Gain aGain

[papernoise]

Not a mistake but I’d set THRU Playback page VOL to +63 instead.

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.

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 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.

[sezare56]

Tried that, but it distorts the audio, while cranking up VOL on the AMP page doesn’t. I

Surprising. I have to check.

I need to try that, but won’t VOL=+63 add +12db?

Yes, but apparently default 0 value is equivalent to - 12db.

[Open_Mike]

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…

[papernoise]

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…

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.

[Open_Mike]

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…

[papernoise]

Yes absolutely correct, and thanks for mentioning it!

Btw. I have done another test, just to be sure. I patched the outputs of my interface into its own inputs and checked if there was some significant level differences going on. As far as I can see there is just a 0.4dB difference.
In the light of this:
Flex and Thru machines behave as expected, if I send out a sine @ -12dbFS and sample it back in and play the result back I get the same volume. which means that the incoming level is maintained. If I do the same with Thru I get a signal that is 12db lower, which confirms that there is a -12db attenuation and tells us that the Thru machines do nothing to make up for it by default.
So far so good.
Now considering that:

• the output level of my audio interface (and please note that as @Open_Mike as stated before, this value changes from interface to interface) hits the overdrive mark in the OT at -8dbFS (in the DAW) and
• that when I import a sinewave at 0dbFS into the OT directly and play that back without any changes of level it will measure 20dBFS in the DAW. 20 - 12 = 8, so you have those 8dbFS again.

So let’s just say that my audio interface has an “offset” of 8dbFS compated to the OT. If we remove those 8db we get how the OT bahaves independently from my audio interface.
For the sake of this explanation let’s just call the -8dbFS in my DAW 0dbFS. so 0bd in the DAW == 0db in the OT

Now, there’s one oddity that emerges if I do the math:

• If I send out a file @ 0dbFS from the audio interface and sample it I end up with a @ 0db on the OT’s CF card not -12db.
• When I import a 0dbFS sample and play that back I get a level of -12dB, which is 12db lower than what I was expecting.

So I deduct that the OT will apply the +12db boost that is automatically set in the sample settings for the recording buffers the moment you hit “save to new sample”. I need to see if the same happens if the setting is at 0db.

[Kayi]

My assumption leans to the idea of attenuation. My MK2 has awful signal pathing. When i sample, i need to have ALL levels at maximum just to gather something with a reasonable volume, and then i still have to apply maximum gain while resampling in order to make a usable sample. Did i throw my scratch at a lemon? Maybe start a support ticket?

[Kayi]

“Wuld”

[Supercolor_T-120]

It was a while ago, but IIRC what I did to help mitigate the interface itself as a variable was do a direct loopback like described above, but on the return channel I used a gain plugin to match the level of the returned signal to the level the tone generator was nominally putting out. Any analysis plugins went after the level compensation. That way any changes in level that were being introduced by my interface and cables were more or less out of the picture and for the purposes of relative level measurement with that specific test signal I could act like the interface didn’t exist - any change between the signal being output and the signal being measured at the return would be happening in the Octatrack. Still can’t make any meaningful observations about absolute values this way of course, but in terms of RELATIVE values - how the Octatrack CHANGES the level of the test signal passing through it - it seems like a pretty good way to make some meaningful observations.

[Open_Mike]

Never had such issues, MKI…
Maybe this post could help:

Gain aGain

I’ve been preaching about pushing hotter signals into the OT for awhile now as I feel I’m in a minority about claiming to get a good mix, my meters may flash red here and there and I never hear clipping, but I tend to keep it just below that… I never actually use the gain parameter in the mixer. I always have the Rytm piped into AB and run it direct. The Rytms output is hot enough that I’ve never messed with OT’s gain, I just use the Rytms volume knob, which I set and forget… Mackie vlz4 with …

[sezare56]

OT lessons?-A safe space for dumb questions with great help. Hashtag how do you

Most codec have an analog amp prior to the ADC stage, I would assume this is what the mixer gain is setting. ( I assume the same is also true on the output side)

Yeah. Digitally controlled, but analog. I assume this too!

Got an answer from the support.
On the mixer page, “gain” “cue” and “main” are digital volume controls.

[oktatracker]

So, for people with short attenionspan(/noobs):
What are the key take aways?

My samples are definitely of very low volume when recording in the OT2.
And in all cases, recording in OT2 or dumping samples, I have way too much doubt, as to which setting/knob I should use, for controlling the volume of the samples being played.

Any final bulletpoints on this topic- or is it still work in progress?

Thanks for your effort

[sezare56]

My samples are definitely of very low volume when recording in the OT2.

Hot enough signal? With some signals it’s better to hit the red led. Increase mixer GAIN.

How is the recording waveform?
Picture?

[oktatracker]

Thank you, I know about the gain option in the mixer.

I am actually more interested in understanding the theory of your above discussion, than understanding my particular use case. There are a lot of word, terms and numbers I don’t understand, after reading through it all. So I was just wondering, if there was going to be a easy-to-understand conclusion of all your testings, perhaps in bullets

[sezare56]

Can’t be easy with all OT gain stages!

GAIN 0 (Mixer)
VOL +63 (Thru only)
GAIN 0 (Pickup only)
VOL 0 (Amp)
FX1 (beware*)
FX2 (beware*)
LEVEL 127 (Default is 108, around -3db)
MAIN LEVEL 0 (impact on SRC3 MAIN recording)
*FX can increase levels drastically, especially the Compressor! 31 GAIN is around 24 db, I didn’t go further for test. It can be interesting for very low signals, as a guitar or dynamic mike straight in OT, with all other parameters maxed!

The values are a reference for unity (what comes in = same ouput level). I liked @philtipping approach / diagram, ( Signal flow/gain staging diagram for Octatrack MKII).
I tried to make a simpler version (at least simpler for me), not sure it suits everyone!

Octatrack-MKII-Gain-Staging_will5.pdf (170.4 KB)

[Supercolor_T-120]

I found that if I turned the main level to +12 and left everything else at the default value, got unity gain across the Octatrack with both thru machines and track recordersm, so I use that as my starting point for all projects now.

[sezare56]

I found that if I turned the main level to +12 and

Make sense, MAIN +12 is arround +3db, corresponding the -3db lost with track Level 108.
Caveats : increasing MAIN increases SRC3 MAIN recording. That’s why I prefer to keep MAIN at 0.

[Supercolor_T-120]

Good point! When I use src3 it’s almost always to record from CUE so I never even thought about that.

[sezare56]

I never even thought about that.

Me neither untill I wanted to play a MAIN recording with MAIN LEVEL settings above 0!
(Played too loud after)

[XC3N]

Sorry to keep rezzing this thread but a lot of unsolved mystery left

I have upgraded to MK2 and sold my MK1 so I can’t compare back, but it seems that +12db gain by default on flex recorders is just messing everything up. everything seems at unity with that +12db turned off using default settings.

So of course I could just set gain to all buffers to 0db, that fixes the issues… but WHYYYYYY hahaha why is it the default?! Did they change something from MK1 to MK2 but forgot to adjust the default? It’s really been bugging me. Oh well, short of an answer I will just manually adjust gain on recordersx8 manually on every new project x_X