No, there are two fundamental misunderstandings in your comment.
You posit an opposition between "[arrays of] point samples with a convolution" and "[arrays of] exceedingly crisp squares". This is incorrect. If the kernel you convolve your array of point samples with is an exceedingly crisp square, the result is an array of exceedingly crisp squares. Convolution with a square is just a special case. Smith explains this, perhaps not very clearly, in Fig. 3, p. 5.
Conceptualizing this as a convolution allows you not only to handle the special case you're saying is "a reasonable first-order approximation," but also to analyze what effects that display or sensor convolution is having on your image, and possibly to correct them. Moreover, it can handle some other kinds of image degradation as well that go beyond that first-order approximation, such as bokeh, diffraction, defocus, and nonuniformity within pixels, as long as it's the same for every pixel.
That's how you find out what those "further adjustments to avoid antialiasing artifacts, desired sharpness, and so forth" are.
And it's how you can go about analyzing what happens to the image when you, for example, resample to a different sampling grid, as Smith explains on p. 4. If you analyze different linear resampling algorithms using the theory of convolutions, you can predict precisely how good the results will be; if you do not use this theory, you will be puzzled and will not understand your results at all. Believe me, I know; I wandered in that desert for years. But you don't have to. Learn from my mistakes.
Second, audio samples are frequently handled exactly the same way as pixels in this sense, using a "zero-order hold". If you oversample by taking 64 consecutive samples from your ADC and summing them, or set the PWM duty cycle on your Arduino to a new value every 125 μs and leave it there until the next sample, or shift a new sample into the shift register connected to your R-2R DAC and leave it there until the next sample, you're doing exactly the same thing as trying to draw a little square for each pixel—just in one temporal dimension instead of two spatial dimensions. A zero-order hold is a perfectly reasonable thing to do with audio samples for both input and output, although it's not the only option, and it can produce audible artifacts. The math for analyzing this—and correcting it, if that's what you want—is exactly the same as the analogous math for displays or camera sensors made out of little squares.
Far from being "far more misleading", the analogy between pixels and audio samples is deeply insightful and unlocks a wealth of mathematical tools that can be applied in both domains, as well as, for example, to linear control theory and GPS position estimation. The results this analogy leads you to are profound truths, not falsehoods as you seem to think. And some of them are very beautiful.
None of this has changed in 25 years (except the detail that now we have LCDs with little rectangles on them to complement the little squares of dye-sublimation printers that Smith mentions) and little of it has changed in 50 years.