emperor's new clothes said:
You are spot on, Spiny. I highlighted the SA&PM10 tech several months ago in threads about class D -the PM alegedly uses a version of the Nord Ncore - and a SACD thread that was hijacked by the usual ill-informed naysayers. PS Audio employs the same method in their high-end DirectStream dacs.
http://www.psaudio.com/directstream-dac/
The launch by KI in Marantz EU HQ used the yet-to-be-launched Q Acoustics concept 500s
http://www.stereophile.com/content/first-look-marantzs-reference-10-series-dms-euro-hq#DQhIDbJJdiGDrCgw.97
The same website distinctly states it uses a digital to analogue converter. There is a digital and analogue stage. DSD is NOT analogue.
Quote from Marantz:
'However, the SA-10 takes things further – just as it features an all-new disc transport mechanism, so the digital to analogue conversion has also been subject to a radical rethink, taking full advantage of the 1-bit conversion technology found in past flagship Marantz players, and incorporating brand-new filtering and upconversion to take advantage of this simple, but elegant solution.'
'The MMM-Stream section[/b] of the process replaces the oversampling filters normally used in digital to analogue conversion, and allows the implementation of the Marantz Musical Mastering filtering. These filters – one providing a slow roll-off and very short impulse response, the other offering the option of a medium roll-off with short pre-ringing and longer post-ringing – are essentially the same as those found in the Marantz SA-11 disc player and NA-11 network music player, but here they’re implemented at a much higher oversampling rate, thanks to that upconversion to DSD11.2.
In fact, two system clocks[/b] are used, to ensure the most accurate upconversion of the incoming signal, whether its from disc or the digital inputs: the 44.1kHz of CD, and its multiples – 88.2kHz, 176.4kHz and so on – are upsampled to 11.2896kHz, while 48kHz and its multiples are taken up to 12.288kHz. This is done for maximum precision, and to avoid any need for sample rate conversion of the kind were the system to have to convert, say, 192kHz audio to DSD11.2.
In addition, all of this conversion is now done in Digital Signal Processing with 32-bit floating-point precision, rather than the 24-bit integer method used in such systems in the past.'[/b]
'[/b]Combining this with the reduction to a 1-bit signal straight after the oversampling filter and Sigma Delta Modulation allows a pure DSD-standard signal to be passed to the conversion section in the form of a very high-frequency stream of pulses, requiring only a very high-quality low-pass filter to remove all the superfluous high frequencies and pass the purest possible audio to the player’s output stage.'