looking good Ray!
Yes, it's interesting (to me) to be able visualize and modify this stuff...
looking good Ray!
Time aligning the drivers (using linear phase XO) and linearizing the phase was audible.
For me, doing it manually is half the fun, and educational. Yet it leads to the slippery slope of continuous improvement which quickly reaches a point where the manual approach just doesn't cut it anymore....
Doing it manually, fix something, remeasure, see what you get, try again, etc. You can fiddle with it for hours. ...
Are you saying that if I manage to measure phase correctly at my LP and linearize it I would get nice step response and the result will be hearable?
Rinse & repeat each iteration gets more finicky, at some point you must rely on software analysis and automation no matter how much you enjoyed doing it all manually. That's not necessarily a bad thing.
No, what I am saying is that the individual divers in the speaker need to be time aligned first. This means an active system using a linear phase digital XO where the software, either manually or automatically, calculates the acoustic center of each driver and aligns them through the use of digital delay. Then, if required, linearizing the phase after... As mentioned in the previous post, linearizing the phase of a passive system, but not having the drivers time aligned, to my ears, has little audible effect.
I've got a mic dilemma. I got the UMik-1 recommended here. Now I've measured my room with 3 different mics, each with very different responses. Each mic produces consistent results when re-measured. So I don't know which to trust.
A picture's worth 1,000 words. Here are the non-EQed raw response curves measured from the same listener position, each using that mic's calibration curve. Pink is the Umik-1, Blue is the Zoom H4, Yellow is the Rode NT1-A. Note 2 dB per division to accentuate the differences.
View attachment 23085
My gut says trust the UMik-1 since its calibration data is supposedly individually measured. But it shows a sharp dip at 72 Hz that neither of the other 2 mics show. All 3 mics agree on the 180 Hz dip. The NT1A is a nice mic that sounds great recording music and measures the lowest noise & distortion of these 3 mics. But that doesn't mean it has flatter frequency response! BTW, the UMik-1 showed exactly the same response from L and R channels, but the other 2 mics each showed variations in the 2-4 dB range.
The other two mics are not measurement mics, so there’s no reason to assume they should have a flat response, partucularly not in the low bass and high treble. I’d trust the UMik more here.
. But, if the UMik-1 is right (and the other 2 mics are wrong), I have a trough from 50 to 90 Hz to deal with, too big to EQ. So I'll be revising room arrangement & treatment:
View attachment 23086
That would be good enough, but in my view it is too much to EQ. The red line shows the effect of +6 dB @ 64 Hz. The room sucks up energy at that freq and I'm only getting half the boost I apply (I added 6 dB but only measured a 3 dB increase). After this +6 dB boost it's only about 3 dB below 35-40 Hz, but getting it will take another +6 dB, for a total of +12 dB which is way too much for EQ. This situation calls for changes to room treatment.
I wanted that particular filter to cover the range from 45 to 90 Hz. That's 1/2 octave on each side, which is Q=1.414, centered at 64 Hz.
All of my parametric EQ settings are 1/3 octave (on each side) or wider. That is Q=2.145 or less. I'd go narrower if I had to, but I try to avoid high Q because I'd rather have a few ripples in the response than phasey bloated sound from steep filters. Don't let the cure be worse than the disease!
This response is measured from listener position with both speakers playing. I think you're right, 72 Hz is a room mode not some kind of speaker phase issue. These planar speakers have a near flat-line phase response. The mic's L and R responses are identical. But I can play the tone with just one speaker to see what that looks like.
Alternatively, I could try center 72 Hz, 1/7 octave each side (Q=5). That would cover 65 to 80 Hz. Then add my current center 64, Q=1.4 filter on top of it, +6 dB each, to flatten the entire range 45 to 90 Hz and bring it up to level. But that still would be +12 dB of boost, which demands 16x more power from the amp in this frequency range, and consequent phase distortion from a steep filter, and a -12 dB overall level reduction to avoid digital clipping.That doesn't take into account that deepest point in that regions is at 74Hz sou you would do better to move your fitler frequency closer to it. For that reason I suggested you use 70Hz. ...
I'm using a Behringer DEQ2496 for the EQ. I'd like to stay within its capabilities. My goal isn't perfection, but something around the knee of the results vs. effort curve. Actually, saying that is self-delusional considering the hours I've spend building giant tube traps, measuring & arranging the room, etc.Amplitude non-linearities are more affecting SQ than phase non-linearities. Besides, you can always correct the phase once you are done with amplitude response. That is the beauty of FIR filters.
I'll try that tip: measure each speaker independently. My DEQ2496 can apply different curves for L and R. Since these are room modes, I don't expect much difference but it's worth a try -- surprises keep it interesting!It's ok to measure both speakers response to check LF response up to Schroeder frequency but you can't make usable correction based on such response. As you will be applying correction to each speaker separately you have to measure them separately as well.