1. Resampling - up or down - is a destructive process that changes the audio.

Sample rate is not “the higher the better”, but playing a file at its native rate keeps its integrity.

2. A computer’s software based resampling doesn’t necessarily use the best algorithm available.
It’s more like offering the functionality with as low processing power as possible.

3. ADI-2 offers all it”s functionality unlimited up to 192 kHz.
Above available DSP-power needs to cut down on the features, see ADI-2 DAC’s manual page 16.

This might have been the major point and is the biggest sound change from 192 to 768 kHz sample rate:
EQ, Loudness function and other DSP processings are switched off.
Only if you anyway didn’t use them it makes no difference for ADI-2.

4. Personally I do hear a bettering sound difference (not night and day) going from 44.1 to 96 kHz.
This is with ADI-2/4 Pro SE and a high end analog turntable connected - listening to analog vinyl discs.

Further up the sound doe not improve or change any more for me.
Still, for that purpose I do use 192 kHz - why not if it’s there

5. When streaming - Tidal - I use the automatic switching based on the audio track’s SR, my combination of iPhone / RME ADI-2 offers.

Thanks for your answer.

I run my set up in a way which audiophiles would disapprove of! Using the RME's default Loudness function gives remarkable sound on my active ATCs as they are well known for really only coming alive at higher SPLs. When oversampling, you state that all DSP in RME is off, and that's why the ATCs go back to sounding very flat.

Out of interest, if oversample the output at 768kHz on a 192kHz Tidal stream but at much higher volume where the ATCs start to work, would I hear any degradation in sound?

ATCs if I recall correctly, are studio monitors.
They are intended to sound “flat”, in the sense of neutral, kind of.
If they would sound hyped in any way, the resulting productions made on them would sound boring, as they would project a wrong picture of the recording.

In studios the standard listening level is 80-85 dBSPL, that’s quite loud, too loud for most domestic users.

So, using ADI-2’s loudness for adaption is a valid way to go.
If set properly, Loudness even goes partly or fully out of the way when dialing up louder.
Adjust it to taste.

According SRs:
I don’t see a point in using anything above 192 kHz, to the contrary there’s no advantage, only downsides.
Upsampling 192 kHz to 768 kHz using a minor quality algorithm doesn’t make sense, the result is a degradation.

The reason I explained in my posting above.

What?

I take what waedi and KaIS writes far more seriously than what you write for the simple reason that they have earned it.

As the saying goes: "What goes around, comes around."

The tone you're using here is completely unacceptable.

KaiS appears to have a depth of knowledge that might exceed your ability to comprehend. Insulting others due to a potential learning difficulty is far from courteous.

If you've lost the ability to self-reflect and can no longer communicate with others appropriately, it might be best for you to stay away from online forums. That would benefit both you and the other participants here.

I have tested a lot of upsamplers and resamplers
To my opinion e.g. PGGB-RT has a really good sound but it takes about 14 seconds on my computer to recalculate all 17 tracks of one CD for 768 kHz:

https://mobidrive.com/sharelink/i/YmVhd … XRt0OqemIR

Additionally a rather high input for Analog is displayed  if I use a rate higher than 192 kHz

https://mobidrive.com/sharelink/i/YmVhd … KuklzQIw8t

I think there should be no Analog Input

Upsampling is just another form of digital filtering to avoid antialiasing on systems that have poor DA.

If audio signal is sampled 44.1 kHz, no upsampling, resampling, digital filtering is not able to add any information to signal. It may only better or worse mimic ideal filter which has to cut off frequencies above 22.05 kHz to avoid aliasing.

And there is no ideal solution for that case by definition. There are always compromises by definition, one cannot beat mathematical reality.  One may prefer one method, the other another, but none of them is perfect, none of them is "right".

Ok, I have checked the manual

I can't hear this effect shown as Analog Input during the USB playback
However the Analog level on the display is higher than my USB 1/2 output level at 768 kHz

https://mobidrive.com/sharelink/i/YmVhd … KuklzQIw8t

Note that this chapter only talks about 192 kHz because DigiCheck is limited to that, and is used to show that there is no change in the audible range. At 384 and 768 kHz the high frequency noise gets much worse.

The AD-side shows any frequency within the current sample rate range and is not limited.

Ok, to my understanding the steps on the curves are much smaller e.g. with the best settings of the PGGB-RT and near to the ideal.
Additionally there might be less wrong interpolations.

