@Ghoostknight
Let me outline a couple of steps that you could take to make members take your questions more seriously here. I see you really don't want to do any blind tests so I won't mention that again.
1) First, you would have to establish that the music you are testing with contains significant amount of power below 20Hz. While it's possible that some instruments generate such frequencies (such as pipe organs or very loosely, essentially "faulty" tuned bass/kick drums are the ones that I can think of the top of my head) these would also have to be picked up with a capable mic and then not get filtered by the people making the mix/master. These frequencies can also be generated accidentally like kicking the mic stand or a singer breathing into the mic that can capture such low frequencies. Both the accidental and the delibarate cases can happen sometimes but that's not something that should happen on every music that contains bass. In general, that would be a hallmark of a low quality recording and/or no effort put in the mixing and mastering stage.
Since you mentioned that it happens to most of your music, what's more likely is that you are deceived by your real-time spectrum analyzer. All spectrum analyzers have to deal with something called
"spectral leakage". You are very likely viewing a real-time analyzer that must use a short time window (to make the graphics look more "real-time") which further exacerbates this problem. If you are too lazy to watch/understand the video, the point is that just because your analyzers shows crap at (or below) 20Hz does not mean that there's any signal there. The analyzer isn't "inaccurate" really, it's just that it's often misunderstood what the analyzer is actually showing. If you can't set the window size and shape of your analyzer (and don't understand how it relates to the input signal), you can't ever be sure what's shown by it.
Here's a gif of what I'm talking about (click to download and view it). I'm feeding a sweeping pure tone to the analyzer yet there's all kind of "crap" around the main tone. There's seemingly an alarming amount of low frequency content, as if I were feeding more than just 1 sweeping harmonic to the analyzer. Your analyzer is doing the same although it might be less visually apparent if you are using a "bar graph" view. Again, there's no low frequency content in the analyzed signal, it's just the analyzer doesn't quite show what you would expect it to show.
2) Second, in case the existence of the low frequency content is established (again, very unlikely), you would have to show that the filter you are applying is actually supposed to be transparent. Just because you say that your filter doesn't attenuate frequencies above 20Hz and some fancy graphics show you that, that doesn't make this assertion true at all. I'm sorry to bring up some of your other posts but in this case I think they are relevant. I remember you struggled to get bit-perfect output from your browser somehow. I remember that you were searching harmonics at 16kHz in a 8kHz square wave. I remember you didn't even try to anti-alias a 44.1-48kHz sample rate square wave that had the fundamental well into the kHz range. I remember you claimed that an FFT EQ is not the same as a FIR EQ. Now, you show that you don't understand FIR filters either. I just remembered that you think if a speaker that has x% THD at some low frequency in its spec sheet you'll have the same THD once you put the speaker in your room. With all these in mind, I simply don't trust you can actually properly high-pass audio in a way it stays transparent. You could capture the digital output of your player with the filter on and off so we could compare if you actually managed to filter the signal properly.
There's just nothing to talk about if we don't know there's actually some low frequencies in your music and if we don't know you are filtering these out properly.
Even with these out the way, your original question does not make sense at all, although you managed to somewhat clear up on that at least.
A symmetric FIR filter and the typical IIR filter will show the same amount of ringing if you use the same input signal, and the filters have the same magnitude response. The difference comes from how the ringing is distributed: IIR filters only have post ringing while the symmetric FIR equivalent will have have half the ringing as post ringing and half the ringing as pre-ringing. I did some filtering to bass heavy edm tracks and I ended up with RMS differences around -55dB. I can't stress how unlikely that this would make any real world difference. If you can't get such a low number your filter might actually suck or you don't know how to properly compare audio files which would be equally as unsurprising although how much it sucks (and whether audibly or not) depends on how much higher your differences are.
I think that somewhere you said that an IIR filter would have more phase shift at low frequencies compared to higher frequencies with all else being the same which is just simply not true. If you set an IIR highpass at 100Hz it would have the same phaseshift at 100Hz as if you set this same filter at 1000Hz and checked the phaseshift at 1000Hz.
I would say that you could check this for yourself but frankly I doubt you have the competence for that and I don't know how else I could convince you of this.
Properly answering you takes a long time because you try to keep the discussion technical and yet you are wrong on so many levels that anyone trying to explain you anything would have to go back to basics and start with adam and eve so to speak. Most of it gets lost on you too and people trying to explain how things work will eventually run out of patience if you keep ignoring their points.
Unless you have questions about how to verify (1)that your music consistently contains such low frequencies, how to verify (2)that you filter properly, or how to verify (3)the amount of differences between the original or unfiltered audio, I won't be replying to you because arguing about your "theories" is absolutely pointless.
