Is it possible to cover the whole spectrum, high SPL, low distortion with a 2-way?

Hello GM

Clipped?? Why would it be clipped?? Are you talking about driver Qts or Qtc. Both??

Rob :)

Like camplo stated i.e. the driver's signal decays 'faster'/quicker than it's supposed to [AKA under-damped]

Hmm, my lately too often in a hurry responses these days bit me in the a$$, i.e. [AKA under-damped] should be [AKA over-damped] as I originally and recently stated.
 
Hi Dr. Geddes,

If they are good at what they do, and put marketing aside, then they should gravitate to: That it doesn't much matter.

It was my understanding from some of your writings that as long as the system is designed for the desired response - potentially including equalization - it does not matter. I hope that I did not misinterpret you.

But, then there are others, like krivium, who I understand is a professional, who claims otherwise.

Kindest regards,

M
 
Hi fluid,
Is that true?
Yes and no I suppose. The Volume displacement of the driver will set how loud it can play at any given frequency.

Q=0.5 is quite demanding in comparison to Q=0.7, more EQ will be needed to take it to that level if the box was not designed for it. That EQ will cost power.

In simulations like you did, take both drivers to the red excursion line and see how loud they can play at any given frequency.

If you relax the F3 the higher Q can play louder.
 
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But, then there are others, like krivium, who I understand is a professional, who claims otherwise.
I think you might have missed something, Q doesn't matter if you have EQ, to be able to tailor the in room response to whatever you want it to be.

If EQ can't or won't be used for some sort of "purist" reason, the the Q or shape of the LF rolloff matters much more. Krivium's point to me is then that a designer could tailor the sort of likely in room response that an average end user might get by varying the low frequency rolloff slope.

Manufacturers do this all the time, putting a bass bump in bookshelf speakers to sound more impressive or rolling off the anechoic bass response of towers so they won't sound over done in the average living room with some form of room gain.

One huge advantage a DIY'er has is that they can design or tweak to whatever room or space they have without the need to take an educated punt like a manufacturer does.
 
Hi fluid,

. . .

Q=0.5 is quite demanding in comparison to Q=0.7, more EQ will be needed to take it that level if the box was not designed for it. That EQ will cost power.

Well, the models suggest just the opposite. It takes less power to reach the excursion limitation for the lower Qtc enclosure.

What does ". . . if the box was not designed for it" mean in this context?

In simulations like you did, take both drivers to the red excursion line and see how loud they can play at any given frequency.

I am afraid that I do understand your suggestion. The red line is excursion limitation, and as such is a constant. Thus, if I equalize both to reach the line, they will play with the same loudness.

Could you clarify?

Kindest regards,

M
 
Well, the models suggest just the opposite. It takes less power to reach the excursion limitation for the lower Qtc enclosure.
Yes because the extra 50 litres of box is doing the work instead of the amp.

Put any driver in a bigger box and you will need less power to reach xmax.

Look at your own graphs at 20Hz. 2.5 dB before excursion limit at the lower Q and 3.5dB left at the higher Q.

If you were to make the Q's 0.5 and 0.7 as I suggested this difference would increase. Getting an extra 1 to 2dB output requires less power to reach the same SPL.


What does ". . . if the box was not designed for it" mean in this context?
A box designed to have a natural Q of 0.5 vs one with a higher Q that is being corrected to have a lower Q. Trading amp power for box size.
 
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But, then there are others, like krivium, who I understand is a professional, who claims otherwise.

Hi,
I was a pro studio sound engineer, involved in technical side of things too ( maintenance and i set up some studios including acoustics and all the nerd thingy...) and was into education of all this too.

Thanks to this i met professionals into loudspeakers design and was lucky to share with them but in no way i was involved into loudspeaker production. And the reason i'm here to learn. :D

Fluid gave a good explanation of what i meant and how i see things: as diyers we can make choices which follow our listening space/habits/preference rather than have to adapt to a 'medium' path choosen from practical or economical reason ( commercial offers have to made assumptions upon some design target).

Tbh, i don't claims otherwise than what you understood from Earl, rather the opposite. :D
 
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And to be clear Fluid, I, as usual, am playing devils advocate, I am not calling you a liar, looking at Kjeldsen's reply you can see how I might have drawn my conclusions.

You were reffering to changing Q via filter slope. This is not possibke. Maybe I have misunderstood this?. Then I suggest how you can change parameters. Linkwitz transform simulates a new system Q, to be able to match in room response.
 
You were reffering to changing Q via filter slope. This is not possibke. Maybe I have misunderstood this?. Then I suggest how you can change parameters. Linkwitz transform simulates a new system Q, to be able to match in room response.



