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A history of Time Aligned Loudspeakers

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Guys...SCREW step response.
Simply show mag and phase....says it all much more directly...and more importantly, imparts some understanding as to what matters, determines step.
 
No issues for a passive speaker it's not perfect but it's time coherent. Decide for yourself.

Rob :)

Thank you @robh36062, Yes, to ponder....
 
Guys...SCREW step response.
Simply show mag and phase....says it all much more directly...and more importantly, imparts some understanding as to what matters, determines step.

Agree to disagree, If you are a DIY builder you should be looking at Excess Group Delay, Step and CSD/Wavelets. All 3 give you more information in an easy to understand form than just Mag and Phase.

YMMV Rob :)
 
I think you are forgetting how this thread has progressed. The OP said that KEF's are time aligned. I posted the Theil and the Blade as well as another KEF in a later post where the UniQ was not time aligned to show the difference to the Thiel. So I know it's not optimal compared to the Thiel. It is however a perfect example of a Time Coherent system where the time offsets in the STEP are below the audibility threshold.

Rob :)
Every manufacturer and advertiser is "free" to write whatever they want.
This Step Response is very good for a passive speaker!
1779831486172.png


Who model is it?
 
Who model is it?

Here:

 
Here:

As almost always with a first order crossover - very good Step Response!
 
Food for thought:


Rob :)
 
Hello is this in headphones? Also I don't understand why the numbers you have are well below the numbers in the peer reviewed studies? The sensitivity to group delay goes down with frequency not up. WRT L/R 12 and .16 cycles at 350Hz is approximately 60 usec ? The most sensitive region is 1-4k and that is well below the thresholds there.

Rob :)
Again,

 
Guys...SCREW step response.
Simply show mag and phase....says it all much more directly...and more importantly, imparts some understanding as to what matters, determines step.
For full information, one actually needs mag + (unwrapped) phase Bode plots for all ways of a multiway. That's the only way to find out what a step response really shows.

-----:-----

On thread topic, aligned acoustic centers with zero offsets is the foundation of (almost) everything because basically all crossover target functions (including 1st order) are assuming this. Some special variants, however, exploit an explicit offset but that does not make it any different, it's still aligned acoustic centers with fixed non-zero offsets.

Often we see a trade-off where the tweeter is offset by some design constraints (like no wave-guide) and that is "fixed" by reversing the target polarity, trading phase for delay which keeping total mag response in check. But when you do this, the result isn't minimum phase and has time smear (extended wavelet response by 1/2, or 3/2 etc, cycles) in the XO region. This abberration is on top of any natural group delay, which makes the already bad performance of, say, a LR24 subwoofer XO even worse. With tweeter the natural group delay of the XO matters less but the extended pulse's time smear can be audible.
 
Let someone else tell you - is this a good time alignment?
To robh36062 is the question.....
View attachment 535111
KEF Blade Two, step response
What you refer to as "incoherence" is, in fact, the correct and predictable step response of a 24 dB/octave Linkwitz-Riley crossover. I know this because I obtained exactly the same result during my own miniDSP measurements using ideal LR24 filters. My step response bears a striking resemblance to your "bad" example—yet, in a technical sense, it is completely coherent. The first curve represents my sound card; the second is the LR24 at 2.5 kHz.
LR4 3kHz miniDSP.jpg
 
What you refer to as "incoherence" is, in fact, the correct and predictable step response of a 24 dB/octave Linkwitz-Riley crossover. I know this because I obtained exactly the same result during my own miniDSP measurements using ideal LR24 filters. My step response bears a striking resemblance to your "bad" example—yet, in a technical sense, it is completely coherent. The first curve represents my sound card; the second is the LR24 at 2.5 kHz.
View attachment 535246
Through good DSP tuning - something optimal should be obtained near the first curve.

From the second one it is evident - not the same polarity of the drivers and uncompensated (in time) acoustic centers.....
 
Dzięki dobremu dostrojeniu DSP - optymalne wyniki powinny zostać osiągnięte w pobliżu pierwszej krzywej.

