Забавные выдержки из ответов Rob Watts (Chord) на вопросы, связанные с DSD и PCM-DSD конвертацией (наверняка многим уже известные, но я впервые прочитал) 
По поводу использования компьютеров для апсемплинга и конвертации (HQP) с продуктами CHORD
Do NOT use your computer to up-sample or change the data when you use one of my DAC’s.
Computers are poor devices to use for manipulating data in real time as they are concurrent serial devices - everything has to go through one to 8 processors in sequence. With hardware and FPGA’s you do not need to do that, you can do thousands of operations in parallel. Dave has 166 DSP cores with each core being able to do one FIR tap in one clock cycle. That is incredibly powerful processing power way more powerful than a PC.
But its not just about raw processing power but the algorithm for the filter. The WTA filter is the only algorithm that has been designed to reduce timing of transients errors, and the only one that has been optimised by thousands of listening tests.
So the long and the short is don’t let the source mess with the signal (except perhaps with a good EQ program) and let Mojo (or DAVE) deal with the original data, as Mojo (or DAVE) is way more capable
По поводу недостатков DSD по сравнению с PCM
DSD as a format has major problems with it; in particular it has two major and serious flaws:
1. Timing. The noise shapers used with DSD have severe timing errors. You can see this easily using Verilog simulations. If you use a step change transient (op is zero, then goes high) with a large signal, then do the same with a small signal, then you get major differences in the analogue output - the large signal has no delay, the small signal has a much larger delay. This is simply due to the noise shaper requiring time for the internal integrators to respond to the error. This amplitude related timing error is of the order of micro seconds and is very audible. Whenever there is a timing inaccuracy, the brain has problems making sense of the sound, and perceives the timing error has a softness to the transient; in short timing errors screw up the ability to hear the starting and stopping of notes.
2. Small signal accuracy. Noise shapers have problems with very small signals in that the 64 times 1 bit output (DSD 64) does not have enough innate resolution to accurately resolve small signals. What happens when small signals are not properly reproduced? You get a big degradation in the ability to perceive depth information, and this makes the sound flat with no layering of instruments in space. Now there is no limit to how accurate the noise shaper needs to be; with the noise shaper that is with Mojo I have 1000 times more small signal resolution than conventional DAC’s - and against DSD 64 its 10,000 times more resolving power. This is why some many users have reported that Mojo has so much better space and sounds more 3D with better layering - and its mostly down to the resolving power of the pulse array noise shaper. This problem of depth perception is unlimited in the sense that to perfectly reproduce depth you need no limit to the resolving power of the noise shaper.
So if you take a PCM signal and convert it to DSD you hear two problems - a softness to the sound, as you can no longer perceive the starting and stopping of notes; and a very flat sound-stage with no layering as the small signals are not reproduced accurately enough, so the brain can’t use the very small signals that are used to give depth perception.
So to conclude; yes I agree, DSD is fundamentally flawed, and unlike PCM where the DAC is the fundamental limit, its in the format itself. And it is mostly limited by the format.
Ну вот по поводу очень плоской сцены - не вполне согласен. Скорее другие у меня впечатления.
А смягчение - скорее да. Вполне возможно, что именно в этом и кроется проблема - “плохо звучит классическая музыка с большими составами”.