I sorta agree with you more than Fluid. I also trust Fluid. So why not just discuss the details. Fluid is saying that by matching the 0.5 slope via filter you will have all the other characteristics that come along with an enclosure that naturally achieves said slope....aside from LF efficiency
Qes has an effect when there is no signal
Qms has an effect when there is no signal
The airmass has an effect when there is no signal

A electronic filter has no effect, when there is no signal...wait I just messed that up....some how filters can cause ringing or whatever...


You can't move the mechanical filter with a electronic one....you can't null out the mechanical filter....so I think Kjeldsen is right.

You modify the system which is what matters, that is why Earl is not concerned about those other parameters

Active Filters



It is the same mechanism that allows the removal of time domain ringing by equalization, a big resonant peak that rings can be removed with a corresponding EQ cut with matched Q.

Ringing isn't always proportional to level is my thought on this. You can have two notes 140hz and 600hz both at the same SPL but different decay character....EQ isn't going to allow you "tweak" decay times and leave SPL untouched. You can't undo the ringing at F with a parametric filter. You can high pass it or compound it. Oh wait here we go, this is a trip into linear phase filtering I bet...where we end up with delay on the signal.......

I want to say here that nothing nonlinear can be seen in an impulse response. To find even 2nd order nonlinearity you need a second impulse at some known spacing. From this you can get the second order nonlinearity. Third order requires three impulse in sequence, etc.. So what you are seeing is nothing like what we normally think of as clipping - a highly nonlinear phenomena. It is simply a change in frequency response as an FFT would show you.
I take this as a way of saying that the aspects we are discussing regarding damping and decay are somewhat beyond just the impulse.... Correct me if I'm wrong....in that case one might believe that they are creating 0.5Qtc, using a filter, by looking at solely the first impulse, when in actuality it is not that simple.
 
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You can't move the mechanical filter with a electronic one....you can't null out the mechanical filter....so I think Kjeldsen is right.



Ringing isn't always proportional to level is my thought on this. You can have two notes 140hz and 600hz both at the same SPL but different decay character....EQ isn't going to allow you "tweak" decay times and leave SPL untouched. You can't undo the ringing at F with a parametric filter. You can high pass it or compound it. Oh wait here we go, this is a trip into linear phase filtering I bet...where we end up with delay on the signal.......


I take this as a way of saying that the aspects we are discussing regarding damping and decay are somewhat beyond just the impulse.... Correct me if I'm wrong....in that case one might believe that they are creating 0.5Qtc, using a filter, by looking at solely the first impulse, when in actuality it is not that simple.

You are getting lots of things mixed up.

The Q of the driver and the Q of the final system are different things. The driver Q is fixed (except that it can be raised with a series resistor, but one does not generally want to do that because it wastes power. The final system Q can be manipulated with and electronic filter as has been shown here many times.

Q, damping, decay, are all the same thing just in different language. Damping usually refers to mechanical resistance and EEs prefer to use the synonymous term Q. Decay is just the view of damping in the time domain. They all track one another. Since these terms are all defined for only a linear system, they are all evident in the impulse response, just as ALL linear variables are.

As to the two tones, let's let them both be turned off at the same time, they will decay. The rate of decay will depend on the frequency response at each of those frequencies. We can make the decays the same, or not, by changing the FR at each of the tones. Hence "EQ" can and does change the decay times of these tones.
 
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Ringing isn't always proportional to level is my thought on this. You can have two notes 140hz and 600hz both at the same SPL but different decay character....EQ isn't going to allow you "tweak" decay times and leave SPL untouched. You can't undo the ringing at F with a parametric filter. You can high pass it or compound it. Oh wait here we go, this is a trip into linear phase filtering I bet...where we end up with delay on the signal.......

This make sense. And why it is hopeless to try to get rid of breakup through eq ( and non minimum phase behavior more broadly).

I take this as a way of saying that the aspects we are discussing regarding damping and decay are somewhat beyond just the impulse.... Correct me if I'm wrong....in that case one might believe that they are creating 0.5Qtc, using a filter, by looking at solely the first impulse, when in actuality it is not that simple.

Maybe. Or when you have the impulse AND the frequency response confirming you are in presence of what you should expect for a given Qtc you are trying to reach... maybe you have reached your goal? :)

Off Topic:
Paul7052,
Euphonics i liked them. I never understood why the principle didn't catch up more customers in EU. There was one thing i was bothered with though: it was the rise of 'mixing with your eyes'... Which one have you used? 2000, 3000? System5 maybe? I've seen the 3 and really liked the 3000 ( never used the System5 but bought a 02r which was replaced by it. The engineer told me they had zero difference in sound between the 2 but he couldn't withstand to have the artist explaining him how to use the desk! o_O ) ).
Neve or Ams/Neve? You American seemed to dislike the Ams/Neve. In the studios i worked in the standard setup was Ams/Neve in A, SSL in B. The wealthy one had a vintage Neve for tracking... the others sidecars.

Good memories.
 
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