Anytime you raise the sample rate you need a reconstruction filter to keep the images of the input from polluting the output with aliasing.
Anytime you lower the sample rate you need an antialiasing filter to keep the frequencies above the Nyquist frequency of the input from aliasing in the output.
When these are done digitally an obvious optimization is to upsample, use a combination filter which filters out everything above 1/2 of the lowest of the input sample rate and the output sample rate, and then downsample. In this case the upsampling and downsampling consist only of inserting zeros and deleting samples respectively.
Tho you may think of that digital filter as being there to get rid of the frequencies that might end up aliasing in the outputs, but that turns out to be exactly the same as a interpolation filter that “smooths” out the input filling in the zeros with the correct samples from the input. That information is available since any other signal would have frequencies above 1/2 of the input frequency and would therefor alias…
In the popular press the technical terms are often mixed up, but some text books also use different nomenclature than others. But there’s no real ambiguity, you need a filter to avoid aliasing/reconstruction images, to interpolate, to change sample rates, etc. The filter needs to filter out everything below 1/2 of the (input or output) sample rate.
No one is claiming that DSDing PCM is required to make PCM sound better, but simple physics says that achieving the component precision to do a multibit DAC is much harder than achieving the component precision for a single bit DAC. Also a single bit DAC is inherently linear and hence has no problems with missed codes, differential non-linearity, bigger errors when higher order bits change, etc. (Note that the chip that the Zanden dac uses achieves the component precision needed with DEM, i.e. by having a lot of sub DACs which aren’t quite as good, and then using them randomly to average out their precision problems.)
I implemented a DSD DAC and threw in PCM to DSD conversion as almost an afterthought (after all that’s only a software problem), but a passive output filter is all that’s needed for a DSD dac, so by converting PCM to DSD you can do a PCM dac with only a passive output fitler and some software.
As it turns out there’s so much more in the original PCM recordings than most of us knew. This isn’t new information that PCM to DSD created, it’s just information that most PCM dacs cover up with perhaps their non-linearities, perhaps their less than ideal power supplies or power supply bypassing, perhaps their less than ideal clocking, perhaps short cuts in the digital filters being used, etc. Also most PCM dacs (by necessity of using a DAC in a chip) suffer from having their digital supplies right next to their analog supplies at the DAC chip and their clocking right next to their power supplies, etc. all of which analog engineers are taught to avoid.