Several of these have graphing functions.
Well … I was led to believe that the one big advantage of upsampling here is that you can convert the signal stream in a way as to make the DACs live easier. Like feeding the DAC exactly that sort of input that would mean less or easier work for it.
My DAC is a S.M.S.L SU-1. I am not too sure about what I can do in order make to its life easier 
Input via USB-C-to-USB-C cable, output via two RCA Cinch cables.
This uses an AKM 4493 DAC chip. If you convert to DSD256 or DSD512 with Audirvana, your DAC won’t have to do any internal upsampling with that chip. The signal will just be passed directly to the final analog filter. Your DAC will be doing less of the work, so we can call that making its life easier.
To the extent your filter settings in Audirvana provide a higher quality signal to the final analog filter than the internal DAC chip would, this will also make your DAC’s life easier.
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The higher the sample-rate that you feed the DAC, the less energy it takes to produce an output and this translates into less influential noise in the DAC clocking topologies… You may or may not appreciate a difference in sound-quality… However, the AKM 4493 provides an unfettered 1-bit signal path to output and both myself and @Jud believe that converting to DSD provides the best output sound-quality… If you are managing your power/ground/earthing strategy properly and not injecting power related noise into the DAC topology through the power signal or RF and EMF noise through the interconnects, the subtleties of the artistic decisions made in the recording are there to be appreciated.
The Volume adjustment is inconsequential… for every -6dB of attenuation you lose 1-bit of dynamic range… and because the up-sampling output is 32bit to the Volume module, you are not losing real-world dynamic range in the process. Your SU-1 has been measured showing its maximum dynamic range is about 19bits… Very few if any DACs can produce 24bits of dynamic range… So you are good. 
(Theoretical dynamic range of a 24 bit file = 144dB)
Note: What must be considered with 1-bit signals is that Volume control cannot be applied without being converted to DXD 24/352.8kHz and then modulated again to to 1-bit DSD… Otherwise Volume control is unavailable or disabled. (preferred)
https://community.sw.siemens.com/articles/en_US/Knowledge/sound-quality-metrics-loudness-and-sones
Examples of real-world dynamic range:
@Audi100 … When you made your room correction acquisitions where did you have your amplifier’s volume set in the acquisition process?
28. I use the latest Room EQ on Mac OS as an Audio Unit. In measurement mode, i want to know what the message « the sound is too loud » mean … which sound ? The sound of the measurement signal ? The sound level of the recording I play to make measurement ?
This message means that the amplitude of the signal on the input of the Room EQ plug-in exceeds 0 dB during the measurement.
Set the volume controls in the signal path of your DAW to 0 dB. If this doesn’t help, your ADC works with clipping during the measurement. Move left the “Sweep signal volume” slider on the Room EQ’s panel to avoid clipping.
It is important to avoid the ADC clipping to accurately measure your room.