Babaluma, after you posted that you have "Line Out set to +1dBu" I looked to see what value I have and I was surprised to see what the default value is: -5 dBu and in the manual it says "Choices are -5 dBu, +1 dBu, +7 dBu, +13 dBu at the RCA output, referenced to digital full scale level (0 dBFS). The levels at the XLR output are 6dB higher, +1 dBu, +7 dBu , +13 dBu, +19 dBu." and I'm on xlr, obviously.
So I have to manually set the +1dBu value you said. I don't know why I was convinced that the default value is the one for xlr.
Please tell me why you chose the Pro version of this model? Are you recording or is it something else? Sound quality in general, headphone outputs or what is it about?

Didn't you see that Babyface Pro FS is optimized for mobile use and works USB bus powered?

The headphone outputs of BBF Pro FS are "sufficient", but because of mobility/USB2 bus power
on the weaker side of the RME product line.

The details you can look up yourself in the technical section of the manuals of the particular products.
The Extreme Power Outputs of ADI-2 DAC/Pro FS and ADI-2/4 Pro FS are much (!) more powerful.

To put a few examples ..

Babyface Pro FS:
Output: 3.5 mm TRS jack, unbalanced
Output impedance: 0.1 Ohm
Output level at 0 dBFS, 1 kOhm load: +7 dBu
Output: 3.5 mm TRS jack, unbalanced
Signal to Noise ratio (SNR): 114.8 dB RMS unweighted, 118 dBA
Max power @ 0.1% THD: 90 mW

UFX III
Output impedance: 2 Ohm
Maximum output level at 0 dBFS, High: +19 dBu
Signal to Noise ratio (SNR) @ High: 116.5 dB (AES17), 118.5 dBA
Max power per channel @ 32 Ohm load, 0.02% THD: 200 mW (2.5 Vrms, +10 dBu)

ADI-2 DAC FS:
- Output impedance: 0.1 Ohm
- Output level at 0 dBFS, High Power, load 100 Ohm or up: +22 dBu (10 V)
- Signal to Noise ratio (SNR) @ +22 dBu: 120 dB RMS unweighted, 123 dBA
- THD @ +18 dBu, 32 Ohm load, 1.2 Watt: -120 dB, 0.0001 %
- Max power @ 0.001% THD: 1.5 W per channel

ADI-2 Pro FS R BE:
- Output: 6.3 mm TRS jack, unbalanced, stereo
- Output impedance: 0.1 Ohm
- Signal to Noise ratio (SNR) @ +22 dBu: 120 dB RMS unweighted, 123 dBA
- Output level at 0 dBFS, Ref Level +19 dBu, load 100 Ohm or up: +22 dBu (10 V)
- THD @ +18 dBu, 32 Ohm load, 1.2 Watt: -110 dB, 0.0003 %
- THD+N @ + 18 dBu, 32 Ohm load: -107 dB, 0.00045 %
- Max power @ 0.001% THD: 1.5 W per channel

Balanced Phones mode, as before, but:
- Output levels at 0 dBFS: Hi-Power off +13 dBu (3.46 V), Hi-Power On +28 dBu (19.5 V)
- Output impedance: 0.2 Ohm
- Signal to Noise ratio (SNR) @ +28 dBu: 123 dB RMS unweighted, 126 dBA
- Output level at 0 dBFS, Hi-Power On, load 150 Ohm or up: +28 dBu (19.5 V)
- Max power @ 0.001% THD: 2.9 W per channel (!!!)

ADI-2/4 Pro SE
- Output impedance: 0.1 Ohm
- Signal to Noise ratio (SNR) @ +19 dBu (Hi-Power): 120 dB (AES17), 122 dBA
- Output level at 0 dBFS, Ref Level +19 dBu, load 100 Ohm or up: +21.5 dBu (9.2 V)
- THD @ +20 dBu, 32 Ohm load (1.9 Watts): -124 dB, 0.000057 %
- Max power @ 0.001% THD: 2.1 W per channel

Balanced Phones mode, as before, but:
- Maximum output level: +27.5 dBu (18.4 V) @ 0 dBFS
- Output levels switchable: IEM +7 dBu, Lo-Power +13 dBu, Hi-Power +25 dBu
- Output impedance: 0.2 Ohm
- Signal to Noise ratio (SNR) @ +25 dBu: 120 dB (AES17), 122 dBA
- Max power @ 0.001% THD: 4 W per channel

Between BBF Pro FS with 90 mW and ADI-2 series of products is quite a difference in volume to drive even the most insensitive headphones with high impedance.

Ramses, thanks for the info!
ADI-2 Pro FS R BE is about 700 euros more expensive than ADI-2 DAC FS and I will see from what you wrote in your blog if it is worth trying to buy it or not, specifically for my needs, but I can`t ignore the fact that there are those balanced phones outputs that can make a difference.
I would like to know a little about what I wrote above regarding the value of -5 dBu as default for ADI-2 DAC FS.
What value or values do you recommend for XLR in the case of studio monitors?
Babaluma was talking about +1 dBu for XLR.
Judging by this logic presented by Babaluma -5 dBu is a good value given the fact that I keep the volume around -60 dBr or am I completely wrong? Or I have to set with the lowest value from XLR, namely +1 dBu.

