Basically yes. But as the ADI is no DAC like any other you will most probably use PEQ, Loudness and BT. In that case you can only go near 0 dB volume until the leter meters show over - a few dBs lower, most probably. Activate AutoRef level  then dial Volume fully up, then a few dBs back as required and you are good to go.

The DAC gets the audio signals digitally, unaltered on all of its digital inputs and will not modify it unless you use EQing (PEQ, B/T, Loudness).

If I understood MC right then you should keep a little "headroom" (reserve) towards your amp, because you might want to make use of PEQ, Bass, Treble and if you raise frequencies the signal becomes louder.

If you were already at the "edge" shortly before your amps input gets issues ("0db/clipping"), then the use of EQ-ing on the DAC could bring the amp to clipping, if you do not lower the level a few dB upfront to reserve some headroom for EQ-ing purposes.

I propose that you use the remap key feature to be able to activate/deactivate "EQ+B/T+Ld" per remote control, you can use any of the programmable keys 1..4, see handbook 14.1.1.

For me the following settings are very useful:
Remap Keys...ON
Vol (1).......Toggle View
I/O (2)......Toggle Ph/Line
EQ (3).......EQ+B/T+Ld
Setup(4)...Mute

By this I can
- toggle the display from the distance, sometimes I prefer the "Dark Volume", sometimes the "Analyzer" view
- toggle between Ph/Line (the device does a ramp-up of signal
- enable/disable EQ-ing entirely
- mute not only via remote control (there is a mute button) but also in front of the device, like I am used to, as this is also standard setting on my Pro in the recording corner

Yes Bernhard,

but still you need
a) to find the proper output level not to overload the inputs of the device behind
    (This of course depends mainly on the capabilities of the Input of the device behind the RME ADI-2 DAC
    for what input levels the circuits have been designed)
b) to choose the best reference level on the RME DAC for the output level that you want to achieve (*)

(*)Either you set the reference level on the RME DAC manually or - what I would recommend you -
to choose the setting - Auto Ref Level    ON -
Then you only need to turn the Volume knob on the RME DAC. With increasing output level
the RME DAC automatically chooses the optimum Reference Level for the converters for optimum D/A conversion results.
In the Display of the RME DAC you see then the Output Level and on the left side in smaller letters the Reference Level
which has been chosen by the DAC. You also hear relays clicking when the reference level is being changed.
The 4 different output levels at 0dB according to handbook chapter 30.2:
+19 dBu, +13 dBu, +7 dBu, +1 dBu @ 0 dBFS

What bad experience ? I am using it all the time here and never had any issue.

And most important, Auto Ref level needs to be enabled if you use the dynamic loudness. Without dynamic loudness I personally can't live anymore, its so perfect ...

And if you really need to ensure to have the unmodified signal without any EQ, then do as I suggested to use the remap-key function to disable all with one push on the knob.

In the display of the ADI you get then also the visual feedback, what function has been issued and whether something has been turned on or off.

I have 2 applications
1. cross check a mix with phones
2. enjoying music

If you are doing mixing and mastering you hear more in the range of reference volumes.

If I only listen to music to enjoy it can be that I want to listen more silent than that.

Therefore I tweaked dynamic loudness for me, that it works for both
to 1. dynamic loudness kicks in when the listening level is below what I use for mixing mastering cross checks
to 2. if I turn down the music the sound doesn't become "weak" as bass/treble becomes dynamically increased.
By this you can decrease the volume by -20dB without getting the impression that the sound becomes thin.

Example: serious listening level for my LCD-X is -35 dB.
But careful, I normalized my audio files with Foobar2000 (because there you can scan with higher resolution)
and wrote replay gain tags, and in my main player MusicBee I am using then intelligent volume adjustment (or similar,
I just translated this on the fly from german).
So on average the songs are more quiet compared to their original volume by the replay gain tags (that is simply metadata information in the FLAC header, the sound file doesn't become altered / ruined in any way.
I say this explicitely, if you have i.e. the same phones, that you do not ruin your ears by using -35dB
You might have louder results compared to my volume, as I normalized the volume of my tracks.

