reconstruction filter - How does it actually work? I'm trying to form my own understanding on the religious war around using 192kHz as a sampling rate for playback (the Internet seems to have a wealth of material on both sides). I'm struggling to understand how reconstruction filters work.

The Nyquist-Shannon sampling theorem, often cited by the anti-192 camp, basically states that a 44.1kHz sample rate is sufficient to reconstruct without loss a band-limited signal of 20kHz. However, when looking at the Whittaker–Shannon interpolation formula, it seems to me that an ideal reconstruction filter would need to have access to all samples, i.e., all past and all future samples.

I'm not an expert in analog audio, but I doubt that such a device can be built. I would assume that, at best, a delay can be introduced to "wait" for enough future samples to arrive, so that the contribution of unavailable future samples to the current output instant becomes negligible.

Can someone explain how practical reconstruction filters work and what are their tradeoffs? Is there a theoretically stricter limit to the Nyquist-Shannon theorem if only a window of samples is available or if reconstruction latency is unacceptable?
The sampling theorem requires a perfectly bandlimited signal, bandlimited to below twice the sampling frequency.  The problem with this is that only an infinite length signal (e.g. exists before the big bang) can be perfectly bandlimited.  This is from the Fourier theorem regarding any domain with finite support.  

Thus all real-world signals are imperfectly band-limited, limited by the length of the recording session and the duration of the impulse response of the low-pass filtering used before sampling (e.g. imperfect band-limiting).  So you must assume a finite noise floor due to sample aliasing.  Therefore your reconstruction also need be no better than this noise floor.  Therefore you can window the perfect reconstruction formula to some reasonable finite duration.  For audio, this window width can be limited to on the order of a small multiple of the period of the lowest frequency audible, as stuff before and after that can't affect audible pitch perception significantly (other stuff, such as masking and adaptive thresholding, etc., likely dominate).

The advantage of very high sample rates is that the aliased frequencies (due to imperfect pre-sampling filtering and post-sampling reconstruction) are even higher, and far less likely to be picked up by a physical microphone or pass with any significant magnitude (above the thermal noise floor, etc.) through the anti-aliasing filters.  As other answers here detail, this also allows physically realizable filters to be flatter and have more linear phase response in generic 20-20k pass-band, and roll-off more approaching half the sample rate.