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Combining drivers
Mark Wilkinson:
--- Quote from: Russell Ault on March 04, 2021, 04:43:09 PM ---Sorry Mark, looking back I think you were clear. I was pretty tired yesterday and I think my imagination was leaving out a delay compensation adjustment. Now that I've thought it through again I totally get what you're describing.
That said, I will stand by a little portion of my statement: flattening the phase response of Helge's HF driver is going to take muscle. :)
This is a fascinating can of worms. Here's my take:
Humans' ability to perceive time in the audio domain isn't particularly precise. For audio technicians this is very useful, since it gives us wiggle room to do things that would otherwise be impossible (the best example I can think of is the "latency budget" for IEMs).
The experiments into human hearing that ultimately gave us the precedence effect suggest that humans will typically "fuse" two identical clicks that arrive within 1 to 5 ms of each other, perceiving them to be just a single click. For more complex sounds that fusion can happen with delays of up to 40 ms depending on the nature of the sound.
In my mind, since phase is ultimately just time, these findings are directly applicable to the question of "can humans hear phase?". Given that phase offset results in time smear (as opposed to separate, distinct arrivals), it's safe to say that the fusion window is going to be bigger than the 1-5 ms range, although how much bigger will likely come down to the individual as much as the material. Conservatively, though, a fairly safe assumption is that any time smearing of less than 5 ms will be entirely imperceptible to practically all Western ears.
In phase terms, then, a second-order all-pass filter inserted at 200 Hz would produce ~5 ms of time offset and might just be perceptible in a double-blind study. At 1.2 kHz, producing that same time offset would require 1800 degrees of phase shift. Conversely, at 50 Hz even a 90 degree phase offset might be perceptible. Anecdotally, I once had the opportunity to hear a sound system that had been processed to be phase-flat from 30 Hz to 20 kHz, and I can tell you that the A/B difference was not only perceptible, it was visceral.
Of course all filters, by their nature, produce delay in the form of phase offset. While zero-phase filters (and FIR processing in general) appear to be magic, the truth is that under the hood they are still bound by the same laws of physics as any other filters, and the only way to "undo" do the phase offset caused by the filtering is to delay the rest of the signal to match the delay inherent to the filters being used (moar taps!). While this is a fairly transparent process with FIR filtering (and a real pain with IIR), either way it produces one very obvious side-effect: latency.
It's this latency, ultimately, that is the real pisser, at least in live sound. The only way to "undo" some entirely-perceivable phase shift at 50 Hz is to, in effect, delay the rest of the frequencies to match it, which can mean introducing 10s of ms of additional latency. As the frequencies go lower (i.e. the periods get longer) phase shift becomes more and more perceptible, but fixing that phase shift becomes more and more expensive (in both latency and dollars).
Now, that all being said, in my mind the real advantage of speakers that are largely flat-phase is that it makes array-building much easier.
-Russ
--- End quote ---
Good stuff Russ ! I found myself nodding my head yes to your comments.
Re: flattening phase needing muscle.
Yep, i can see if someone is not in the habit of already doing in-band flattening, adding out-of-band flattening would be a task.
If they are accustomed to doing in-band EQs, I think they will be surprised by how easy out-of band flattening is, when using steep xovers.
I find, it doesn't take much work when the out-of-band summation range is narrow, and often helps make the in-band fall into place easier.
But it takes lots of work with traditional shallow xovers like 24dB/oct or less, in which cases i usually i say screw it.
As an aside, another advantage I've found with steep linear phase xovers, is since the range of xover summation is so narrow, off-axis combing issues between the two drivers is minimized to a much narrower freq range than historically typical.
But again, since steep equals excessive phase wrap/group delay, it only works with linear phase xovers. Which of course means there is a low freq limit where steep works, when latency is an issue.
Re: audibility of phase
This truly is a fascinating can of worms! A can I've been digging in for about 5 years now, since i first started playing with FIR.
Here's my 2c take.
Imo, the jury is out. I'm not convinced the historical research has adequately addressed phase audibility in music reproduction.
I can find many studies that usually used test tones and occasionally music, where they ask listeners to judge between whether degrees of applied all-pass filters were audible etc.
Or some tests used simpler time delays at given frequencies.
Typically, the tests were done with headphones or smaller two-ways, since until recently it wasn't readily possible to get larger speakers with flat phase..
At any rate, phase audibility is continually debated huh?, and my take is generally deemed inaudible.
Which reminds of the beginning of the digital photography era, when DSLR's were in the 2-6 megapixel camp. Debates raged on professional photography forums as to whether more megapixels were needed for normal sized prints.
