Thanks for this. I have read the articles. I am not very knowledgeable with regards to the detail, so please tell me if this simple interpretation is right.

Firstly : Ideally you want ADI  volume to be at about -18db(r) at normal listening levels.
For me normal is about -28db(r) when using RCA.
If I used XLR normal would require a further 6dB of attenuation, making normal about -34dB.
So RCA is preferred here.

Secondly: My Power amp gain or input sensitivity is not adjustable. But I could use passive line attenuators to reduce the line level by 10dB, which would mean the ADI  normal volume would then be around the preferred -18dB(r).

Thirdly: As it is, there is no audible hum or buz at no signal, from the ADI / AMP  combination. So is it worth making any changes?

Please let me know if I’ve understood roughly right or am confused.

Thanks for you help and patience.

I don't think you can make that sweeping statement. Yes, it's true that XLR of ADI 2 DAC FS has a gain + 6dB compared to RCA, but: There are power amps out there, which have a -6 dB reduced sensitivity with XLR compared to RCA (e.g. T+A A200 - +37 vs. +31 dB (RCA / XLR)) - therefore it doesn't matter using RCA or XLR from a level perspective. Or did I get something wrong?

You're perfectly right. The T+A A200 is a pretty bad amp - it's not worth the money at all (even 500 € would be too much). It's just horribly bad from my perspective. I wouldn't buy it today any more if I would have understood those facts completely a few years ago.

The T+A A200 noises audibly even using speakers with lower sensitivity. The measured values presented by T+A are so much "misleading" compared to the values measured by specialized press (see the links on the T+A A200 homepage). If you read those values, you understand, that this amplifier must be noisy.
They brag about a development partnership or something like that with Purifi - I don't understand, why Purifi accepted a massive worsening of there concept. A AUDIOPHONICS HPA-S400ET is much better (measured by amir of audioscience review) as the T+A device at much lower costs.

But the "good news" is: because the T+A A200 noises so much even with inputs shortened(!!), you can't hear any noise from the ADI 2 DAC FS at all (even if DAC is set to + 0 dB), because the noise created by the DAC is much to low to be audible after the T+A A200. The bad development quality of the T+A A200 overlays all ... .

Before I bought the ADI 2 as preamp for the T+A A200, I used the T+A VVM (Vorverstärkermodule). This is really bad (in conjunction with T+A music player balanced). T+A rates it with 109 dB(A) noise voltage ratio - never ever! The VVM (310 €) basically consists of a TDA 7303 (price per unit $ 0.18 if > 100 pieces).

Looks like a good choice.

Gain/sensitivity is not spec‘ed, but seems practically reasonable (read: on the lower side) which is a requirement for clean sound.
120 dBA SNR points in the same direction.
Saves from using attenuators.

The distortion vs power chart is textbook perfect, if such has influence on sound - on this low level of distortions - it‘s probably positive.
http://www.hifisound.de/out/media/pdf.p … %3D%3D.pdf

BTW:
Measured SNR is 6 better for RCA input, over XLR.

The 92 dB SNR here is referenced to 1 W @ 8 Ohm / 2.83 V:
(1) https://www.reichmann-audiosysteme.de/i … eoplay.pdf

Interesting the same magazine published 11 to 16 dB better SNRs here:
(2) http://www.hifisound.de/out/media/pdf.p … %3D%3D.pdf

Based on the lesser result from (1):
Combined with a typical Hifi loudspeaker‘s efficiency of ca. 85 dB SPL @ 1 meter distance @ 1 W, the audible noise would be 7 dB BELOW the threshold of hearing of a young, perfectly healthy ear.

Even in a perfectly quiet room there‘s no chance to hear this noise, and air molecules motion’s noise is in the same magnitude.

Factory specs of power amps SNR usually refer to nominal full power, “Dynamic Range” would be the better term.
230 W @ 8 Ohm is 24 dB above 1 W.

92+24=116 dB, close to the 120 dB factory specs.
None of the measurements tell the filter (A, CCIR, Lin) or detector characteristics (peak, rms, quasi-peak) and input termination used, which can significantly vary the result, so don’t expect 100% congruence.

Resume:
Crunchynut doesn’t hear noise, so there’s no noise problem.

