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Author Topic: Combining drivers  (Read 1443 times)

Helge A Bentsen

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Combining drivers
« on: March 02, 2021, 05:25:39 pm »

Attached is the phase plot of two drivers post equalization with the same reference delay.

What would be your method for combining these for the most linear phase response?

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Russell Ault

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Re: Combining drivers
« Reply #1 on: March 02, 2021, 10:45:30 pm »

Attached is the phase plot of two drivers post equalization with the same reference delay.

What would be your method for combining these for the most linear phase response?

That red plot is barely visible on my screen. Is your coherence blanking jacked way up? Can you attach the two .trf to your post (or dump them out as ascii)?

Also, to make any meaningful recommendations it's important to see the mag trace as well, since the goal is phase alignment within the overlap band (and without the mag traces we don't know what the overlap band is).

-Russ
« Last Edit: March 02, 2021, 10:47:42 pm by Russell Ault »
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Helge A Bentsen

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Re: Combining drivers
« Reply #2 on: March 03, 2021, 04:40:32 am »

That red plot is barely visible on my screen. Is your coherence blanking jacked way up? Can you attach the two .trf to your post (or dump them out as ascii)?

Also, to make any meaningful recommendations it's important to see the mag trace as well, since the goal is phase alignment within the overlap band (and without the mag traces we don't know what the overlap band is).

-Russ

I've turned off coherence just to clear up the screenshot.
Also forgot to save the two captured traces showing overlap, this is with a Linear Phase BW crossover inserted.
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Helge A Bentsen

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Re: Combining drivers
« Reply #3 on: March 03, 2021, 04:42:50 am »

Btw, it's a measurement indoors in my awful sounding loading bay with the speaker on top of a road case.
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Russell Ault

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Re: Combining drivers
« Reply #4 on: March 03, 2021, 06:14:36 am »

I've turned off coherence just to clear up the screenshot.
Also forgot to save the two captured traces showing overlap, this is with a Linear Phase BW crossover inserted.

Ah, okay; the reason I thought you had coherence blanking jacked way up is because your crossovers are super steep and pass-band overlap is tiny.

It's been a little while; let's see if I still remember how to do this. (If I don't, I know someone will be along shortly to correct me; my apologies in advance.)

It looks like your crossover frequency is ~1.26 kHz. In the area of overlap between the two pass-bands (i.e. where there's less than a 10 dB level difference between the two mag traces) it looks like the LF is leading by ~15 degrees, and the HF is lagging by ~165 degrees.

To time-align the two passbands you'll need to add delay to the LF; specifically, you'll need to delay the LF enough to produce ~210 degrees of phase shift at the crossover frequency (15 degrees to go from leading to in time, plus 180 degrees to produce a wrap, plus another 15 degrees to go from 180 to 165), which works out to ~0.46 ms.

Does that sound about right?

-Russ
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Helge A Bentsen

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Re: Combining drivers
« Reply #5 on: March 03, 2021, 08:34:09 am »

Ah, okay; the reason I thought you had coherence blanking jacked way up is because your crossovers are super steep and pass-band overlap is tiny.

It's been a little while; let's see if I still remember how to do this. (If I don't, I know someone will be along shortly to correct me; my apologies in advance.)

It looks like your crossover frequency is ~1.26 kHz. In the area of overlap between the two pass-bands (i.e. where there's less than a 10 dB level difference between the two mag traces) it looks like the LF is leading by ~15 degrees, and the HF is lagging by ~165 degrees.

To time-align the two passbands you'll need to add delay to the LF; specifically, you'll need to delay the LF enough to produce ~210 degrees of phase shift at the crossover frequency (15 degrees to go from leading to in time, plus 180 degrees to produce a wrap, plus another 15 degrees to go from 180 to 165), which works out to ~0.46 ms.

Does that sound about right?

-Russ

Almost exactly what I did :) 0.44ms delay on LF if I remember correctly.

Sounds perfectly all right, both on and off-axis. However, I end up with one phase wrap through the crossover, is there a way of doing this without this wrap?

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Mark Wilkinson

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Re: Combining drivers
« Reply #6 on: March 03, 2021, 10:43:48 am »

Almost exactly what I did :) 0.44ms delay on LF if I remember correctly.

Sounds perfectly all right, both on and off-axis. However, I end up with one phase wrap through the crossover, is there a way of doing this without this wrap?

