Ok, can you offer a description of the filter effects on your players then?
Firstly, what an antialiasing filter is, and why it's there:
It should affect all DAPs both in DAP and DAC mode. The digital to analog converter (DAC) reconstructs the audio signal from the digital signals given (by the audio file being played by the player, or the connected device in the case of DAC mode). At the end of this reconstruction, components of the output signal that are higher than half the sample rate must be discarded (filtered out). In an attempt to simply the explanation: those are spurious signals created by the converter that do not appear in the original signal that was digitized. And no, this is not a flaw of the DAC used in a particular player: all DACs work like this.
The reconstruction lowpass filter is responsible for filtering out these spurious signals. The steep slope filter attempts to preserve as much of the digitized signal in the output as possible while filtering out all of the spurious signal; this is the "correct" filter, but if you play artificially generated square waves / Kronecker impulses through the system, what comes out at the output (as measured by an oscilloscope) is square waves that have extra waves of overshoot before and after each step transition (google "Gibbs phenomenon").
People misinterpret these oscilloscope outputs to mean that the "correct" filter produces audible ringing artifacts, when in fact these "artifacts" occur outside of the audible range and in any case only occurs with artificial square wave / Kronecker impulse input signals, in which case those are the mathematically correct output.
Nevertheless, a "gentle slope" filter may now be chosen, which will eliminate the ringing "artifacts" in the square waves. It does almost as good a job at eliminating the spurious output signals, but also attenuates some of the high frequencies--it starts cutting off signal in a "gentle" slope between the high audible frequencies and the top frequency that can be digitized (at half the sample rate). The possibly audible effect of this is more "rounded" highs--somewhat comparable to how high frequencies are attenuated by the air between loudspeakers and your ears, except you can now apply this to headphones.
Note however that e.g. Chord prides itself on having the longest AA filters, the sharpest cutoffs and hence "ringing" that pretty much goes on forever.
Now for the effects of different AA filters
--this is all assuming you can hear at or above the cutoff frequency of the sampling rate (22.05kHz for 44.1kHz sampled audio, 24kHz for 48kHz sampled audio, 48kHz for 96kHz sampled audio, etc.). If the frequencies at the cutoff are neither heard nor have an effect on you, none of this matters.
There are slow rolloffs vs fast rolloffs, linear phase filters that put half the ringing before the main impulse vs minimum phase filters that put all of it after the main impulse. Slow vs fast and linear vs minimum phase are the two major characteristics of these filters and each filter name should describe both characteristics to be complete. e.g. slow minimum phase filter, fast linear phase filter
Slower rolloffs take out some of the top end. The slower the rolloff the more so.
The short-delay filters put any ringing caused by antialiasing at the back end of the impulse, which theoretically masks the ringing better.
The rationale for using each filter may be as follows:
Sharp rolloff: your high frequency hearing doesn't go up to the Nyquist frequency (e.g. 24000Hz in the case of 48kHz sample rate
(there is no indication that anyone's hearing does, aside from kids / babies probably too young to understand music in the first place)) so you choose to use a sharp rolloff filter. This way you get flat response up to the limits of your hearing and no audible ringing (only inaudible ringing at the cutoff frequency which is above your hearing frequency limit)
Slow (or super slow) rolloff: your high frequency hearing does go up to and above 20000Hz so much that you do hear ringing at those frequencies
(more likely at 44.1kHz sampling rate than anything else, albeit still not very likely at all); besides, modern pop music is mastered too brightly for your tastes. You kill two birds with one stone, the slow rolloff filter: it makes ringing less noticeable and the highest frequencies less prominent.
Linear phase filters (i.e. the ones not labelled "short delay"): you believe preserving the phase relationships of instrumental sounds up to the highest frequencies is beneficial to correct soundstage reconstruction.
Minimum phase filters (i.e. the ones labelled "short delay"): you hear ringing with the "not short delay" version of the same filter and want to make it less obvious without switching to a slower rolloff filter. Or, you really have use for a shorter delay (playing twitch video games on your R8???)
That said, don't be surprised if you don't hear a difference; the differences are subtle at best. And remember, at e.g. 192kHz sample rate, all this is taking place at e.g 90-96kHz rather than 20-22.05kHz.