Z drugiego wynika, że nie jest to ta sama polaryzacja przetworników i nieskompensowane (w czasie) centra akustyczne...
Pierwsza krzywa – bezpośrednia odpowiedź Twojej karty dźwiękowej (pętla zwrotna lub charakterystyka ścieżki pomiarowej).

Druga krzywa – wynik sumowania sygnałów z wyjść miniDSP w rzeczywistym wzmacniaczu (lub pasywnym/aktywnym układzie sumującym), gdzie:

  • Jednym z wyjść jest filtr dolnoprzepustowy LR24 (Linkwitz–Riley 24 dB/okt.)
  • Drugim wyjściem jest filtr górnoprzepustowy LR24 (Linkwitz–Riley 24 dB/okt.)
Suma tych dwóch sygnałów podawana jest do wzmacniacza (jeden kanał?), a następnie nagrywana przez kartę dźwiękową (wejście ARTA).

miniDSP LR24 2500.jpg
 
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Just like in the graph, the sum of ideally time‑aligned LR4 filters is measured.
lr4.jpg
 
For full information, one actually needs mag + (unwrapped) phase Bode plots for all ways of a multiway. That's the only way to find out what a step response really shows.

Yes, exactly. And with that excellent and useful information, my opinion is why even look at step.

Well, I take that back....given the ability of FIR to make near-perfect flat mag and zero phase transfers, equating into near-perfect impulse and step....
....step ends up being the best way I know to see how the speakers low end roll-off was handled, and/or how a low-end high-pass if used, was implemented.
IOW, Step is the easiest way I know to assess 'pre-ring' potential (due to a lack of complementary summation at the bottom end, that xovers provide throughout the rest of the spectrum). Something I think will become more common knowledge as active DSP FIT becomes more prevalent.
 
Just like in the graph, the sum of ideally time‑aligned LR4 filters is measured.
View attachment 535259

Good post. And helps me explain my (rather negative) opinion on step.

Some may look at that step and find polarity or timing problems with it....yet it is perfect ! As you know, it is polarity, phase, and time aligned.
A two-way using an IIR LR24 cannot have a better step.

For every IIR crossover topology, two-way, three-way, etc...and then whatever types and orders of acoustic xovers were employed in between ways, there is a unique Step response, that represents the optimal polarity, phase, and time alignment. A unique best fingerprint achievable so to speak.

And it will most likely look like dogshit compared to an ideal first-order Step response. But if fact be excellent, given a non-first order design.
 
I was merely pointing to the effects of distortion and the use of passive components. Active solutions seldom take that into the equation.

I had seen that a while ago. I've been designing my (passive) system crossovers exactly that way since my first 3-way about 25 year ago. I had no clue to the specifics related to distortion, though I made assumptions about it. I did it because that was the best way to approximate the ideal lowpass SPL response of a midrange or woofer in the pass-band and into the stop-band. I've only recently started doing distortion measurements with new microphones and software so I may investigate this. Good to know that what I did also resulted in much improved distortion results.

I'm using DSP for my main system now so that's unecessary or so I think. It would be interesting to see if a combination of passive peaking SPL correction along with DSP would make a difference in the distortion profile. My initial thought is that it would not. Past practice by many was often just using the passive filter to push any SPL deviation into the stop-band, often not very far down. I found it better to reduce them directly as shown at that link by Purifi Audio. Simply pushing into the stop-band still left "remants" of it. No doubt that often had to do with the increased cost of a passive crossover due to the additional components required. I had no limitation, so I used whatever was required.
 
Just like in the graph, the sum of ideally time‑aligned LR4 filters is measured.
View attachment 535259
Oh, dear LSPhil - there are rules, methods and tasks
that I did not invent. The topic is not easy and simple - as some imagine! Without understanding Principles, Processes and Problems through Transient Analysis with a suitable simulation program - there can be wandering and misleading!
Whether with DSP or with a suitable Method (for passive filters) - a good Step Response should resemble(similarity) this:
1779887393274.png

The complication comes from the fact that Linkwitz filters only have a perfectly flat frequency response, but in the Time Domain they are not optimally good. As is known - they can be compensated with DSP or in another design (with other filters) with separate boxes for each driver and exact distances between the Acoustic Centers (λ/2). With Linkwitz filters this is impossible!
 
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