I've read the SACD is the same as the DVD-A so all I can think of is that you don't like the 5.1 to 2.0 downmix or you just dislike the recording itself. Will compare the downmix to the CD-layer version.

What version do you recommend, if any?

Personally I find that album overrated for mastering quality.  I never user it for testing.

Exactly.  I do use "Walk Between the Raindrops" sometimes when I want to hear something with shitty compressed sound.

I like Donald Fagen a lot, but that album just pissed me off.

Anti-aliasing filters are used in ADCs, not DACs [correction by ultrabike: DACs can implement decimation filters which are a form of digital anti-aliasing].

An anti-aliasing filter is a steep lowpass filter which tries to eliminate frequencies above Nyquist, which is half the sample rate.
For instance, if you are converting an analog signal to digital at 48kHz, your ADC should have a lowpass that ideally rejects everything above 24kHz. This is because any frequency above the halfway mark (24kHz) could be falsely interpreted by the DAC stage as a corresponding lower frequency.

Look at image d in this figure:


The analog signal's frequency is 0.95 of the sample rate...in the case of 48kHz, we have a 45.6kHz sine wave. The black squares are the actual data points captured by the ADC stage. Now, when a DAC tries to reconstruct the sine wave, you can see that the data points form a nice super-imposed sine at 0.05 of the sample rate or 2.4kHz. There's just no way for the DAC to reconstruct anything above the Nyquist frequency, so we must get rid of those frequencies. You can imagine how nasty this could sound with actual music instead of sine waves. Ultrasonic harmonics/noise/garbage being interpreted as lower frequencies, mixed into the audible spectrum. Nasty.

These filters can't be theoretically perfect in practice because we're still dealing with the limitations of analog. The filter cannot be infinitely steep, as would be most ideal. The best ADCs/DACs typically are those with painstakingly-designed analog stages.

There are software tricks that can be done to reduce the complexity of the analog anti-aliasing filters (on the ADC side) and reconstruction filters (on the DAC side) required, BUT you still need great analog design to get quality results from digital...whether converting from analog or to analog. The design and quality of these analog filters is one big reason why DACs and ADCs can sound so different. It's not just ones and zeroes.

Anti-aliasing filters are used in ADCs, not DACs.

Depends...

In the case of an up-sampling DAC, if the up-convert rate is an integer multiple of the original rate, then an anti-imaging (or interpolation) filter is used. However, if the up-convert rate is not an integer multiple of the original rate then down-conversion might be required as well. Down-conversion might imply a decimation filter, which is kind of a digital anti-aliasing filter...

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On a different note, I find interesting that the ringing in the test track was applied in the same (roughly) region that my KSC75 and HD558 exhibit ringing... around 4 and 5 kHz... Maybe something like a modded HD800 with less nasties in that range will be more revealing in an ABX.

I thought a low-pass would only work in this application if the DAC could somehow generate a true impulse train (which is impossible in the real-world since the impulses are infinitely narrow).
So, if your DAC uses zero-order hold (sounds like what you're describing by "voltage steps"), then you need a filter that looks like this:

OR a multirate filter (downsampling, upsampling).

But a low-pass will not reconstruct the effects of zero-order hold. Am I wrong here?

I think you are correct. Those are compensation filters which can be implemented in the digital domain...
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So... Did you give the audio files comparo a shot?