... this alone is indeed remarkable!

I even don’t get the point.
A class A/B power amp isn’t possible without some idle current.
The 70 W is completely in the usual range.

Contemporary switched mode/digital power amps are a different story, but I wouldn’t compare those- although sound-wise I even have very good experience with them too.

Measurements that don’t publish ALL circumstances can just be taken as no more than thumb-estimates.

This is why standards like the DIN HIFI 45500, IHF A-202/ EIA RS-490, IEC 60268 etc. exist, but most contemporary review(er)s don’t even know them or don’t care, or measure to their own “standards“.

The amplifier noise measurement in dBA RMS (A-weighting filters out hum) relative to nominal full power has established as marketing’s “quasi”-standard, because it delivers the highest dB-values possible.

It’s perfect to impress potential buyers.

To further “improve” the figures without improving the actual device, the Chinese started to measure power as momentary Peak to Peak, opposed to long term RMS.
Peak to Peak is a power that’s at no single moment delivered to the load, in reality.
This alone achieves 9 dB or more higher values (if you stay with RMS for the noise) for the same amp - true progress

Anyway, measurements can tell a lot, but finally it’s the practical result that counts, for the good or the bad.
As you found with your T&A A-200.

I’m happy we have such great devices like RME’s ADI-2-series where, e.g., it’s close to impossible to squeeze out the slightest amount of audible noise in any configuration.
A fact that highly correlates with ADI-2’s extremely well documented measurements too.

I’m not 100% sure about your classification of Class D amps not being “digital”:

The analog signal in fact is converted into a PWM or pulse train signal, containing two state signal levels only, the primary definition of digital.
There’s even a clock.
On the other hand “analog” design parts can often be found too, like an overall feedback loop.

Probably Class-D is a good label for the whole range of these designs.

If you switch from unbalanced to balanced different electronics are used in the signal path.
If your unbalanced path sounds “grainy” it’s technically broken, defective.

That’s a possible reason:

PC’s, via the USB path, can spill a lot of audible and inaudible garbage into the signal path.

ADI-2 by itself, internally, is perfectly immune to this.
All garbage is kept on the outside.
You will never have any loss of audio quality on headphones plugged directly into it, no matter what happens on PSU or USB side.

Power- and headphone-amps connected to ADI-2 externally is a completly different story:

On unbalanced interconnects the cable’s shield is shared for a double-porpose:

(1) Shielding from electric stray fields.
(2) Providing the common ground reference for the audio signal, which incorporates running possible noise garbage to the next device.

While (1) usually works quite well, (2) incorporates a big problem, if noise and hum currents that origin from mains power and USB run through this shield:
Due to the voltage drop at the cable shield’s impedance of typically ca. 1 Ohm a noise signal develops, that fully adds to the audio signal.

The best known of these effects is the hum, caused from the so called “ground loop”.

Less obvious are distortions from high frequency intermodulations:

As power amps have a limited “speed”, they cannot handle the mentioned USB high frequency garbage, they can distort if they don’t use proper filters at their input.
This distortion “demodulates” the primarily inaudible high frequencies into the audio band.

If you ever heard the “tock, tock, tock...” from a cellphone spilling into the stereo just before it rings- thats such an effect.

Furthermore the HF saturates the power transistors, preventing them from doing their audio job properly.

With balanced interconnects audio and shield are separated, which circumvents those detrimental effects.

Following this philosophy an ADC or DAC isn‘t digital, as by the dither noise, both conversions don‘t deliver an exactly deterministic result throughout.

For a Class D amplifier I‘d like to think it from the other side:
It‘s not a classical analog amp, as it‘s signal handling has binary inter-steps.
This results in limitations an analog amp doesn’t have, like a certain upper band-limit (related to the Class D’s clock).

It might, on the other hand, not be a digital amp too, as it does not use binary symbols that represent numbers, and it does not amplify by processing binary symbols with multiplications.

Anyway, the main point and basic idea behind a Class D Amp always had been to be as energy-efficient as possible.
Because of this special feature, those amps are used everywhere these days, not just in the audio world.
In the technical applications the variant that uses a digital input signal with binary processing all through is widespread, as it interfaces directly with a processor.