A statement of the designer (Mike) in a thread where a lot of stupidity was/is flung around.
Euphonic? I never thought I was producing euphonies. As I have written in my own thread, a true audio pioneer, Mr Peter Walker told me 40 some years ago at the beginning of this road for me that it was my duty to reproduce music, not to alter it. I still have a set of his original Quad Loudspeakers today. Your statement about bypassing the filter puzzles me - if you have a mb Schiit DAC, the filter is always there, yup, no way to get rid of it.
I am also puzzled by your NOS DAC statement. "New-old-stock"? "Non-oversampling"? Precise resistors? Like the automobile priced DACs who construct their own DAC chip equivalents whose fragile linearity varies with farts in the room?
The design philosophy between MB and DS DACs does not differ. the goal of both DACs is to output an analog (and flowing, as in not stair-stepped or noisy) output signal that is as close as possible to the waveform that was described by the samples and is a very close approximation of the analog signal that has been recorded.
Other chips are used and other technical solutions are needed to achieve the same goal. One can play with treble roll-off.
The only difference in design philosophy between DS and MB is the digital upsampling/filter is already present in DS chips and when using MB DAC chips they have to programmed into an IC or 'off the shelve' chips can be used to do the upsampling/filtering before it goes to the DAC chip.
Mike claims he uses the original samples which makes it more accurate. (sales pitch based on subjective findings)
All DACs need analog post filtering.
The MB DAC's have a relative steep analog filter that sits just below the Nyquist of the highest bitrate it can handle and all incoming bit rates that are lower are upsampled to the same (highest) bitrate it can handle.
You can see that analog post filter as a device that 'invents' voltages between the upsampled steps so even if the burrito filter uses the original samples. This post filter simply creates smooth transitions according to the laws of physics, there is no design philosophy used. ALL DACs with proper post filtering will (just in a very brief moment) have an output voltage that is exactly the same as the original samples have described.
So what IF the original sample values are used or not ? The analog post filter does not care one bit and only during a
very brief moment the actual output voltage will ALWAYS be equal to the described values. Regardless if the DAC is MB, DS, Pulse array or whatever.
DS chips (and pulse array) over-sample a LOT more times than 4 times and all analog output values are calculated so that the average value, in a short time frame, is very close to the described/expected output voltage of a post analog filtered output signal.
The upside is that the post filter used can be very simple and easy to make and does not have to be sharp at all. Downside is the noise levels above the audible range are somewhat higher than that of MB.
Where is the 'magic' here and the 'philosophy' in design that has a 'more natural music sound' in it ?
The reality is that 99.9% (gros estimation) of the time the actual output voltage is 'calculated' or determined by an analog post filter and only 0.1% of the time the actual described sample value is present in ALL DAC types.
One can believe in the 'magic' all day long but most of the perceived 'magic' is invented. Not in the DAC but in the mind of the listener who KNOWS what DAC is being used.
Maybe in some cases early treble roll-off sounds pleasant to owners of certain DACs as well. Magic? no...
