Why DSD is never over-sampled like PCM in a modern DAC?

Audio Gear Talk
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This is puzzling question I always wanting to ask… DSD is always converted to analogue at its native sampling frequency. In PCM, 44.1kHz is always over-sampled to 352.8kHz inside the DAC then convert to analogue, this provided excellent signal to noise ratio across the audio range.

DSD64 noise-shaping is only effective up to 22.05kHz, after that ultra-sonic noise will start to appear. If DSD64 is over-sampled to DSD256 or even DSD512 inside a DAC, the ultra-sonic noise will be shifted even further away by a few octaves! This means, DSD64 recording is all it needs to get a good signal to noise ratio if it is oversampled in a DAC then to use DSD128 and DSD256 recordings (file size becomes too large).

Is there any reason why DSD is never over-sampled or up-sampled in a modern DAC?

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A naive guess would be lack of processing power on a typical DAC integrated circuit. Upsampling DSD seems to require substantially more CPU resources in Roon and HQP than PCM upsampling. Dedicated FPGAs like the PS Audio DirectStream are an obvious exception (20x DSD upsampling).