Hmm, interesting. I wonder why all these volume reducing products have the lowest value of -30 (or at least that's what I noticed).
I would have liked the minimum to be -60 with steps from 10 to 10.
Speaking of Palmer MONICON that appears in the recommendations, I'm not sure if it's this version, but I understood that it's not exactly "transparent".
Is the recommendation made due to the fact that the Palmer MONICON is a passive controller and it is assumed that it cannot "color" the sound?

So if MC's first recommendation is the JTS Mic attenuator MA-123, it means that it is the most suitable, especially since he is part of the RME stuff from what I saw in his description.

Babaluma, did you choose the ones from Shure hoping they are better?

I was also curious about the XLR cables - are there any recommendations in this regard as well? -

The Shure ones had lower capacitance, and MC had said the lower the capacitance the better, which is why I went for those instead of the JTS. I'll always spend more if there's even a slight chance of better sound quality, but I'm betting I wouldn't be able to tell the difference in a blind test.

https://www.jts-europe.com/products/mic … es/ma-123/

https://www.shure.com/en-US/docs/guide/A15AS

I don’t know which “audiophile” source claims somthing like this, and why.

In fact ADI-2 drives every headphone exist with no problems, even power-hungry ones.

Planars in general are an easy load for amps, as they are purely resistive (no phase shift) and have almost no back EMF (Electro Magnetic Force) that can tax an amp.
This means, the amp does not need to eliminate energy stored in the vibrating diaphragm that comes back into the amp, like it happens with dynamic drivers that can be relatively strong “motors”.

Not that ADI-2 has any problem to do that in case needed.
It’s headphone output stage works very stable, which I can confirm by having it pushed really hard at times.

Tip: the best headphone I found for mixing, despite owning a lot TOTL ‘phones, are the cheap Drop/Hifiman HE4xx.
The resulting mixes transfer great into the real world.

Their headband construction is said to be a bit fragile, but at that price - who cares.
I had no problems with mine, and don’t treat them like raw eggs.

> Ramses, is there a quick way to switch from studio monitors to headphones,

I am switching between monitors and phones by reassigning one of the 4 front buttons (e.g. I/O) for this purpse.
When doing this you have a ramp-up of volume for safety reasons.

I do not need Hi-Power outputs because none of my headphones require this (Audeze LCD-3, Aeon 2 Closed Back).

> Also, maybe it's just me, but don't you think that the "Loudness" function can be useful to check the mix
> you're working on from time to time?

I do not use Loudness is for mixing and mastering. Makes no sense.
I use it for listening at lower volumes where it prevents a too flat sound when listening at lower volumes.

Its not about emulating a HiFi system. Mixing and Mastering happens usually at defined volume levels.
Some people master at volumes up to 80dB. Do you listen to music every time at this volume? Surely not.
When being at work I sometimes listen to music.
And this background music sounds definitively better with dynamic loudness otherwise it sounds too thing.
What could be better than making this adjustment dynamic?
This dynamic loudness is exactly for this purpose.

Same for my HiFi.. I often use my HiFi for streaming TV, but I am not that kind of guy listening very loud.
Same for concerts ..
There I prefer the dynamic loudness so that it sounds good even if the volume is on the lower side.
Well .. and this works fantastic dynamically for every lower volume ...
Its even possible to add +3 dB Bass additionally to have a little bit more bass if the lower volume doesn't drive your speakers well enough in the bass district at lower volumes. All in all, very useful.

Thanks for the info!
Hmm, for me the default was "off" for "auto ref" in the case of headphones. It makes me think about those who sold me the dac because I bought it as a new, sealed product.
It was obviously sealed when I received it, but nowadays it is easy to forge a seal.
When I opened it from the box, it seemed strange that I saw 2 fingerprints on the dac, but I thought that maybe it was like that from the factory, but now I'm thinking about other possibilities.
Unless I pressed something wrong on the remote control at some point, but I have the impression that this function cannot be changed from the remote control.
I want to mention that when I unseal any product, whatever it is, I wear gloves just to be sure that the product is in good condition, it has no fingerprints or even defects.

Of course it is off as default. Safety first! Want to get your ears blasted on first use when using sensitive headphones? Or IEMs going up in smoke? No. So Low Power is default.

That is a misunderstanding. Chapter 12.1 is named 'Subpage Settings' and describes the settings of all Settings subpages. As the picture shows AutoRefLevel is On for Line Out, but it is Off for Phones. Easy to verify by resetting the unit.

Note that your quote is not part of the manual. It does not say 'Phones: AutoRefLevel...'.

The AD and DA resolution is the same for all filters.

The filters offer various compromises between:

- Frequency response linearity.
- Impulse response.
- Aliasing suppression.

One has to give for the others to take.

