108 > - 3 db (108 is default)
Track Level 127 to 64 > - 12 db
VOL = +63 > +12db
VOL = -32 > -12db
I’d set MAIN / CUE LEVEL to 0 because it increases SRC3 MAIN / CUE recording Level.
I almost hate to bump this thread, but I feel like I should to:
I know a lot more about gainstaging the OT from all this.
Thanks again fam!
Thanks for asking! 
Always something to learn, trying to answer.
This thread ended up sending me down a rabbit hole of intersample peak reading and I learned a lot more about it than I even realized there was left to learn, so if anything I should be thanking you!
Among other things, I learned that 8ntersample peaks actually CAN cause problems in the digital domain, not just during DA conversion, although from what I could tell they’re mostly related to metering and resampling and wouldn’t have been relevant here anyhow.
Haha thanks for saying that. It makes me feel better lol.
Yeah, if you can master gainstaging and levels in the OT, than everything else is easy, especially DAWs.
Most importantly, at least for me, I now understand a lot about the importance of gain staging in floating point DAWs. Even when you can’t clip your buses in DAWs, working with compressors and other dynamic effects requires care. The skills gained from wrangling the OT translate nicely in that regard.
Have anyone experienced clipping to the point of audio-drop out? ran some experiments today since i had some problems that i submitted earlier.
Normalized sine wave at AMP 0, no FX. If the VOL goes above 0 i hear digital artifacts, distortion, and then complete silence. What is going on here?
Well if its normalised you’re at 0db already right? Then increasing VOL will internally clip, this means distortion, which basically means squaring off the sine wave, ( you can check that with resampling) i guess you would reach a point where the pulse width hits zero = no audible sound. Thats what I reckon, others might have more knowledge?
You are correct. I was using incoming noise and comb filter to create sounds, and the sampled waveform showed a square wave close to zero pulse width.