@mikhailark I don’t disagree with you about the importance of the “analog part” or the power supply(ies). The question I have is this: assuming a modern high end DAC with sophisticated power supply and presumably much higher quality “analog parts”, like say a Mola Mola Tambaqui, and measures essentially the same as several hundred dollar Topping D50III, how can they sound different? Wouldn’t we expect all those great “analog parts” in the Mola Mola to not only affect the sound, but also affect the measurements? Are the measurements or are our ears wrong?
I purposely pick two devices at opposite ends of the cost spectrum to ask this question because any difference in perceived sound cannot be from some artifact or coloration that tricks us into believing it sounds more musical. Either the marketing and our biases related to value are coloring our judgement, or there is more to accurate and enjoyable sound reproduction than can currently be measured on the bench top.
This may be part of it. From an email exchange between Mola Mola’s Bruno Putzeys and Stereophile’s Herb Reichert regarding the design of the Tambaqui DAC:
"Initially, I looked at using only a single, high-current switch to convert the PWM signal, but it soon struck me that running a number of them in a time-staggered fashion would allow me to remove most of the PWM carrier right away and so reduce noise. That was the core of the design. The remainder of the project was being completely anal about all the other stages of the converter: digital filtering, clocking, and analogue-output filtering.
"Of those, only the digital filter needed to be optimized by ear. It’s pretty obvious that a more stable clock is more ideal, and an output filter with lower noise and distortion is also more ideal. But there’s no ideal upsampling filter, a priori: The ear is not a spectrum analyzer. You need to listen to original high-rez files, filter them down, upsample them again, and then hear which kind of filter chain leaves the smallest sonic fingerprint. That is to say, how do you get from high-rez to (eg) "Red Book" and back whilst getting the smallest possible audible change? And then it turns out that a lot of filters out there sound really impressive, but only because they’re heavily euphonic—not because they’re sonically neutral....To make matters worse, the optimum design differs for different sampling rates....”
So perhaps there are elements of sound reproduction related to filtering and signal reconstruction that are as much art as they are engineering and physics, and sometimes what sounds more natural to our ears - especially in reconstructing a 16/44.1 signal can’t be deciphered by measurements, or in some cases looks like errors or spurious information? But this is still in the realm of digital processing, and has little to do with the analog section.
Putzey’s email goes on to say:
"Clocking was addressed using a very stable, non-adjustable crystal oscillator—adjustable ones are quite noisy—and synchronizing the signal using a homegrown asynchronous sample-rate converter that forms part of the digital filter. How that was done is a story in its own right, but it might take us a bit far [afield]. Same for the analogue output filter stage, which is also rather original in its conception. So, as much as you’d like to know what the magic ingredient is, I can only tell you that it’s about getting all the parts right, not just individually but as a system. It’s not sexy, but then real engineering rarely is."
Also on the analog and power supply side of things, it is possible that a device with a cheap power supply is injecting noise into your system and reducing the overall performance. Why some people immediately replace wall warts typically found with less expensive electronics with more substantial linear supplies. Also why some of these less expensive device may perform well when measured in isolation, but sound less pleasant when placed in a high end system.
kn