When samples are not output at their correct time relative to other samples, we have clocatz jitter and the associated distortion it causes. Fortunately, the current state of the art is very good for stable clocking, so this is not a problem for CD players and other digital audio units. And since the output from the recording media (CD, or DAT, for instance) is buffered and servo-controlled, transport variations are completely isolated from the digital audio output clocking.
Clocking external sources
Clock jitter can arise when we combine multiple units, though. When each unit runs on its own clock, compensating for small differences between the clocks can cause output errors. For instance, even if both clocks are at exactly the same frequency, they will almost certainly not be in phase.
For example, consider connecting the digital output of your computer-based digital recording system to a DAT recorder, and monitoring the analog output of the DAT unit. Because the digital output (S/PDIF or AES/EBU) doesn’t carry a separate clock signal, the DAT unit must output the audio using its own clock.
Since the DAT player can’t synchronize its clock to that of the source, it has to either derive a clock signal from the digital input (using a Phase Locked Loop—PLL), or make the digital input march to its own clock (buffering and reclocking, or sample rate conversion). The PLL method will certainly be subject to jitter on playback, dependent on the quality of the digital signal at the input. In other words, poor cables would make the audio sound worse! It’s important to note that this will only affect monitoring; if you record the signal and play it back, there will be no change from the original (barring serious problems with the cabling or other transfer factors). This because the recorder will store the correct sample values, despite jitter, then reclock the digital stream on playback.
If the clock rate of the input digital stream and the playback unit differ (44.1 KHz and 48 KHz, for instance), the playback unit has no choice but to sample rate convert. If they are the same, the playback unit may use sample rate conversion to oversample the input, then pick the samples that “line up” with its own clock, or it may simply buffer the incoming digital stream and reclock it for output. Either method will not be subject to jitter, since the D/A convertor is using its own local clock.
Note that the resampling (sample rate conversion) techniques actually change the digital stream before converting it to analog, whereas buffering does not. This is a particularly important distinction when making digital copies and transfers.
Be sure to check out Bob Katz’s web article on the subject for a more detailed look.