Best Cable Option: Streamer to DAC


I was recently told that the inherent limitation of SPDIF connection is PCM 192Hz. I didn't know that. Many new streamers spec 384KHz and I am also told that to achieve higher sample rates (and presumably the full capabilities of the new units) I should use USB rather than SPDIF.  So it made me wonder what actually is the best connection between streamer and DAC:  USB, COAX BNC-SPDIF, AES/EBU or something else?   From a practical standpoint, is there any audible difference from the higher sample rates?  If so, my system should be able to reproduce it.  I'm just looking for help, not trying to start any arguments on here.
papafrgog
This is great information and I appreciate the thoughtful responses. I have experimented with my system connections, but never tried USB, thinking it would be inherently noisy. Noted definite improvement of SPDIF over Toslink. I use Transparent Reference or Ref XL cabling exclusively. I am getting Aesthetix Romulus DAC upgraded from Signature to Eclipse at this moment so I want to extract everything it can offer when I get it back. 
Intrigued by the idea of DDC between streamer & DAC. Will be the subject of more research. The obvious consensus is USB, and my takeaway is to get one that is well made and shielded. The notion that SPDIF has timing encoded with data was new to me vs USB only data. Never really thought of it in those terms. 
This is how the forum is supposed to work. Thank you all
99% of the back catalogue is redbook. Presently the best connection is mainly dependent on the quality of the interface implementation between streamer and DAC. Increasingly, the engineers spend their efforts predominantly on USB. There are however still instances where specific proprietary solutuons (e.g. I2S) outperform.
@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.

 
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.
@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."