I forgot to add that delta-sigma modulation is required on top of that (shame on me).
In case anyone is interested: sound/music consists of multiple frequencies which overlap and form a more or less chaotic frequency wave. In the real world, this frequency mixture is transfered over the air by pushing/pulling the molecules of the air (you can feel this if you stand in front of a kicker bass blasting at a low frequency and making your hairs fly).
DSD now uses this idea of "air density" (or air flow, or air being pushed/pulled around) for data encoding into a bit stream. What DSD does is to send a constant stream of 0 and 1, with the encoding that a 1 pushes the virtual air into your direction (so to speak), while a 0 pulls it away from you.
Here is a picture with the same data ( a sinus wave) encoded in two ways, one with the DSD data, the other the typical analog sinus wave.
If a cable or a power supply wants to enhance the sound quality ("more clear bass"), this 0 1 0 1 0 bit stream has first to be translated back into data which uses frequency/time as encoding and not a pulse/density, then change the balance of the different frequencies to enhance some sound ranges and last but not least use delta-sigma modulation to translate the information back into the stream the DAC is expecting. Or if you look at the linked picture: You first need to translate these 0 1 0 1 bit combination to get the sinus wave. Then you can play around with that wave, but to make your DAC happy you then need to translate this back into the 0 1 01 stream.
The same applies if you use a cheaper cable in the oposite direction, and that my friends is a bit too much intelligence for such a cable.
On top there is zero chance that any noisy frequency coming from the power supply or other source could modify this data stream (and switches exactly the 0 and 1 in the correct position of the data stream) in such a specific way to achieve any sound enhancement by accident, while bypassing the reqired transcodings The frequency used to generate such a bit stream is by magnitudes higher (2.8 MHz) than any frequency around (e.g. 220V supply with 50/60 Hz). All you can do is to erase the whole bit stream, and then the DAC stops acepting data. You can't use something which can change data from 0 to 1 or the other way around working at a frequency of 60 Hz to modify a data stream using 2.8 MHz.
Last but not least it's impossible to get a "darker, less noisy background". Silence in DSD language means a 1 0 1 0 1 0 1 0 bit stream (check the linked picture, look at the area where the sinus wave is near zero amplitude) and how the corresponding bit stream looks like. So any "thing" which wants to make such a stream less noisy needs to be able to identify the areas of the data stream where 0 and 1 need to be adjusted in a way to form a better 0 1 0 1 0 1 alternating data stream, and this stream pushes out at 2.8 MHz, so that thing needs to analyse and "enhance" 2.8 mio bits/sec. So what people think by applying rules of the analog world doesn't work in the DSD world. If you achieve to generate a low noise constant electrical signal (using an audiophile LSP instead of a noisy cheap power supply) this either results in a constant stream of 00000000 or 111111, which is exactly the opposite of what DSD uses to expresss silenece (01010101010101).
Over and out. Don't let this any audiophile crack see, becaus all that counts are 0 and 1