Have you tried to sample from your phone? Just to check if it is your laptop or the OT.
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Still very low…getting around it by going through my Komplete Audio 6 . Seems to fix it. Thanks!
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I can’t believe GAIN is digital. Digitally controlled maybe, but for me it has analog consequence, before analog / digital converter.
If it was digital, recording noise / signal ratio would be the same, whatever GAIN settings are. It is not the case. If you record with low gain settings, noise / signal ratio is higher, from what I just tested.
GAIN = -64 > no sound
GAIN = -63, NORMALIZED ! BEWARE !
GAIN = +63
Maybe someone can clarify that.
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That’s truncation and dither, the OT uses fixed point math so if you lower the gain to -63 and then normalize it you’re essentially reducing the bitrate. That clip sounds just like like 1 bit or 2 bit audio poking out of the dither to me. Definitely doesn’t sound like analog line noise.
But it does answer the “does the OT dither” question.
Compare it with this.
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So, to which level should i normalize my samples? Is there any recommendation to have a standarized sample loudness (individual drum hits mostly)? Is normalization a standarized loudness anyway?
What would be best for mixing purposes=? I.e. have enough headroom left for later treatment?
I did double check my libary - it depends on the Vendor - the Zennhisers are down to -6 db, and for example Mike Vale Tech House is down to -10 db. It kind a sucks to constantly shift loudness levels in octatrack - it isnt such a problem in abelton, because i can more easy see what happens - in OT Live session context its more disrupting to the guys i make music with.
That’s why I’m so for this tiny tweak - at least it would get done in one action:
If I’m feeling fastidious:
•normalise
then
• -3
then
• -3
but mostly I don’t want the hassle and just let it get messy with all sorts of compensating here and there
But I’m not working live or with other musicians…
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Its not that simple I’m afraid as absolute peak levels (dBFS) are not the same as loudness. Some samps might be peaking @ -10dBFS but might still be so squashed by compression/limiting/clipping/etc that their RMS is about the same as other samps peaking @ -3dBFS
For evaluating perceived loudness of sounds, you need to use some other metering system than peak level, RMS or LUFS or something like that… Then there’s also Bob Katz’s K-system…
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I have Fab Filters Limiter L2, if i would run all my samples through that limiter, and set them to -6 dBFS, would that even out ? I know that frequency influences the listener. I.e. high frequency noises are perceived louder than lower frequencys.
I mostly want to even out one shot drum hits, which tend to have a high portion of white noise anyways.
Hmm, I am not aure if just slapping a limiter on would do it either. If you have a batch processing system that you can try with, I guess you can give it a shot…
But the way I have been taught audio engineering is, ”you need to evaluate loudness by ear” (although to be fair, this was before LUFS metering became a standard etc). Would be great if there was some method for batch processing audio to the exact same loundess…
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I have sometimes put 64 one shots as a chain onto a flex machine (chain made using Octachainer). Then using the editor you can highlight slices and adjust gain by ear, monitoring on speakers. Then save the adjusted file
When I’ve done this it’s been with a selection of bought in samples which already sound good. I wouldn’t want to change their internal dynamics,/envelope so I’m just using gain adjustments to match their subjective ‘force’ by ear.
You can use the crossfader to jump around the chain for comparison, so not just comparing each hit with its neighbour
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Pro tips!
If one seeks a simple solution i can recommend wavosaur:
Its very lightweight, and can do vst batch processing.
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(Wavosaur can also be used for transient detection. Export samples > Octachainer)
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Since I had some things that drove me crazy and I did not find a final solution or answer to them here in any thread I also sat down with some sine waves and did some testing.
First of all, I have an issue which seems inconsistent with what other people are experiencing: any audio that goes through a THRU track will be 12db lower than audio that goes through a FLEX track, though that is likely due to the +12db boost that samples get by default, and that the THRU track does not make up for the -12db pad applied to the inputs.
My aim was to understand how to get somewhat consistent levels among THRU and FLEX machines and with direct and imported samples.
