Removing phase shifts from analogue EQ..?

There is no such thing as linear phase analog EQ.  All analog EQ is based on phase relationships.  Changing the phase relationship is what causes the nulling/adding and frequency selection..  

Thanks for taking your time to answer my question, Svart!

I guess your explanation makes sense in the context of 'blending' but for me what was interesting about this video is how you can remove phase shifts from your analogue EQ.  

What I wish to know more about is how time reversal of a signal resulted in the flip of a phase shift which resulted in the cancellation of phase shifts. I can vaguely remember going over this topic in continuous time signals back in Uni but can't remember the exact details!

I don’t know , that’s what I was taught as well, if it used some kind of compensation or what JW would know, but when I did my first SIMM training when they still used a TEF anylizer there was no change, except in frequency response. when boosting and cutting.
The CP10 was developed for the SIMM system so that you could EQ parts of the Array without phase issues so that it would not effect the acoustical summation. All Meyers boxes exhibit the same phase response and approximate Freq response so they can be used as building blocks ( this should be a dah in large scale PA world but sadly isn’t). There probably are some old white papers on the Meyer Sound site, Apogee Sound had a similar box as well but I don’t remember it as well.

"Phase" is only relative to the starting point of two signals.  

I think the video is mis-appropriating the word "phase" when it really means to say something like "time alignment".

This trick they are using is manipulating time alignment, not phase.  It might be changing and aligning phase as well, but that's not the attribute they are adjusting.

The hardware linear phase EQs still use phase relationships to add/null frequencies but they attempt to adjust all affected frequencies by the same amount which necessitates much more complex circuitry with it's own issues, IE; all pass filters, etc..  That's why most of them call themselves "minimum phase EQ" rather than linear phase, because it's impossible to do in analog. 

Svart, I read your comment again but I'm still confused. I'm not talking about a discrepancy in time introduced by round trip but phase shifts introduced by the hardware EQ. Combining an EQd signal with the original track doesn't sound like a good idea to me even after making adjustments for the round trip latency. Have you watched the video by any chance? I think I'm either not catching your point or my question wasn't very clear. 

I don't know.. he says 'linear phase analog EQ' at 6:07. He is claiming that this technique is flattening out any phase shifts. Someone on the mastering forum mentioned two pass IIR and that might ring a bell for some of you?

True that. But it's fun to have more options. How do you feel about analog EQs with M/S function? I think EQ's with M/S could benefit from minimised phase shifts?

 I do use MS all the time, and never note any problems if I EQ the side.  I haven't experimented with linear phase EQ on side channel.  You may note the delays effects that occur with linear phase EQ, and they in my experience have been a bigger problem than any phase shift in an analog EQ.   There's as much a phase argument that if you chose a side mic with the rolloff you wanted, it would have a similar phase response as a flat mic that's been EQ'd similarly.  No free lunch, you pay any way you go.  

sin(wt + phi)

Sure, same thing with amplitude applied.

The lightbulb for me (recently) was when my brain correlated circuit resonance with mechanical resonance. For example, a passive inductor filter to a mechanical system. It's resonating at the critical (notch) frequency.

You can even model the electrical circuits as mechanical systems - resistors are springs, inductors are masses, and capacitors are dampers. The math works out the same.

When you have a mechanical system there must be, always will be, has to be a phase change as the system goes through resonance. A thought exercise for this is a weight hanging from a spring underneath a platform that is moving at some frequency. Imagine hanging a box by a spring under a trampoline. Now move the trampoline. For low frequency (slow) bounces, the box, spring, and trampoline mat all move together - in phase. Once you hit oscillation, resonance, critical frequency, whatever you want to call it, the box underneath will be moving down when the trampoline is moving up and vice versa. This is what generates the positive feedback of oscillation with the spring. If you go faster, they'll begin to move together again as you leave resonance.

If we're thinking of this as a passive boost circuit, the "box" is the RC or RLC network passing audio, and the trampoline is the incoming signal. When they're all in phase, no EQ is happening - flat frequency response between in and out. When you're at the resonant frequency, the box (out) is moving differently than the trampoline (in) - so you have EQ happening.

This notionally the same behavior as in EQ or filters. We just use that resonant behavior to selectively cut or amplify signals, either passively (by bypassing a voltage drop or shorting the signal at a specific frequency) or actively (by putting this same behavior inside an amp's feedback loop to selectively boost or cut the specific frequency). But, at least as far is I know, at the actual filter itself there has to be a phase change. Willing to be shown that I am wrong here, however. I'm just a mechanical engineer, all this electrical stuff is pretty much voodoo to me.