But you need the optional “volume control” box.
The volume control box is either the Vivaldi DAC or a pre amplifier.
Thank you Pete @PAR, that’s what I will aim for. I’m developing a better intuitive feel for what the different settings do to different music. The ‘pressing buttons’ phase is down to half a minute now. 
I listen to headphones only, so the depth of the soundstage is hard to perceive for me. I did encounter recordings though, where upsampling made placement of vocals fuzzy or even slightly wandering left and right. Moreover, when some records were upsampled the music presentation became too orderly or more boring.
Does padding mean the eight extra bits are filled with zeroes? My understanding was you can represent any given waveform that much better with more available values - 24 bit instead of 16 bit (subdivision of the Y-axis of the waveform whereas the sampling rate determines how often you slice the waveform - subdivision of the X-Axis). I only looked at Andrew’s article on the Ring DAC but not further. Would an upsampler in some form interpolate between the values a 16 bit word delivers to create the 24-bit word?
Attention to detail!Yes it does but whether or not there is interpolation in this specific case needs technical advice from dCS as James Cook’s series of articles has not really touched on this subject ( so far) and Andrew’s was really to explain that DXD is a resolution in its own right and not a method of upsampling. I would think however that using interpolation would offer only a small gain, if any, as a 24 bit word offers a 144dB dynamic range which for music is an over specification and effectively means that part of this capacity can contain no musically relevant data. Paul Miller ( Group Editor AV Tech Media, President EISA etc.) has speculated that simply using 24 bit words with 24 bit processors provides a benefit in processing “ease” and it is that which may result in a sonic benefit.
I dug a bit deeper and pulled the following from here
(…)
That doesn’t give us the answer to the question whether Vivaldi Upsampler does any processing beyond adding zeroes in order to get from 16 to 24 bit depth. It tells us however there is value in upsampling/processing the bit depth beyond adding zeroes.
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Thanks for searching that out. I am happy that I am still learning things in my seventies.
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The Rossini manual indicates that upsampling is optional.
That is not correct ( unfortunately). There are indeed options. However they only concern the mode of upsampling. They do not enable non-upsampling replay. The options are to select between DSD , DSDx2 and DXD upsampling. There is no Clone Mode which is required for upsampling to be avoided. NB; I have just looked through the latest User Manual for Rossini DAC; v.03 . See pages 23 and 39 in the manual and also the menu function map p.32.
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The manual seems to differentiate DSD upsampling vs DXD oversampling.
Yes, they are different. One is a form of encoding using delta sigma modulation with the trade name DSD ( Direct Stream Digital) and the other, DXD, uses pulse code modulation ( PCM).
The manual says DSD upsampling is optional. Selecting DXD causes the dac to not do upsampling just oversampling. All dCS dacs do oversampling of pcm data. The manual never says the Rossini upsamples before oversampling. Based on the manual I’d say the statement “Can’t shut off upsampling on Rossini” is false. Not sure if there is a functional difference between upsampling + oversampling and oversampling.
I am sorry Katzsky but you are just playing with semantics. To put it very simply; if you send a redbook 16/44.1 file to Bartok or Rossini the sample rate will be changed (upsampled) to DSD64 or DSD128. Those two options are what the manual refers to as optional 1-bit upsampling ( my emphasis and this is for PCM sources, ). If those options are not selected then there is default PCM upsampling implemented; to 24/352800 ( DXD). In short , speaking figuratively, you cannot put 16/44.1 in and get 16/44.1 out. With Vivaldi you can do this as it can be used without an upsampler or be set to Clone Mode.
Upsampling is a form of data conversion and differs in intent from the upsampling used as part of the Ring DAC process;
All dCS DACs do use an upsampling process as part of the way that Ring DAC works. All incoming data is upsampled to 706.8kHz or 768kHz as this is how it is fed to the 5 bit modulator as the mapping process ( how the data is distributed to the current sources) is 5 bit. However this differs from an upsampler as the resulting output from the DAC is not converted to another sample rate. 16/44.1 goes in, there is some processing which includes upsampling and, again figuratively, 16/44.1 comes out. I say figuratively as what actually comes out is an analogue signal. I have been careful to differentiate the two when writing about my particular misgivings about the sonic results, as I hear them, when selecting upsampling in contrast to no upsampling :
Unfortunately I’m not capable of playing with semantics on this topic. DCS refers to DXD as oversampling. If it’s just semantics then it’s dCS.
