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.
I’ve tweaked my SoX settings before too, and yeah, small changes there can make a surprising difference. I’m also on a Mac mini and used Audirvana 3.5 for a long time before testing out Studio. I didn’t feel a huge leap either until I played with the filter settings. Once I dialed them in, it definitely felt tighter, more focused sound. Roon’s good for library management, but the playback never gave me that same richness Audirvana does. By the way, when I needed to convert a few old MMF ringtone files from an older phone to something usable, I found this MMF converter that worked online and handled the job fast, https://www.movavi.com/audio-converter/mmf-converter.html.
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Руководствуясь Вашими настройками своей аппаратуры могу сказать всё отлично. Пользуясь случаем хочу поблагодарить Вас. Отличная работа!!!
Wow…how about just a BIG Thank you
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Thanks again. After numerous set ups’ (All Headphone centered) I am currently pulling together the following:
Warwick Aperio
Innuos Network switch
Currently Have a sinology has running MinimServer
Have Roon Nucleusone (not happy with SQ)
Have Audrivana and recently Jplay IOS
Wanting to move to a headless MacMini with Audrivana or Jplay.
Im interested in anyone thoughts / comments. Also I have the MacMini as one of my home PC’s but want to optimize it for Audio only? Any suggestions there including best way to go about that? Much appreciated
JPlay is interesting, but it doesn’t do upsampling (as far as I know).
I am using Audirvana most of the time as it has both streaming services integrated: Qobuz and Prest Music (as a Jazz fan, Presto offers a nice collection of Jazz albums).
I am running Audirvana on a Mac Mini (last Intel generation) that streams via WiFi to my router, to which a Raspberry Pi 5 is connected via Ethernet. The Rspberry Pi runs Ausiolinux and has mod/upmpdcli enabled. The Raspi then connects via USB to my Devialet.
Very reliable configuration.
I also tested JPlay by having MinimServer running. It‘s quite OK, but I miss Presto music. That‘s something I very much appreciate about Ausirvana!
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PS
JPlay iOS works with HQPlayer both, with HQPlayer Desktop and HQPlayer Embedded, should upsamplng be an issue. See also
When Audirvana does DSD Upsampling, in my opinion it will be the perfect sound system.
What tangibly audible improvement do you think is gained in modulating 2.8MHz, 5.6MHz or 11.2MHz 1-bit PDM files to a higher 1-bit sample-rate???
There may not even be a tangible audible improvement, but it’s a feature missing from Audirvana.
I’ll only know if there’s an improvement when I can test it.
Modulation of 2.8MHz, 5.6MHz or 11.2MHz 1-bit PDM files, to a higher 1-bit PDM sample-rate is typically done offline due to the processing overhead required and latency introduced…
Knock yourself out…
Depending on how powerful the CPU in your computer is (and possibly your GPU), you might be able to try this with HQPlayer and see what you think.
Though I have a fair amount of older DSD64 stuff, these days I almost exclusively listen to PCM upsampled to DSD512 or higher resolution DSD that needs no upsampling. PCM upsampled to DSD512 in Audirvana sounds better to me than DSD64 (Rickie Lee Jones’ Traffic From Paradise is an example of a recording I own and have listened to extensively in both formats, and the upsampled PCM sounds much better to me), and the higher resolution DSD stuff is recorded with evident care. So I personally don’t much miss the lack of ability to upsample DSD, though I can certainly understand being curious about it.
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@Jud
I use a Mac Mini M2 with 16GB of RAM exclusively for Audirvana.
I also have a lot of discs in DSD64, some in DSD128, DSD256, and a very few in DSD512.
But really, that’s more out of curiosity.
I tested the PCM discs by upsampling them to DSD256, that’s the maximum possible on macOS, but today I’m using them to upsample to DXD.
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It is not clear that the SACD was produced from the tape master or digital master… I believe during this period Geffen Records primarily used the Mitsubishi digital multitracks…
The provenance of the SACD production is not clear, so this would not be my first choice to compare 1-bit 2.8MHz DSD64 to the modulated 16/44.1kHz PCM file, as most likely you really don’t know much about the provenance of the SACD master.
My two versions are the Analogue Productions remastered SACD and the 24/96 Qobuz download. And you’re quite right of course that they wouldn’t be directly comparable.
But it’s part of a general pattern of SACD purchases that I found not to exceed PCM upsampled with Audirvāna in sound quality, so I stopped buying them.
That of course is my own preference and not meant to discourage anyone else from upsampling DSD online or offline.