It looks like you are using linear phase xovers, given how steep the mag response is, and that the phase curves are still relatively flat. Yes? No?

If yes, the remaining phase curvature is from the drivers' natural mag rolloff. 
So the way to get rid of that wrap, is to flatten the drivers' out of band response before appling the xovers.

Do this with minimum phase EQ's.
I like to try to get mag and phase flat out-of-band down to the frequency where they contribute -30dB to summation. But -20 dB is fine.

It's kinda cool how raw driver mag and phase mirror each other over a horizontal axis.
IOW, if mag is a frowny face (almost always); phase will be a smiley face.  Flatten one, the the other flattens too.

I's also cool how that steep linear phase xovers  take care of what would normally be dangerous excursion from out-of-band boost.

It's truly a have your cake and eat it too scenario, imho. 
As long as you're working with xover frequencies that don't require too much FIR latency for using the linear phase xovers.

If you're not using lin phase xovers with what you posted...please flush all this down the crapper!!!  And quick haha
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Russell Ault

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Re: Combining drivers
« Reply #7 on: March 03, 2021, 02:28:29 pm »

{...} So the way to get rid of that wrap, is to flatten the drivers' out of band response before appling the xovers. {...}

I'm sure Mark's answer is better than mine; my inclination would be to do what you've already done and then apply processing to the pre-crossover signal to offset the phase wrap.

In FIR a single "maximum phase" all-pass filter should do the trick. In IIR (i.e. if you're John Meyer) you can achieve the same effect using a series of second-order minimum phase all-pass filters (and some patience).

-Russ

ETA: Of course it should be noted that, while there are many valid reasons for wanting speakers with a flat phase response, a phase wrap at the frequencies in question will be entirely inaudible to humans.
« Last Edit: March 03, 2021, 02:37:28 pm by Russell Ault »
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Helge A Bentsen

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Re: Combining drivers
« Reply #8 on: March 03, 2021, 03:07:38 pm »

Ok, I'll try to get outside next week and refine my pre-alignemt EQ and see what happens.
IIRC it's a linear phase brick wall filter, 1.2Khz/79db/oct.


Thanks :)
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Russell Ault

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Re: Combining drivers
« Reply #9 on: March 03, 2021, 06:39:15 pm »

{...} If yes, the remaining phase curvature is from the drivers' natural mag rolloff.  {...}

So, I've just been having another think about this, and I feel like there must be more to this alignment than just the drivers' natural roll-off. If this speaker uses any kind of an HF waveguide (horn, etc.) there's almost certainly a physical offset between the HF and LF, and compensating for this requires some kind of time-domain solution, wouldn't it? Without knowing the speaker's design it's difficult to say for sure, but I still feel like the only way for this speaker to have a "pc75" phase response is going to involve delaying the LF and then dealing with the resulting wrap one way or another.

-Russ
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Mark Wilkinson

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Re: Combining drivers
« Reply #10 on: March 03, 2021, 08:31:20 pm »

So, I've just been having another think about this, and I feel like there must be more to this alignment than just the drivers' natural roll-off. If this speaker uses any kind of an HF waveguide (horn, etc.) there's almost certainly a physical offset between the HF and LF, and compensating for this requires some kind of time-domain solution, wouldn't it? Without knowing the speaker's design it's difficult to say for sure, but I still feel like the only way for this speaker to have a "pc75" phase response is going to involve delaying the LF and then dealing with the resulting wrap one way or another.

-Russ

For sure Russ, (and thanks for the earlier vote of confidence, but i'd trade what i know for what you know in a heartbeat !)
with both drivers having flat phase on their own, it still takes some time delay to tie them together so that the entire trace is mag and phase flat.

After flattening each driver's out-of-band response, the difference in Smaart's Delay Finder between the two, will exactly equal the processor delay needed to tie them together.
And that processor delay also equals  physical distance between acoustic centers of the two drivers.

That's one of the beauties of flat phase....makes timing easy peasy....and also allows common sense distance observation/verification like you were mentioning.

If Helge gets the out of bound flattening in place on the upper end of the lower driver, and the lower end of the upper driver, he will find he can move the FIR brickwall xover to any freq  within the jointly flattened range, and not have to change processor delay at all (as found via the above).
That's another of the beauties.