So -35dB with LCD-X and normalized songs is still a level which allows me to listen very comfortably for 30-60min
with lots of headroom to turn it louder, if I want.

Dynamic Loudness I want to start at -35 dB and lower levels.
From handbook we know that this is a fixed range of 20dB where the Bass/Treble gain is being adjusted up to the configured maximum (default +7dB for Bass and Treble and configurable separately).

I set "Low Vol Ref" for Phones 3/4 to -55dB, this is the value where the maximum Loudness adjustment will be reached.
So .. starting with -35dB and lower levels Loudness will be increased up to +7dB Gain Boost for Bass and Treble.
This maximum will be reached at -55dB which I configured as "Low Vol Ref".

If I want to ensure, that I do not use EQing in any way (Loudness, B/T, PEQ) then I use the re-map key feature.
The I/O key is remapped to enable/disable this for Main Out 1/2
The EQ key is remapped to enable/disable this for Phones 3/4
With the vol key I can switch between Main Out 1/2 and Phones 3/4
SETUP is remapped to Mute all ...

Another trick, as the normal listening volume is perhaps a little bit too loud for restoring a stored config profile
I set the volume -5dB so that a little bit loudness is already in use.
-5dB therefore, because from that you can easily deduce, at which output level the loudness is inactive.
Example: I store
-35 for Main out 1/2 and
-40dB for Phones 3/4
because the output level, where loudness is not yet in effect is
-30dB for Main out 1/2 (config: Low Vol Ref at -50dB)
-35dB for Phones 3/4 (config: Low Vol Ref at -55dB)

Hi jiw, thanks for your comment, yes it could be the case that MC referred to internal processing.

But somehow I had in memory that we had a similar thread where the focus was more on the analog input of another DAC cascaded behind a DAC or Pro, where distortion could also happen. Therefore I thought he refers to the same or something similar in this case. I think in normal cases the analog inputs sections (at least of RME gear) can accept a few dB more anyway by design. Maybe the same for other equipment, not so sure 'bout this.
In that regards it appears indeed to be more likely, that MC refered to internal signal processing than to overload on the next devices analog input.

If you don't want to use any of the signal processing, then set the DAC to +24dBu and max out the volume. Then you use it like any other DAC straight into your HPA4.

Hi jiw,

now as MC clarified .. what was wrong with my statement ? At the end I wrote the same, to reduce output level a few dB to create headroom for the use of PEQ, Ld and B/T. Only would like to get the point what was wrong with my statement from technical perspective, if there was something wrong.

Hi ramses,

you wrote about keeping the signal within the input capabilities of the downstream device (the amp). I have emphasised the relevant parts with italic (as you already used bold).

I wrote that I think MC was writing that the signal should be kept within the output capabilities of the ADI.

Also, I don't think what MC wrote in his second answer (Nr. 15) refers to anything you and I have been discussing (emphasis mine).

I think this is a poor example. 0 dBFS is a particular amplitude of a signal in a device (the amplitude corresponding to the threshold of clipping). If the signal reaches 0 dBFS after B/T boosting, the ADI will not overload (assuming no sufficiently high intersample peaks are present in the signal). Even if the signal reaches 0 dBFS before B/T boosting, this might not overload the ADI as the signal might be outside the bandwidth (reach) of the B/T filters.

If you meant attenuation through the ADI's volume control, you should use dB (or dBr using AutoRef) because you are describing a ratio relative to ADI's reference output level (where 0 dB attenuation corresponds to reference level for a signal with level of 0 dBFS).

Sure it does. The DAC chip gets fed too high levels then, which is obvious as the output level meters also show a too high level. Not sure what music you listen too, but on the most popular ones (not classical)  B/T with +6 dB will definitely clip the DAC if volume is not reduced to below 0 dB.

[Edit]: wrong quote fixed.

Are we talking about sines or music? Which - as so often these days - hits 0 dBFS all the time? I run into this effect regularly, Vol at 0 dB and getting overs, just to find that BT or Loudness were active.

The processing has 24 dB headroom, but that headroom is of no use directly in front of the DAC chip. There the level must fit.

Or did I misunderstand you?