The vast consensus said no, that more MP were only needed for enlargements, because study after study had proven the human eye can only see a little better than 200 lines-per-inch.
Well, twas true...the eye could only see 200 Lpi at the times all the studies were made...because that's all the dang printers could print clearly !!!! Lol
So circling back to 'typically, the tests were done with headphones or smaller two-ways'.
My experience is that headphones and smaller speakers cannot impart the bottom octaves with sufficient impact, to gauge the effect of phase audibility down low.
I think our full-range larger speakers, with realistic bass and dynamics, have been like the old printers, historically incapable of playing music with flat phase for meaningful audible comparisons across the full spectrum.
And to my ears, that is where the advantage of flat phase is....down low. (Like in your 1.2kHz vs 50 Hz example.)
And I completely agree how audible and visceral flat-phase is 30Hz to 20kHz.
It's freaking awesome really...and was why i was belaboring about timing a bit on a recent Transient Response thread in the Sub Forum....saying it's a move beyond punch into the world of slam ;D
But since the latency required to do such is so great, i've become hesitant to talk about it on the Live forums.
Just doesn't seem appropriate.
Anyway, for live, I've given myself a limit of 15ms total latency. Which sets the transition freq for moving from from FIR to IIR.
For playback, i use up to 170ms. (16K taps in an Ebay QSys Core 500...not that $$ at all.)
Mike Caldwell:
As for finding driver offset timing have you done a comparison between using the delay finder
and response null test?
What I'm calling response null amounts to flipping the polarity between the woofer and high frequency driver and if front mounted delaying the woofer back and watching the transfer function until you get the deepest null at the crossover frequency.
Helge A Bentsen:
--- Quote from: Mike Caldwell on March 05, 2021, 05:42:09 PM ---As for finding driver offset timing have you done a comparison between using the delay finder
and response null test?
What I'm calling response null amounts to flipping the polarity between the woofer and high frequency driver and if front mounted delaying the woofer back and watching the transfer function until you get the deepest null at the crossover frequency.
--- End quote ---
Yes, did that to fine-tune my delay time using a mic on-axis and one off-axis.
Russell Ault:
--- Quote from: Mark Wilkinson on March 05, 2021, 12:17:33 PM ---Re: flattening phase needing muscle.
Yep, i can see if someone is not in the habit of already doing in-band flattening, adding out-of-band flattening would be a task.
If they are accustomed to doing in-band EQs, I think they will be surprised by how easy out-of band flattening is, when using steep xovers.
I find, it doesn't take much work when the out-of-band summation range is narrow, and often helps make the in-band fall into place easier.
But it takes lots of work with traditional shallow xovers like 24dB/oct or less, in which cases i usually i say screw it.
{...}
--- End quote ---
I think it depends on how you define "muscle". In the world of FIR, flattening out Helge's HF response would be relatively easy. For me, though, basically everything I've interacted with in my career has been IIR only (the first time I actually got my hands on a DSP that was even capable of doing FIR filtering was after the pandemic started, and even then I haven't actually used its FIR functionality yet), and in IIR flattening a phase trace that's lagging by 180 degrees takes a lot of filters (sometimes more filters than an IIR-oriented system processors really wants to deal with, hence "muscle"). :)
-Russ
Chris Grimshaw:
--- Quote from: Russell Ault on March 04, 2021, 04:43:09 PM ---
The experiments into human hearing that ultimately gave us the precedence effect suggest that humans will typically "fuse" two identical clicks that arrive within 1 to 5 ms of each other, perceiving them to be just a single click. For more complex sounds that fusion can happen with delays of up to 40 ms depending on the nature of the sound.
--- End quote ---
To throw a slight curve-ball into the mix...
To me, an acoustically reflective room sounds fine at lower levels, but can turn to an awful mess if the levels are brought up a bit. To take an example I'm very familiar with, my living room has the speakers against one wall, and the sofa against the opposite wall. At sensible volumes, all is well. Turn it up, and the slap echo between the two walls becomes very obviously audible on snare drums, claps, etc. Anything transient.
I designed the speakers to be tolerant of relatively high power levels, so they're still acting linearly when the problems become audible.
However, if I go around with the measurement mic, the reflections are always there. It's only at higher levels that I notice them, which makes me suspect there's some filtering going on between my ears, which stops working above a particular SPL.
Following that, I'd suggest that any study into the precedence effect etc would be incomplete without introducing SPL as a variable.
Chris
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