Topic: ADC: how to preserve analog sound in the digital domain?
Try removing the A/D and D/A converter from your audio chain and check with your ears if it is the weakest link. I removed my Apogee Element (192kHz 24bit) and the sound becomes fluid and pure analog.
With good equipment like the SE Electronics Gemini II microphone, Blue Robbie preamp, a Violectric HPA V200 headphone amplifier and a Beyerdynamic DT770/DT880 headphone the difference is qualitatively perceptible. Of course 192kHz is better than 48kHz but the basic problem remains with Apogee Element AD/DA.
"For years now people have been trying to figure out why their digital recordings don’t have the warmth and feel of analog tape recordings. We try using tube Mic Pres and great compressors, but there is still something missing. There is still that blurriness, that graininess and lack of depth that comes with digital recordings."
Apogee Element uses the same ADC conversion chip as RME ADI-2 PRO FS: AKM AK5574. But even Merging Horus/Hapi use that AKM AK5578 chip.
As a comparison I tried an old Korg MR-1000 recorder but the sound is worse (although the DSD128 seems less boxy than the PCM). I had the impression of a braking and degraded sound.
It is clear that the ADC DAC chip is as important as the electronic circuit around it. For example Burl Audio B2 Bomber ADC inserts transformers in the circuit but a colorful and altered sound may not always be appreciated.
So RME ADI-2 PRO FS can really make the qualitative difference to preserve the clean analog quality sound (also through HQPlayer DSP) with DSD256 (12Mbps) or PCM_768kHz/32bit_integer (24Mbps)?
In the specific case, is it better to use the DSD256 to better preserve the impulse response as recommended by Merging https://www.merging.com/highlights/high-resolution ?