E.g.:

- NOS has perfect impulse response, but early high frequency rolloff and extremely bad aliasing suppression, that IMO clearly spoils the sound with audible artifacts.

- SHARP has perfect frequency response and good aliasing suppression, but unnatural pre-echoes in the impulse, that I receive as smeared treble response.

- AKM’s SLOW and ESS’s SD-LD have good impulse response and aliasing suppression, on cost of a little rolloff in the upmost treble, that even could be compensated with the PEQ (even though for me it’s good enough as is).

All this is valid for sample rates of 44.1 and 48 kHz only.
Above the effects are way out of the audio band and inaudible for me.

MC, I would like to know if in the recommendations you made regarding the volume attenuators, the Palmer MONICON is a version you have verified and works as well as the JTS Mic attenuator MA-123 which I understood to be your first choice on this topic. I ask you because Palmer MONICON is much more flexible than any of the volume attenuators.

What do you think if a vst like BitShiftGain(https://www.airwindows.com/bitshiftgain/) was put on the master bus(last in the chain) and it would reduce the volume by 30 dB(5x6dB) or as much as everyone wants to reduce.
Can this be considered a temporary solution to check how it sounds with a volume attenuator?

Bringing down the digital volume of course is NOT the same like passive attenuators that work in the analog domain.

For a quick an dirty test of ADI-2’s functionality, specifically Auto Ref Level, it does work, but that’s it.

For signal integrity ADI-2’s own Digital Volume Control is superior, as it avoids the typical artifacts of lowered resolution.
See DAC manual page 65, 66, chapter “31.13 Digital Volume Control”.

I’d immediately go with the passive attenuators, they don’t cost a fortune.

I do use my MAGIX Sequoia DAW, it has sample rate conversion built in with selectable quality steps.

Even the 2nd best quality step here is already beyond good and bad.

There’s a freeware, R8brain, that can be configured in all parameters, specially very fine adjustable sample rate figures.
I use it for the purpose of fine-tuning audio/video synchronization on multi-camera video productions.
R8brain has perfect audio quality too, no artifacts.

Regarding the "Loudness" function, I was looking at the picture in the manual about this subject and what I'm curious about is: basically it's an eq+volume utility that can be programmed in a daw like Ableton/Bitwig with the help of some macros to copy what the ADI-2 DAC FS does when "Loudness" is activated?
Also, why was the value of 7 dB chosen for B/T in "Loudness"?

Ramses, this is not just about the V shape EQ setting I guess? Are Fletcher-Munson curves integrated in this process for the "Loudness" function? I guess not just as points on an eq but it's more than that.
I'm very surprised by "Loudness" and I always tend to return to this function because somehow it helps me find those piercing frequencies in the mix much easier. I can hear more easily which frequencies "cut" the ear.
It's probably also due to the fact that I'm exclusively in the bass heavy music area where everything is incredibly loud(only drum and bass count in this type of mix).
But when I go back to the old school of bass heavy music where the sound was not clipped and maximized like today the "Loudness" function doesn't seem to have the same effect anymore and it can start to be a bit strange as if it were a dynamic eq inside "Loudness".
For example, an old school bass heavy track, if it were remastered and brought to today's levels, would benefit even more from the "Loudness" function. At least that's how it seems to me, but I don't know why I perceive this.
Obviously I could be wrong, but that's how it seemed to me.

Kais, checking the forum and reading here and there I noticed that you like the Slow Filter for mixing, but I don't understand what this „ESS’s SD-LD" filter that you mentioned above means. Short Delay (SD) but LD?

But for the ADI-2 DAC FS(ESS version), since there is no SD LD, which filter would you choose?

In the RME ADI-2 DAC FS manual I see that it says: „The unit with AKM has a filter called Short Delay Low Dispersion, while the one with ESS has a filter called Brickwall (the other filters, SD Sharp, SD Slow, Sharp, Slow, NOS, are identical).”

Yes, see the table with the model differences...
https://forum.rme-audio.de/viewtopic.php?id=32506

In addition to all this, MC wrote in 2017 on the forum: „In the studio I would not use the ones that negatively impact frequency response, for obvious reasons. That leaves you with SD Sharp and Sharp. As you can see in the DSO pictures you then have the choice between pure post ringing (default) and reduced ringing, but both pre and post. Choose whatever you like!”

The picture that Ramses posted shows everything very clearly.

About the SRC function, I would like to understand a little: it exists only in the Pro version and I would like to know in which situations in the studio it helps?
I mean for someone who is only ITB and only needs DAC and nothing more.
I quote from the manual: „When using the internal clock, every SRC also works as a jitter killer. However, the ADI-2 Pro is equipped with SteadyClock FS, thus operating as perfect jitter killer with any clock source. However again, a jittery input signal might degrade the quality of the sample rate conversion. The ADI-2 Pro therefore has a second SteadyClock exclusively for the current SRC input signal to make the sample rate conversion process as reliable and transparent as possible.”

Can you practically feel the difference in sound between SRC: ON and SRC: OFF?