In designing sampled-time systems, the variables that we need to juggle are signal accuracy (or fidelity) and various kinds of system cost (dollar cost, power consumption, size, etc.).
True statement. It applies to all engineering in general, whether electronic or mechanical, even software engineering. Quality generally costs more. Even when you do find some elegant solution that raises quality while lowering cost, chances are it took you longer to figure it out, which costs money.
Measured purely from a sample rate perspective, increasing the signal sample rate
will always increase the signal fidelity.
True again. However, the operative phrase is "measured purely from a sample rate perspective". We're not talking about digital oscilloscopes or RADAR here, but audio, which is unambiguously defined by the mechanical limitations of human hearing. The writer is using "signal fidelity" as a general concept, irrespective of any specific design goals. In the case of audio, "fidelity" is more specific, and means how closely the recorded signal sounds like the original. Ears are the final arbiter, not laboratory test equipment (which will always find shortcomings, even those that are inaudible).
So while the statement is true in a broad sense, it is somewhat misleading to apply it to the discussion at hand.
It will often decrease the cost of any analog antialiasing and reconstruction filters, but it will always increase the cost of the system digital hardware, which will not only have to do it’s computations faster, but which will need to operate on more data.
True again. I only take issue with "always". All digital components have upper speed limits, so the designer will choose them with the project's design requirements in mind. Usually, the selected components will be capable of performing far beyond the requirements. In purely digital circuitry, increasing speed and capacity is often trivial and incurs little or no additional cost, as long as the designer hasn't painted himself into a corner with bad early decisions.
Also, the cost of analog anti-aliasing and reconstruction filters might be a factor in some equipment, but it's trivial in audio devices. We're talking 5-cent capacitors here.
Where design compromises most often come into play are in circuits that have an
analog component, namely oscillators and sample-and-hold circuits.
With oscillators, it's mainly a cost tradeoff. Fortunately, at the frequencies used in oversampled ADCs, designing an accurate and stable oscillator is neither difficult nor expensive. You could shave a few pennies off at the expense of accuracy, but I can't imagine why anyone would bother given how crucial it is to the device's performance. Plus you'll normally only have one oscillator, even in a multi-channel interface, so it's not the most cost-effective place to save money anyway.
It's the sample-and-hold circuit that's going to have inherent tradeoffs regardless of cost, because its accuracy is not ever going to be consistent across all sampling frequencies. In this particular part of the ADC, the designer has no choice but to pick a target sample rate to optimize the S/H for. If his market is primarily professional studio, he'll assume 96KHz - and have to accept
slightly less-than-optimal performance at 44.1KHz. This is why some interfaces perform better at one rate over the other. It's also why Dan Lavry refuses to sell a 192KHz interface, and why the ability to sample at 192KHz should not be taken as a predictor of how well a device will perform at more commonly-used rates.
(BTW, I have designed sampling circuits. Not for audio, but for industrial controls. Same principles, different requirements.)