As far as I know I'm always using the ADI 2 Pro with the SRC switched off

Additionally I'm meaning the steps as part of a frequency curve as shown below

https://www.open-end-music.com/filedata … 1684725226

Ok, here is the link:

https://mobidrive.com/sharelink/i/YmVhd … uwp3mF5uSP

I have just found a similar frequency curve with steps also at the end of page 77 in the ADI 2 Pro manual

Apart from that I just want to listen to my favorite films and music with the ADI 2 Pro in the best possible way.
If I have done something wrong with the PGGB-RT and e.g. the switched off ADI 2 Pro SRC or other settings I would like to correct it.

Ok, I think the comparison between e.g. the original and the upsampled version is harder if I turn PGGB-RT on than off

Switching PGGB-RT off is done immediately but switching upsampling to 768 kHz on again takes about 15 seconds for one track on my computer

I think it would be almost unusable if I had to wait 15 seconds after each track, but fortunately it is gapless because the upsampling of the next track will be executed in time before the next track begins.

If I got you correctly (based on an ADI 2 DAC FS w/ ESS chip), the Brickwall filter would be your favorite.

If I compare the impulse responses diagrams of the manual on page 58 - Brickwall and Sharp on page 57, I can't see any (audible) difference - they look pretty much the same to me. But the frequency response of the sharp filter reaches 20 kHz without any loss, while Brickwall already lost -4.5 dB (beginning at ~ 19 kHz). Therefore, I doubt, that Brickwall could be audibly different compared to the sharp filter.

I’m on ADI-2/4 Pro SE, and SD-LD is my favorite for this ESS chip model.
Manual pages 88, 89.
I only lately stumbled over the Brickwall filter and kind of like it too.
It does sound different than the Sharp filter to me.
Still the SD-LD sounds more “open”.

The whole thing is only relevant at 44.1 and a little less at 48 kHz sample rate, just to bring this fact into mind again.

That scope pic shows a 'signal curve', if at all. And no steps at all.

Maybe time to throw in Monty:

https://www.youtube.com/watch?v=cIQ9IXSUzuM

And as this thread is about sample rates here is Dan on this subject:

https://www.youtube.com/watch?v=-jCwIsT0X8M

That is what we try to explain you. Upsampling does not bring any new information. It is not going to improve signal. Especially for mastering grade DA. Hi quality upsampling might help in case one uses very poor DA hardware.

Yes, it is true for any signal composed of frequencies up to half of sampling frequency. It is mathematically prooven, so there is no question about it.

There was that nice example with square signal. Ideal square signal has frequencies going indefinitely high.

It shows that part of video with square signal. What goes in goes after reconstruction out.....

I haven't seen the mathematical proof of that it works. But I doubt I would be able to comprehend the math behind.... And I had lots of math lessons including digital domain math and I like math....

Simply, it works just fine. Yes, it is rather hard to believe, but it evidently works just fine. When touching digital one cannot use most of experience from analog. As I said earlies, digital domain and analog domain are very different beasts/worlds. They have its own rules and its own math.

For example analog world has nothing like aliasing. That is unique to digital. Similar with dithering....

But if you do not trust it works, you may make a test. Get complex signal in analog, make A/D at 48 kHz, make D/A, feed signal to A/D and compare first version of signal afrer A/D and second version after A/D. They should be almost the same.

Or you could compare signals in analog. The best would be to invert phase of one of them and add them. If everything works right, they should cancel and the result would be just residual noise. But one has to adjust signals for delay made by AD and DA so that they are time alligned.

Put both versions into a DAW and make the null-test.
It will turn out the two files will cancel out each other and your impression was wishful thinking.
Also use the free plugin Youlean loudness meter for measure peak dB and LUFS.

I think I haven't written what you have quoted

According to my check the quote above is from @Kubrak

Thanks for your notes, Kai. Adi 2 DAC FS @ESS doesn't provide a SD-LD filter but a Brickwall filter, which shows different behavior as your SD-LD.
Do you by any chance have a Youtube link which shows best the differences? I would be interested. I would compare Brickwall with Sharp here - but I doubt I could hear any difference because they look pretty similar.

You can't put two files with totally different sample rates into a DAW. Unless one of them is automatically downsampled, which makes the comparison invalid.

No matter how you downsample this will not happen. The upsampling alone already introduce slight phase shifts over frequency, which does not allow simple null tests anymore. You could do a better test comparing them with Deltawave, which can be set to ignore (compensate) inaudible phase shifts.

Yes, there is something called “Hybrid…” or so that I liked best.
I don’t have the unit with me and it’s not in the manual, so I can’t look for the full name.