Why is a critically damped Q factor bad?

It doesn't need to have settled to a zero state in order to start responding to the next portion of the signal.
What if the 'next portion of the signal...' happens to be silence; will a loudspeaker having lower stored energy more accurately reproduce such a signal in the time domain due to reduced settling time, or are we back to the 'equalised response' apparent paradox?
 
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That's an interesting question. My experience has been that music gets to silence in between tracks or at the end of the album. In getting to the silence, there is usually a period where the amplitude of the music is progressively reduced (faded out) on its way to zero amplitude.

The question you have posed has, in effect, answered itself. Of course, without fail, if there is reduced stored energy (i.e., reduced resonances), then the time domain signal will be more accurately reproduced.

Consider the two different frequency response functions shown below. Their resulting time domain behaviours are also different. Which one results in greater accuracy? A or B?

(A)
1728042275920.png


(B)
1728042362858.png
 
I tried to but was unable to confirm these observations. I took the original track in Audacity and then high-pass filtered it with a 24dB/octave filter with a −3dB cut-off frequency of 25Hz. I couldn't tell any difference between the filtered and unfiltered tracks using an instantaneous A/B comparison. The only thing that was plainly audible to me was that I couldn't hear any difference.

I then went and applied a 48dB/octave filter with a −3dB cut-off frequency of 20Hz to the original track. I still couldn't hear any difference.

I then went and changed my audiophile-grade headphones to another pair of audiophile-grade headphones. Still no difference, although there was a difference between headphones. Phew! I was worried that I needed to get my hearing checked. 🙂
Some people are more sensitive to this than others it seems, and of course the effect is subtle, not a night and day difference. Quick brute-force A/B switching while you are emotionally excited to get a quick result usually pulls detection threshold way up.

Once you've experienced the difference in what I call non-forced/non-focussed listening, you cannot "unhear" it again. It's still subtle but at some point you get the hang of it. Same with polarity switch or general phase distortion.

I personally find the "lagging bass" effect to be rendered more pronounced on (linear phase) speakers vs headphones and it naturally gets more pronounced at higher frequencies (try 40Hz, for example). Also, I found pretty low listening levels give better results. But that's just me, of course.
 
I consider that I wasn't emotionally excited to get a quick result. I was trying to listen critically and objectively to see if I could hear a difference. Similarly, I have experienced hearing differences in unfocused, non-forced listening. Only to have any differences become no longer audible in more critical objective listening.
 
Note the distinctly large peak at about 5Hz as well. Where did it come from?
Could be anything from captured HVAC noise to compression artifacts.
You could put a very steep low pass on at say, 15Hz, like a Bu4 applied two times (the second pass in time reversal to get a linear-phase lowpass), giving a LR8 low-pass. Then play it back at 2x, 4x or 8x speed to see what it sounds like and how it correlates to the music.
 
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So now what has all this to do with Q factor. If I want to have new shoes then the Quality must be 100% not 50%, It may look good but squeaks when walking or it hurts my toes. Someone with slightly elongated feet may find them more comfortable and in that they assign it a higher Quality factor than I. Are we just not pulling the a$$ out of the chicken?
What if we play the signal backward, i.e. the decay happened before the attack, would Q affect this?
 
I was trying to listen critically and objectively to see if I could hear a difference.
IMHO, that doesn't work, or better said, it only works once you already know what to listen for.
But as mentioned, there are people who are "phase-deaf" so to say, and also don't hear lagging bass unless it a quarter-second late ;-)
My former boss at a speaker company was one of those people but he had extreme listening skills in other areas, mainly general frequency response and especially subtle resonances and standing waves in the midrange... he could pinpoint things in an instant easily whereas I had a really tough time deciphering the subtle differences.
 
Yeah if someone has played with REW and it's alignment tool trying to align sub and mains and make "nice response" around listening spot one might be familiar with this delayed sound. At least I'm 😀 The alignment tool could suggest many millisecond delay and if one puts this into the system it'll make this audible lagging thump effect as well. On the other hand, just reducing the delay fixes it to an extent, it's not so obvious anymore regardless of slopes. Anyway, it can be quite obvious sound when everything is not ok down low.
 
