> I don't believe I'm hearing any of the unwanted aural effects you described.
Sample jitter will be a very subtle effect. It's well into the audiophile 'super ear and no instruments' realm. But it is a real effect, with a basis in engineering fact, unlike much of the 'woo'. It can result in that indefinable 'better', 'crisper', 'more defined' sound you read about in audiophile reviews.
The internal/external clock issue comes down to whether the DAC pulls data from the source, or whether the source pushes data to the DAC. If the DAC pulls data from the source, it can be master of its clock, provided the FIFO buffering in the system is adequate to prevent underflow. If the source pushes data to the DAC since source and destination clocks are not identical, a FIFO buffer will eventually overflow or underflow, unless the DAC does clock recovery. A recovered clock will have more jitter than an independent clock (we have to pull an oscillator to match the input clock).
The hifi world seems to have adopted term 'asynchronous' for the pull method, presumably referring to the relationship between source clock and DAC clock. By synchronous, they mean that the DAC is clocked by the source clock (either directly, or using a recovered clock).
SPDIF cannot support data being pulled by the DAC, so it relies on low jitter on the SPDIF signal, and clock recovery in the DAC.
The VoIP phone protocol has this clock mismatch problem, but gets over it by allowing frame drop or repeat, which would be unacceptable in a hifi streaming system.
You might like to see a rather thorough technical review of your DAC:
http://kenrockwell.com/audio/cambridge/dacmagic-plus.htm