Greetings Gentlemen,
This is an interesting thread... SACD and associated DSD signal was originally designed as the digital signal "closest" to analog, and was thought of mostly for archival purposes. By removing the "brickwall" decimation filter from the signal chain, Sony/Philips thought they would also remove the part of the signal path that affected the sound the most, and end the "vinyl vs. CD" war. I believe they have- and I also believe they really were onto something... Please note that Philips had an extensive catalog of very high quality tape recordings in 4 channels from the 70's that a lot of people really wanted put on the market, but there was no distribution format for it. They now can be found in the Pentatone SACD collection, in their original 4 channel format, for your listening pleasure...
Things have changed- most music today is recorded digitally, and as mentioned several times in this thread, the 1 bit signal only allows a very limited amount of signal processing- certainly not enough to allow EQing, reverb, mixing and all other tools required to mix down an album. Sony -also a provider of studios gear- worked on that issue, and after careful listening test came up with the DXD fiormat: 384kHz sampling rate, 24 bits depth, as indistinguishable from the DSD and DSD2 signals (DSD is 1 bit/2.82MHz, DSD2 is 1 bit, 5.64MHz). Most "high end" DSD recording studios today employ DXD during the mixing process, and transform that signal into DSD at the last stage of the mastering, using a sigma-delta modulator (traditionally 4th order, although higher order modulators are also in use). Please see the Sony Oxford consoles for more information...
So in essence, the "downsampling" process from SACD to LPCM can be done transparently, at the condition both bit depth and sampling rate are high enough.
Now, regarding the SCD-XA5400ES SACD player- it does play the DSD stream natively. Both PCM and DSD signal paths are visible on the PCB- I have put both on the scope, and noticed the PCM channel to be quiet (GND) during SACD playback, and vice versa during CD playback. Please note that the Sony is using a PCM1795 DAC from Burr Brown that supports both signals on separate inputs. That DAC is the classic "current steering" topology, very dear to the Burr Brown design team; although there are better spec'ed DACs on the BB offering, I find their latest to have improved definition, and a more controlled slam (The 1792/1794 have a bit of an overemphasized slam to my taste). Those DACs are "multibit DACs" in the sense that they will turn the PCM OR the DSD signal into a 5 bits signal using a digital Low Pass Filter- this is done in order to minimize the impact of jitter, very detrimental to traditional 1 bits DACs (both MASH and straight 1 bit DACs are completely obsolete today because of their high sensitivity to timing errors). All modern signa-delta modulators today employ this technique for jitter reduction; however, the BB DACs employ a current steering technique in the DAC itself, still sensitive to jitter compared to switched-capacitors designs. Additionally, they display a high level of out-of-band noise that needs to be filtered out (3rd or 4th order LPF is required to end-up with a flat noise floor). All those draw-backs do certainly not reduce their sound quality- but Burr Brown DACs are sensitive to jitter, and need stiff analog low-pass filtering. Period. The SCD-XA5400ES designers were well aware of this fact, and designed a very clean clock section- it can be seen close to the DAC in the upper right section of the PCB: locally regulated supply, double buffering local oscillator. They also built a 3rd order, multiple feedback LPF in the output section using "orange drop" polypropylene capacitors in the output stage (not Wima- but what is the difference, really

). Simple and very effective- and likely one of the many details that make this player a winner without spending millions.
Now, when this player is connected to the 5400ES receiver and you are playing music, what really happens? Well, the receiver proposes many digital signal processing functions that cannot be performed on the DSD signal itself, so although the connection from the player to the receiver operates in the "native" format (DSD for SACD both stereo and multichannel, 16 bits / 44.1kHz for CD), it will turn this signal into a workable payload- and all points this to be 4*Fs... As they are using 32 bits DSP, the resolution is maintained above 24 bits at all times (32 bits are never maintained because of rounding errors). Not as good as DXD, but close! So close that it will probably make no difference to the end user. They could have used 8* instead of 4*, but this at the cost of doubling the MIPS requirements for the same processes, which is likely not worth it (double the number of DSPs, etc). The signal is then upsampled to 384kHz at the end of the signal chain to feed their digital input, class D amplifier stages, or fed directly to the DACs at 4*Fs for the analog outputs.
All in all, I find those compromises very acceptable. The DSD format is really a great storage and transport medium- but is never, I repeat never directly transformed to analog without being turned into a multibit signal- even when DACs supporting native DSD inputs are used, because all of those DACs without exception are multibit DACs- this is also true for Meitner or DCs offering (the RING DAC is a 5 bits, multibit DAC). The whole point is to keep the PCM resolution "high and deep"...
Cheers!