I did use a cheap Presonus Firebox interface and Reaper, sending a pure sinwave from the DAW out into the OT and then back into it.
So far it seems to me that anything that goes out from Reaper with more than 8dbFS distorts the inputs. But I do not have an absolute value of how much Vpp that is, since I have no way to measure it. The actual level going out from the analogue outputs varies from interface to interface I guess.
Sending a -12dbFS sine from Reaper, sampling that in a recorder and then importing the sampled file into Reaper again was showing a 4db level, which means that the 8dbFS totally make sense, since that would equate to 0dBFS when I import back in (but I am just speculating).
What is totally consistent with everybody else’s findings is that red does not mean overdriven. I was way into red with -12dbFS and it recorded a clean sine.
When importing samples directly into the OT I noticed that they play lower. If I import a 0dbFS sine I have to add 8db in the sample settings to make it play at the same level as the 12dbFS sine coming from the DAW and being sampled into a FLEX via the AB inputs. Which is kind of weird.
My conclusions are more or less these:
- I don’t see why I should not keep everything at their defaults. It makes it less prone to doing something wrong.
- use the GAIN and VOL controls mostly for attenuating, not boosting, except for the following thing:
- crank AMP VOL up to +63 for THRU machines to have unity gain with the FLEX and PU machines
- crank master volume in the mixer page up if the overall level from the OT is too low.
- send audio at -12dbFS or less (from the DAW) to the OT, or at a comparable level from other piece of kit, so there’s a bit of headroom.
- no problem normalizing samples at 0db, but anything between 6db and 0db will work
Let me know if I’m making any big mistakes in my thinking here!
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Not a mistake but I’d set THRU Playback page VOL to +63 instead.
VOL +63 seems to be equivalent to +12 db.
I made some tests here :
With a 0db sample, 0db Attributes, Level 127, Amp Vol = +63, Main Level = 0, you should be at 0db. Correct me if I’m wrong!
I prefer Main Level set to 0 because it changes Main recording level.
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Tried that, but it distorts the audio, while cranking up VOL on the AMP page doesn’t. I agree that it would make more sense to use the one on the playback page though.
I need to try that, but won’t VOL=+63 add +12db?
When I raise the volume of the sample in the sample editor by +8db I get that is plays at the same level as the -12dbFS sine wave coming from the DAW.
It has to be noted that I have no idea what the inputs on my audio interface do to the audio, so it’s hard to talk in absolute terms here.
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Surprising. I have to check.
Yes, but apparently default 0 value is equivalent to - 12db.
Something to consider is that every audio interface has a different rated output level. 0dbFS on the DAW will output the max level for the specific interface being used, and usually there’s a toggle to set for +4dBu or -10dBV reference which will each yield a different output voltage…
On my apogee 0dbFS at +4dBu will output +20dBu, OTmkI’s inputs are rated at +8dBu so will definitely clip. At -10dBV the apogee will output +6dBV which is very close to +8dBu (http://www.sengpielaudio.com/calculator-db-volt.htm) so much more OT friendly…
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Yes exactly. So it’s a bit hard to do scientific tests without first making some measurements… but it can still be some sort of reference I guess.
One thing I didn’t think of, but which I will test, to complete the picture is recording the output from the audio interface back into the interface itself, without nothing inbetweeen, so I can see if there’s some attenuation going on.
At the end of the day I don’t even think that we need to get all scientific, I just need to know two things: what do I do to make levels consistent and how to I prevent overdriving the inputs/outputs.
From there it’s just a matter of adjusting the workflow.
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Yeah, I guess I just said that to clarify that the results you get will be right for you, but someone using a different interface will get different results as they don’t all have the same output voltage for a given dbFS reading on a DAW…
The inputs of an audio interface would follow the same concept. Whatever max level the inputs are rated for will be what shows 0dbFS in the DAW, and again there will be two different settings… On my apogee they are rated the same, I imagine most are but not really sure…
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