@perrymarshall, would you mind extracting a portion of the the waveform that relates to the "thump" and the "slap" that you are referring to, and provide a plot? I'm having trouble identifying what you are describing.

The bass spectrum on that track is certainly "wideband", that's for sure, as shown in the plot below.
View attachment 1363895

Note the distinctly large peak at about 5Hz as well. Where did it come from?
In real life, the slap and the thump arrive at the exact same time. You hear the smack and feel the punch in the chest simultaneously.

In systems with steep filters (let's say 36 db/octave) and large group delay, with this song you physically feel the thump a split second later. It's not super obvious, but once you notice it, you can't "un-notice" it.

It's very common for bass drums to have significant output up to 5KHz and above. This distinction is arbitrary but in my vernacular the "thump" is the parts below 100Hz and the "slap" is those above 100Hz.

I can only guess but maybe the 5Hz you see is simply in the original recording, because the mic is close to the bass drum head. It seems to me that a bass drum head could produce meaningful output down to almost DC.

I've not done this experiment with headphones. I've done it with high-Xmax acoustic suspension subs that I could EQ flat down to 10Hz. When I add steep HP filters I can clearly sense the group delay. It is as much felt as heard.

This issue directly pertains to the subject of this thread, because a critically damped system settles to zero as fast as theoretically possible, instead of oscillating. A 36dB/octave filter takes a long time to settle down.

Re: Audacity: Does Audacity use IIR or FIR filters? IIR filters would create the group delay I'm speaking of, FIRs likely not.
 
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Please provide the name of one track that you feel is a good example of "square bass".

Not a track but there is many examples of sin/saw/square electronic sub bass:


Sine sound like a lone fundemental ( which it is!), triangle have fundemental and only even harmonics ( sound 'fuller'), square fundemental and only odd harmonics ( sound a bit 'hollow'), sawtooth have fundamental + even and odds harmonics ( sound a bit 'screetchy'/shredding).

I must have more cleaner example from older style genre if you are interested ( or can generate them from my own toys).

Meanwhile as i'm lazy here is how it sounded and was done in 90's house:

 
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As far as I experienced from my listening tests you have no advantage of a too low Q like 0.5 because it simply is not loud enough and psychoacoustically the mid frequencies dominate the sound.

By electronically correcting the amplitude you gain nothing as the Q is then the summed electric and mechanic Q forming the higher Q. You end up turning the bass up in order to adjust it to your taste.

I worked some years intensively with a dsp assisted high Q fullrange driver system. And you could adjust it to the Q you desire.

Psychoacoustically I had days I turned the 40hz up by one or two db - the other days back again.

Same with the 150hz region.

Electrically there is only one right linear alignment which is correct. But the perception can change and you play (in small grades) with the highs and lows with the dsp.

No pardon was given in the mids here I never touched the once found perfect balance of on and off axis summed linearity.

Maybe you can put half a db more or less here over two octaves in order the emphasize the mids or not.

With dsp you could tune the bass perception between dry and articulate and slow and mellow without changing the box.
There's also no standard speaker a recording was intended to be listened at home with. Personal preference plus speaker differences plus room differences to me mean to me- if you're fussy in any way, it's probably a good idea to have some kind of tone control.
 
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Yes because it's not so easy to point to a 'square wave' sound as in electronic music anything is tweaked to death at production/mixing stage ( and even ssynth square waveforms can sound different between synths) and the only way to get how they sound is to play with them.

Did you notice how different the two video sounds despite they use the same 'basic' waveform? Because the music style are differents...

No need to become a musician, just a knob tweaker! 😉
 
Nobody is arguing Nico Ras, Witwald asked for example, KSTR and I answered, this raised a question i answered, end of the sequence.
When you talk about original, what are you talking about, i don't get it?

That said it's not off topic: i produce electronic music and i'm audio engineer too. In both case i prefer QTC 0.5 for my sealed subwoofer/woofers despite it's not what would be expected from such kind of music ( when i produce electronic music) from answers i've read in this thread... why?

I've got my own pov on this, and let's say time domain is too often underated wrt frequency domain. And even if room mess things below schroeder frequency when a loudspeaker modify too much tail of signal to be reproduced it's not good imho/experience.