Best Cable Option: Streamer to DAC

Nyquist applies only to continuous waves, so intermittent (music) 22kHz signal would have sidebands that may encroach on the 1/2 sampling frequency limit. It is remotely possible, but amplitudes of folded signals will be extremely small (as are 22kHz harmonics). 96kHz sampling would fix it for sure, but some people claim they need 192kHz. Same goes for number of bits. 16 bits means about 96dB dynamic range requiring 96dB of system total S/N to hear it. 24dB provides about 144dB dynamic range. In order to hear lowest bits you need 144dB S/N. Nobody has that, but you want 32 bits (192dB), claiming it sounds better?
On one hand it is insanity, but on the other - if it helps (placebo effect) it helps, and there is nothing wrong with it. ;)
Other than injected noise (computer cable is connected) asyn USB should be better, since it delivers only data (timing is not involved). S/Pdif has timing encoded with data and that can produce timing jitter (DAC clock is based on it). There are many factors involved in selecting best interface (Ethernet, USB, Toslink, Coax) and nobody can know for sure what is the best solution in your home. Trying at home and comparing seems to be the best course of action.

@cleeds It is likely that harmonics of 22kHz come from percussion instruments, that are not continuous by nature.  100Hz (drum roll) will give you (modulated) 22kHz with bunch of sidebands spaced 100Hz apart.  First two sidebands (21.9kHz and 22kHz) will have already very small amplitude and the rest of them can be ignored.  Sidebands appear because signal is modulated (not continuous).  The danger here is that anything above half of sampling frequency will, in D/A process, fold into 0Hz and up.  With 44kHz sampling 22.1kHz will become 100Hz signal and there is no way to remove it.  We might not be able to hear 22.1kHz, but will definitely hear 100Hz.  I stated that even with 44.1kHz it is very remote possibility and folded-over frequencies will have extremely small amplitudes.  96kHz would eliminate any possibility of this happening while 192kHz is a waste of space on HD, IMHO.  I have few plain redbook CDs with such breathtaking quality, that makes me believe that 16/44.1 format is not the limiting factor.  192kHz sampling or 24bit resolution might be important in studio during mixing, to avoid loss of quality, but final product in 16/44.1 is fine with me.

 

@cleeds instead "First two sidebands (21.9kHz and 22kHz) will have already very small amplitude and the rest of them can be ignored."
it should say:
"First two sidebands (21.9kHz and 22.1kHz) will have already very small amplitude and the rest of them can be ignored."

papafrgog,  rate of incoming signal and rate of D/A conversion have to be synchronized.  With async USB computer sends bunch of samples (frames) at certain rate (usually 1 kHz).  DAC receives such frame and places it into buffer.  Dac signals back to computer when buffer is over or underflowed and computer adjusts the size of next frame.  That way rate of D/A conversion is based on DACs internal stable clock only and no data sample is lost.

Any percussion instrument is an "continuous"

No, it is not. If you record drum roll it will show individual bursts with silence in between.

The signal is analog, it is continuous.

Analog and continuous are two different terms. When you strike drum once signal is not continuous. There is a silence before and after. Any modulated frequency will show on FFT as a root frequency and sidebands. It applies not only to amplitude modulation, but to any modulation, including time jitter of digital signal.

Anything exceeding half the sampling rate is filtered out as part of the A-D conversion process. It doesn’t "fold" into anything. You can easily prove this with measurements.

There are analog filters before A/D process but they cannot be very sharp. By definition they have to be even group delay (Bessel), and those are very hard to make in analog domain. Bessel filter characteristic show practically the same, weak attenuation within 2x Fc, no matter how many poles you use. The whole idea of single bit converters (Delta-Sigma) was to avoid sharp filters by pushing quantization noise higher. In any A/D and D/A process there is ALWAYS something that folds over. The only issue is the scale (amplitude) and all I stated is that Nyquist applies only to non-interrupted (continuous) frequencies, but artifacts of this violation are likely non-audible with redbook CD and definitely non-audible with 96kHz sampling.

Again, any signal that is amplitude modulated will have sidebands.  Organ note played will show single frequency on FFT, but interrupted 10 times a second will show root frequency and sidebands spaced 10Hz apart.   

I have to agree with you, that filters in front of A/D will be essentially perfect.  I forgot that they record at 192kHz (not 44.1kHz) and 10 pole Bessel will have, at 10x Fc, attenuation above 100dB (below one bit of resolution).