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Russell Ault

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Re: Combining drivers
« Reply #11 on: March 03, 2021, 10:57:54 pm »

{...} with both drivers having flat phase on their own, it still takes some time delay to tie them together so that the entire trace is mag and phase flat. {...}

Sorry, something still isn't working for me, but my experience with zero-phase filters is basically, er, zero, so let me see if I have this correct:

  • Setup a speaker and a measurement system somewhere where no one will want to punch you if you run pink noise for an hour straight
  • Put pink noise through HF driver
  • Use Delay Finder, note the delay compensation time
  • Adjust EQ/phase response of the driver until everything is as flat as you can get
  • Put pink noise through LF driver
  • Use Delay Finder, note the delay compensation time
  • Adjust EQ/phase response of the driver until everything is as flat as you can get
  • Insert zero-phase crossovers to taste

Right so far? If so, here's the problem I'm seeing:

We have our beautifully flat HF trace and our beautifully flat LF trace, but the LF is arriving early. As you say, no problem, we just delay the LF channel by the Delay Finder difference. Now everything lines up, but the LF trace is no longer flat, because adding delay to the LF causes its phase trace to start sloping.

The only way to avoid this entirely that I can think of would be to do all the component-level flattening at the single, fixed delay compensation setting found with the latest-arriving component. Done this way, any needed alignment delay would be already baked into any phase-altering processing.

In the case of Helge's traces, flattening the phase trace of the HF is going to take some significant processing muscle since it's lagging by ~180 degrees at the crossover point. An FIR maximum-phase all-pass filter will do the trick, but will effectively result in the highest HF frequencies arriving later than they originally were, which will necessitate re-setting the delay compensation. (Doing this IIR with a whole mess of minimum-phase all-pass filters would have the same effect since, of course, you can't actually make the sound at 1.2 kHz arrive earlier, you can only make the sound at higher frequencies arrive late enough to match.)

Now that the HF is flat it's time to face the bad news waiting for us in the LF. Since the delay compensation has increased from 20.90 ms to ~21.31 ms (or more), the LF trace that was once within 15 degrees of in time through the crossover is now significantly leading, so we'll have to add one (or more) minimum-phase all-pass filters to re-flatten it, effectively using all-pass filters instead of channel delay.

With all that done, we will have two drivers that are in in time with each other throughout both pass-bands, but it'll take a while.

Personally, I think I'd rather do what Helge's done and then phase-tweak the combined response. It just seems easier to me! :)

-Russ
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Mark Wilkinson

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Re: Combining drivers
« Reply #12 on: March 04, 2021, 01:21:38 am »

Sorry, something still isn't working for me, but my experience with zero-phase filters is basically, er, zero, so let me see if I have this correct:

  • Setup a speaker and a measurement system somewhere where no one will want to punch you if you run pink noise for an hour straight
  • Put pink noise through HF driver
  • Use Delay Finder, note the delay compensation time
  • Adjust EQ/phase response of the driver until everything is as flat as you can get
  • Put pink noise through LF driver
  • Use Delay Finder, note the delay compensation time
  • Adjust EQ/phase response of the driver until everything is as flat as you can get
  • Insert zero-phase crossovers to taste

Right so far? If so, here's the problem I'm seeing:




Personally, I think I'd rather do what Helge's done and then phase-tweak the combined response. It just seems easier to me! :)

-Russ

Aaah, my apologies Russ, i wasn't clear in last post when to use Delay Finder's readings..
And just to be clearer backing up further to first post, I was trying to answer Helge's question about how to remove the final remaining phase wrap, not suggest an overall different methodology to what you two were discussing.)

May I rearrange your sequence:
  • Put pink noise through HF driver
  • Adjust EQ/phase response of the driver until everything is as flat as you can get
  • Put pink noise through LF driver
  • Adjust EQ/phase response of the driver until everything is as flat as you can get
  • Insert zero-phase crossovers to taste
  • Use Delay Finder, note the HF delay compensation time
  • Use Delay Finder, note the LF delay compensation time

   *Use the Delay Finder difference as the delay to be put into processor for delaying whichever driver had the lower Delay Finder reading.
   *Done, in hopefully a lot less than an hour  :)

I guess we each have learned our methods, and they become the easiest for us. I use phase trace overlay a lot too, because dang, we most often have to.
Honestly, I consider guys like you and all the others that have mastered aligning sloping phase traces to have learned the universal tool, the more difficult tool to learn.
Because the additional flattening steps i proposed only work when both flattening and linear phase xovers are practical.
But when they are practical, alignment gets easier than phase overlay ime/imo.



