I'll say it right now: upsampling really works.
How? Nobody knows. not even DCS have a conclusive explanation, but THAT it works becomes clear as soon as you've swithed samplerates once.
Using any digital input source, the Purcell allows switching between "cloning" or "non cloning". Cloning means that the output signal will be exactly like the inputsignal. This really works very well as a bypass. Tests have shown that there is indeed very little difference between plugging the cables into each other directly and the connection through the Purcell switched to cloning. This allowed us to easily assess the differences between upsampling and non-upsampling. Of course we also assessed the Delius dac by itself, without the Purcell. Lastly it is important to note that we used the dual AES option for all testing purposes. First, this is neccesary when going higher than 92khz, second, this sounds a lot more "analogue" than coax.
When switching from 44.1khz to 96khz immediately the soundfield became wider, airier and more nimble. But the typical square PCM sound remained evident. It's not as if everything is suddenly analog. But when you switch to 176.4khz (fourfold the original samplerate) something amazing happens: sound fills the room, so wide and deep that walls seem to have disappeared. It is so airy, fluid and amazingly finely detailed that you'd swear that the cd player was changed for a record player. Well, not quite, of course:-) But Voices really float as if attached to rubber strings in the room in front of you, not hammered to the floor as they are with regular cd, but at the same time they are amazingly sharply focused. Together with these amazing floating voices, comes a truly engulfing soundstage that wraps around you that makes you feel like taking a warm foamy bath. And everything just lingers in the air, long notes continuing seemingly forever instead of being cut off too soon. This is an effect I am familiar with only from playing lp's.
It also quickly becomes clear that a precise multiple of the original samplerate sounds tighter and more dynamic, more spot-on than a frequency that is not exactly dividable. For example using cd's in 44khz, 192khz sounds really spacious and fluid but has less attack and drive than 176.4khz. the same goes for 92khz as opposed to 88.2. The story shifts of course when you play 48khz sources. In that case 192khz is an exact multipleof the source frequency and there is no loss of energy.
So upsampling has no trade offs?
Sadly yes, there is a trade-off.
But to me it is a small one: the bass is somewhat less firm and has less drive. When switching back to the original samplerate it becomes obvious that the sound is tighter, has more pressure and sounds more "live". But now it also sounds grey, small, narrow and frankly boring! What's more: whereas we previously found 16/44 to be fine, now it cannot be accepted anymore for its grey boringness.
So upsampled CD matches LP playback?
If only. That would've been great but sadly it's still not the same. DCS gets you frighteningly close though. I have to say at this point that LP playback will only be better if you've got everything right: you need a high end record playback setup and have everything calibrated just so. Then you will beat the sound of digital through DCS. I'm used to high end record playback and analog still has the edge in that in combines a relaxed fluidness with absolute transient-quickness and dynamics. It can be smooth and quick at the same time. This is still difficult for digital.
What is the difference between upsampling and oversampling?
This is a story about mathematics and that's not my strong suit, but I can try and lay it out really broadly. Mathematical these two methods are the same, but soundwise they are very different. Why? Again (almost) nobody knows. Fact is that oversampling is always synchronous. There is always a fixed multiple by which the samplerate is multiplied. For example 4 times oversampling inserts 3 empty samples without interpolating and the bit depth remains at 16 bits. There are no new samples mad up but the task of the DA converter is easier and the neccesary filtering can be done to a lesser extent which improves the phase linearity. Don't worry: it also sounds a bit abstract to me. But it makes sense as soon as you compare it to upsampling.
Upsampling is asynchronous. It gives you the possibility to convert any frequency to any other frequency. The frequency doesn't have to be a 2 or 4 times multiple and the bit depth can be interpolated to for example 24 bits. Upsampling does create new samples by means of interpolation. This is a process of calculating the differences between two existing samples. After upsampling again the DA process is simpler and less phase problems are introduced. Still this doesn't explain why upsampling should sound better. The original 16 bit data is still the limiting factor and no actual new data can be magically created. But the difference is clear when you compare the highend digital products from the various brands that use oversampling versus those who use upsampling: upsampling almost always sounds more airy and lightfooted than oversampling.