S/PDIF - Digital Transfers - Resampling - Digital Audio and Jitter

When samples are not output at their correct time relative to other samples,
you will experience clock jitter and the inevitable distortion due to the
clocks not being in sync.

Since S/PDIF does not carry a separate clock signal, the receiving device will
attempt to synchronize the samples to its own clock from the digital input
usually using a Phase Locked Loop or PLL. In some instances it will try to
rebuffer or reclock the signal or use sample rate conversion. The PLL method
will certainly be subject to jitter on playback, dependent on the quality of
the signal at the input. This is why it is important to make sure that you 
are using high quality S/PDIF cables for digital transfers. 

A low quality cable will only make the audio sound worse. This will only
affect the monitoring. If you record the signal and play it back, there will 
be no change from the original (again, unless you are using low quality
cabling or there is a transfer error of some sort). 

-  If, for example, the clock rate of the digital stream and the playback unit
   differ (44.1k and 48k, for instance), the playback unit has no choice but
   to sample rate convert. 

-  If the sample rates are the same, the playback unit may use sample rate 
   conversion to oversample the input and pick the samples that "line up" 
   with its own clock, or it may buffer the incoming stream and reclock it
   for output.
 
   In this case, either method will not be subject to jitter, since the D/A 
   converter is using its own local clock. It is important to remember that 
   resampling (sample rate conversion) techniques actually change the digital
   stream before converting it to analog, whereas buffering does not.
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