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Mark Wilkinson

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Re: Combining drivers
« Reply #13 on: March 04, 2021, 01:27:05 am »

Ok, I'll try to get outside next week and refine my pre-alignemt EQ and see what happens.
IIRC it's a linear phase brick wall filter, 1.2Khz/79db/oct.


Thanks :)

You bet !
Yep, looked like a brick wall.
Hope flattening works, please see some further comments about timing made in posts with Russ.
Also, must say I agree with Russ you're unlikely to hear a change at that freq, but it does make for nice looking traces haha. 
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Russell Ault

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Re: Combining drivers
« Reply #14 on: March 04, 2021, 04:43:09 pm »

Aaah, my apologies Russ, i wasn't clear in last post when to use Delay Finder's readings.. {...}

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. :)

{...} Also, must say I agree with Russ you're unlikely to hear a change at that freq, but it does make for nice looking traces haha. 

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
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Mark Wilkinson

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Re: Combining drivers
« Reply #15 on: March 05, 2021, 12:17:33 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

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.)





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Mike Caldwell

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Re: Combining drivers
« Reply #16 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.

Helge A Bentsen

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Re: Combining drivers
« Reply #17 on: March 05, 2021, 06:39:16 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.
Yes, did that to fine-tune my delay time using a mic on-axis and one off-axis.
« Last Edit: March 05, 2021, 06:42:13 pm by Helge A Bentsen »
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Russell Ault

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Re: Combining drivers
« Reply #18 on: March 05, 2021, 07:43:00 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.
{...}

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
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Chris Grimshaw

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Re: Combining drivers
« Reply #19 on: March 07, 2021, 02:50:24 am »


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.

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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Helge A Bentsen

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Re: Combining drivers
« Reply #20 on: March 07, 2021, 07:21:44 am »

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

I've observed similar things in every warehouse-demo I've been in. Listen to a speaker at low levels is usually ok, turn it up and you need Peltors.
My storage is just like that, concrete floors, walls, roof and very little soft things absorbing sounds. Road cases are deliberately not soft.
Just opening the loading bay door improves SQ.
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Russell Ault

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Re: Combining drivers
« Reply #21 on: March 15, 2021, 02:26:28 am »

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

I've spent a lot of time pondering this since you posted it. What you described is something I've definitely experienced myself, but I was having trouble reconciling it with my understanding of the precedence effect, and I think I've finally figured out why:

I don't think what you're describing is related to the precedence effect. To the best of my knowledge (and in some fairly un-scientific tests I performed on myself) the precedence effect doesn't change significantly with absolute level (although relative levels are hugely important). Moreover, my guess is that the delay time of the echo you're describing is too long for the brain to perform fusion on it anyway, especially for something impulse-y like a snare hit (where I would expect precedence-based fusion to break down at roughly the 5 ms mark for most people).

Instead, my guess is that what you're describing is actually a form of reverberance. There are studies going back at least 20 years (Hase's "Reverberance of an existing hall in relation to subsequent reverberation time and SPL" is the earliest my brief searching found) demonstrating that, while reverberation (i.e. the, measurable, physical property of a room or sound) doesn't typically change with absolute level, reverberance (i.e. the psychoacoustic experience of reverberation) does. Basically, higher SPL produces greater reverberance, even though the reverberation remains the same.

Does that fit with your experience?

-Russ
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Mark Wilkinson

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Re: Combining drivers
« Reply #22 on: March 18, 2021, 10:44:21 am »

I've spent a lot of time pondering this since you posted it. What you described is something I've definitely experienced myself, but I was having trouble reconciling it with my understanding of the precedence effect, and I think I've finally figured out why:

I don't think what you're describing is related to the precedence effect. To the best of my knowledge (and in some fairly un-scientific tests I performed on myself) the precedence effect doesn't change significantly with absolute level (although relative levels are hugely important). Moreover, my guess is that the delay time of the echo you're describing is too long for the brain to perform fusion on it anyway, especially for something impulse-y like a snare hit (where I would expect precedence-based fusion to break down at roughly the 5 ms mark for most people).

Instead, my guess is that what you're describing is actually a form of reverberance. There are studies going back at least 20 years (Hase's "Reverberance of an existing hall in relation to subsequent reverberation time and SPL" is the earliest my brief searching found) demonstrating that, while reverberation (i.e. the, measurable, physical property of a room or sound) doesn't typically change with absolute level, reverberance (i.e. the psychoacoustic experience of reverberation) does. Basically, higher SPL produces greater reverberance, even though the reverberation remains the same.