What you’re saying is “DXD upsamples to 352khz using oversampling sequence then oversamples pcm data”? Maybe I expect too much from technical writers since this isn’t communicated via the plain text. Either way… how is it functionally different to upsample to DXD when the process is exactly the same code used to oversample?
The only thing I know for sure is that the writer is very careful using the word “oversampling”. I don’t know the difference but the writer is telling me there is a difference.
Yes indeed, the difference between oversampling and upsampling has always been made by dCS and as I mentioned elsewhere when they introduced the first home use upsampler they both invented the term and published a white paper on the difference between upsampling and oversampling which, frankly, I found pretty incomprehensible back then. Sadly I don’t think it is still available. If anyone has a link I would appreciate it.
However, whatever the precise wording, the bottom line is that DXD is always 24/352800 and not 16/44.1 or 24/44.1 or 24/96 or any other PCM rate. There is no option to deselect this except by selecting one of the the two 1-bit upsample rates in which case the PCM is then sample rate converted to DSD64 or DSD128. NB: native DSD files are not upsampled in this way.
I guess dCS uses the word ‘oversampling’ when applied processing is ‘only’ changing the frequency and adding zero + dithering at the end of the 16 bit data sample. In short, PCM In gives PCM Out after processing. DXD is PCM data. So 16/44.1 over sampled to 24/352.8 is oversampling according to dCS.
However dCS seems to prefer referring to upsampling when PCM data is converted to a DSD audio stream.(quite different, in nature, from the PCM data).
They probably have a good reason to do so.
Over… versus Up… (sampling); well, this is always manipulation/processing of the original data. The ones Peter is not Fan of … (same for me, although I never tested with a genuine dCS upsampler itself).
I have spent a little time this afternoon looking on the web at answers to the question “what is the difference between upsampling and oversampling ?” Frankly I think that Hans Beekhuyzen comes closest. After explaining that technically they are similar at one point he concludes that the words are selected in accordance with how the manufacturer chooses to use them 
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John Atkinson in Stereophile touches on the subject but I don’t think moves far away from explaining why he thinks that upsampling improves the sound.
I am posting links to both but , from my viewpoint or from my hobby horse if you like, I am taken by a point made by the well known recording engineer Bob Katz in a letter following Atkinson’s piece. He remarks " some people may prefer the sound of the asynchronous converter on some recordings, because the vagueness of imaging is aesthetically pleasing (shades of vinyl?)." According to Katz the dCS upsamplers of the time ( 2001) used a Crystal Semiconductor CS 8420 chip which was asynchronous. I have no idea what ingredients are in the current upsampler recipe but, irrespective of cause, I find it heartening that the artefact of odd imaging which I am very aware of was recognised 20 years ago, at least in professional circles .I no longer feel so isolated. I owned the dCS Purcell mentioned by Katz and, as I have written in other threads, that was around the time that I started to have my concerns on this issue
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Pete, slight mistake there.
The actual quote from Bob Katz in his letter responding to that Stereophile article was:
I note that you have been exploring upsamplers. Of the commercial units you have reviewed, to the best of my knowledge, all of these samplers and upsampling converters, except for the dCS 972 and Purcell models, are based on the Crystal Semiconductor CS 8420.
I haven’t peeked inside my dCS 974, but I seriously doubt dCS was using any commercial off-the-shelf ASRC, even at the time, it was always their own algorithms on FPGAs 
Yes, I too was surprised by the mention of the CS8420. Honestly my reading ability is getting increasingly compromised by my sight loss. Anyway my main point was that, irrespective of cause, the possibility of odd imaging related to upsampling was already recognised 20 years ago.
Actually Pete, you may have missed a little nuance to Bob Katz’ comments.
I have done the listening tests, and, under identical jitter conditions, recordings that have been upsampled via a good-quality synchronous converter sound about “two points” better, with a wider, more stable soundstage and imaging.
Of course, some people may prefer the sound of the asynchronous converter on some recordings, because the vagueness of imaging is aesthetically pleasing (shades of vinyl?). Professionals who upsample will generally avoid asynchronous converters because we want to get those last two points of sonic performance.
So, he was actually comparing the sonic results of a synchronous converter to an asynchronous one, not to no upsampling. He wasn’t against upsampling per se, just ASRCs.
Which begs the question, exactly how does dCS’ Upsampling interpolation work? I don’t think I’ve ever seen any technical explanation 
In a quick scan of all the Upsampler manuals, only the Purcell Manual has this to say;
This is achieved using extremely powerful and accurate digital interpolation filters, which introduce negligible levels of distortion.
No other manual even mentions the word interpolation 