In my own thinking, the time of first air movement was more important than alignment at the crossover frequency, and I "figured" if there were any correction, it would be additional woofer delay. I tried testing the other direction only briefly with a big difference.
Anyway, Thursday night's alignment ended up in total chaos, things were not looking correct at all (though it sounded ok).
So I decided to start all over on the subwoofer/panel alignment. Starting from the time of first air movement. THEN, from that point, I "fairly" looked fore and aft, starting with the smallest increments in delay adjustments so as not to miss something close (as I had apparently done on Thursday night). The smallest increment is 0.02ms.
On the right speaker, I optimized the total output at the crossover frequency, 100 Hz. At first I worried how I would do this, knowing acoustical measurements to be hard to make sufficiently precise. But I simply decided to use my new program in the ordinary frequency response calculation, and see what it calculates for 100 Hz. It is precise to 0.1% (in linear voltage) and apparently quite repeatable.
Technically I run an impulse, using the sweep signal, and then convert it to frequency response (with distortion harmonic estimations also, an added bonus). To help improve the repeatability and accuracy, I set it to run 3 sweeps and average.
It was quite clear from the beginning that I had been optimizing the bass in the wrong direction. Instead of adding MORE delay to the bass, to move the alignment further out past the "phase lead" or whatever it is, it turned out I needed to subtract even more delay from the bass.
After seeing which direction to go, I scaled up the changes and then scaled them back down as nearing what appeared to be the optimum, peak response at 100 Hz.
Satisfied with right channel, I moved on to left channel. But this time, after a few measurements, I decided to try something different. I inverted the panels on the panels (sadly I have no easy way to do this for the bass) by swapping speaker connections, and then maximized the null. I was surprised and very pleased that in fact there is a very deep null at the crossover frequency when I do this. And I made it as deep as it can be, within the closest 0.02ms of delay.
The nulling approach is really the better one, I think, and if I remember correctly it's also what Linkwitz recommends.
The result of doing one channel by peaking and the other channel by nulling nevertheless makes sense. The difference between the two seems about what I would expect based on the fact that one subwoofer is about 5.5 inches further back (a situation not easy to change, it's very cramped with equipment in the front of the room). Correspondingly, the delay for the closer subwoofer is 0.3ms more.
(On the previous day's adjustment, the delay difference was opposite from expectations, leading me to believe something was wrong.)
But then, seeing generally how poor the curve below 200 Hz was in the left channel, I decided to optimize the parametric EQ settings (PEQs). I didn't add or subtract any PEQ's, just change their magnitudes, in some cases by fairly large amounts. There is little doubt that ARTA is showing a more accurate frequency response curve than my phone's RTA app, which is generally what I have used before (though, combined with very slow hand sweeping, which I still believe is the best way to find the critical points where adjustments should be centered).
I wasn't sure this was going to work out well, because it was already late and I wasn't going to have much time, but it did work out pretty well. There had been about a 20dB rise below 100 Hz, 100Hz being at a low point (even after time delay adjustment and putting the polarity back to normal). I lowered the excess deep bass, and raised the area around and just above 100 Hz, so it's much closer to flat overall.
I've also been working on getting the miniDSP used for the supertweeter changed from 48kHz sampling rate (response to about 24kHz at best) to 96kHz sampling rate. These units convert all inputs to their internal sampling rate--whatever it is. Using the standard "plugin" supplied by miniDSP, you get 48khz. Actually the website says it differently, it says a different plug-in in standard and will get you 96kHz, not even mentioning the old standard 48kHz one I got last year.
I ordered a new miniDSP a few weeks ago for experimentation, and I thought therefore I would get the new 96kHz plug in. But the website info was grong, I still got the old 48kHz plugin. I tried buying the one they indicated for 96kHz. It only costs $10 so I didn't want to wait to try to send get them to me for free.
Well, sadly, that didn't work at all, I just got error messages. I also tried another plug in that seemed close to what had been recommended in forums. That didn't work either.
So finally yesterday I posted my problem as a question to the miniDSP user forum. Within a few hours, a veteran user suggested what I needed. So I've obtained that plug in, and hope to get the 96kHz working today.
That will require me to readjust the delay because likely the new plugin will have different latency. But now I have good ways of setting the time alignment.
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