Does that fit with your experience?

-Russ

I've been pondering the variables in this thread too.  I think there are several different things being kicked around.

With regard to sound going non-linear indoors when volume is turned up, I think that is due to differences in the time it takes for frequencies to decay across the spectrum, like in RT-60 measurements.  Unless in a room with super solid walls, ceiling, floor... bass absorption is invariably greater than high freq, and the resultant unpleasant tonal shift occurs from relatively longer lingering high freq reverb.

I'm thinking slap echo, etc, heightens the tonal shift and muddiness further, because i saw Don Davis say on a SynAudCon video, that the Haas effect keys off peaks.
That it's peaks tied together, not just a general time window for all equal level frequencies.  He said if a string of peaks occur within 20ms each, the Haas effect can be extended on and on.    Which sounds like echo slap ear bleed to me  ;)

In circling back to phase audibility itself, I don't think either RT-60 room problems, or the Haas effect have anything to do with it.
I mean, RT-60 differnces are an acoustic issue. 
And Haas is more strictly defined/measured for the ear's integration of identical sounds, than for complex music.  For clicks, wiki says it's under 5sec , but for music It appears to turn into a guessing range of up to 40ms, before it's heard as echo.
The clincher for me that Haas is a separate issue from phase audibility, is how I've heard single transients,... kick, snare, bell, etc.... sound considerably stronger/clearer after flattening phase.

Anyway, just a few more 2c thoughts...
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Art Welter

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Re: Combining drivers
« Reply #23 on: March 23, 2021, 08:23:39 pm »

I've been pondering the variables in this thread too.  I think there are several different things being kicked around.

With regard to sound going non-linear indoors when volume is turned up, I think that is due to differences in the time it takes for frequencies to decay across the spectrum, like in RT-60 measurements.  Unless in a room with super solid walls, ceiling, floor... bass absorption is invariably greater than high freq, and the resultant unpleasant tonal shift occurs from relatively longer lingering high freq reverb.
Mark,

High frequency absorption is usually greater than low frequency in most domestic size rooms, and in large halls the audience can provide HF absorption, but has less effect at low frequency.

That said, the experience of short reverb times sounding benign at lower levels, but progressively "sounding" worse as level is increased seems not to be RT-60 related, but due to the way our hearing works- the echos are perceived more as "noise" at louder levels.

I'll provide a quote from the "King of Reverb", (30+ years with Lexicon) David H. Griesinger, to explain:
 
"The basilar membrane divides sound pressure into more than 40 overlapping channels, each with a bandwidth proportional to its frequency. So a critical band at 1000Hz is inherently capable of carrying ten times as much information as a critical band at 100Hz. Indeed, we know that most of the intelligibility of speech lies in frequencies between 700 and 4000Hz.

At the vocal formant frequencies each basilar membrane filter typically contains three or more harmonics of speech or musical fundamentals. These harmonics interfere with each other to create a strongly amplitude modulated signal.   Modulation depth of those frequencies are large, and the peak amplitudes align in time. The modulations in the signal are detected linearly by the hair cells, but the nerve firing rate for time variations longer than about 20 milliseconds is approximately logarithmically proportional to the sound pressure. The brain stem separates these modulations by pitch using a number of comb filters each ~100ms long.  The filters detect pitches using the travel speed of nerve pulses in tiny fibers.
Once separated by pitch the brain stem compares the amplitude of the modulations for each pitch across the critical bands to determine the timbre of the source, and compares the amplitude and timing of the modulations at each pitch between the two ears to determine sound direction. Using these cues the brain stem assembles events into separate foreground sound streams, one for each source. Sound left over after the foreground is extracted is assigned to a background sound stream.
Reflections and reverberation randomize the phases of the harmonics. When the reflections are too strong the modulations in each frequency band become noise-like, and although pitch is still detectable, timbre and direction are not. "

Anyway, that seems to be a far better explanation than anything I came across before, and I did spend quite a few decades thinking about the cause of it before coming across the above in David's "The audibility of direct sound as a key to measuring the clarity of speech and music".

Art 
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Re: Combining drivers
« Reply #23 on: March 23, 2021, 08:23:39 pm »


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