One thing I will say about this thread is that it's helped me to think through the concepts.
Let's put aside audibility for a moment. Why would we do that? For one, there seems to be much more ready acceptance amongst members of this site that the pursuit of engineering excellence is worthwhile, even for it's own sake. Look at the fervour for ever better SINAD on dacs.
Also let's put aside our reservations about DRM and commercial motivation.
Would we not therefore give some credence to a system which, if proven, is measurably closer to perfect analog reconstruction?
From a pure theory perspective, there is an issue with shannon/nyquist sampling in that the sin(x)/x function requires an infinite extent in order to perfectly reconstruct the analog waveform. In practice this means truncating the time extent of the impulse filter. I'm quoting from "Modern Sampling: A Tutorial" by Jamie Angus (which was linked to by a member earlie) here, which I've attached again along with the Stuart paper for easy reference.
1. The filter no longer has an infinite rate of cut-off and thus needs a guard band between the upper frequency of the continuous signal and the lowest frequency of the first alias.
2. More subtly, because the impulse response is now finite in extent, it is impossible to realize a stop-band response of zero (-∞ dB of attenuation) over the whole frequency range of the stop band.
3. In fact, unless the stop band achieves infinite attenuation at infinite frequency, there will always be some alias components present even if the sampling frequency is increased to infinity.
4. The truncated sin(x)/x functions no longer add up to a constant value when the sampled continuous signal is constant. This means that there is some difference between the reconstructed signal and the original signal.
5. The truncated sin(x)/x functions are no longer orthogonal for a time shift equal to multiples of a sample period. This means that the samples are no longer projected properly onto a sampleable space and therefore the samples will have leakage into each other (alias distortion), even if the continuous signal was white noise.
One possible way to overcome this issue is to use a bi-orthogonal approach using β-splines, which can allow for an exponential fall off in impulse response rather than a linear one. By bi-orthogonal the authors mean the effects of the ADC filter being reversed at output with the DAC filter.
This seems to be part of what MQA attempt to do. Now, it seems to me, putting aside the considerations relating to DRM, commercial motive, and audibility, that there is nothing inherently wrong with seeking to pursue a greater degree of engineering excellence in sampling. Especially if we have the means to do so.