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First speaker to reproduce a 60hz sq wave

MAB

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Do you agree with the above calculations?
I do!
But perhaps not visualizing it the same way.
I am measuring the relative phase. ctrl mentioned this earlier. To construct a good square wave, you need a region of relatively flat phase response between the fundamental relative to the first few harmonics. The first harmonic at 3x the fundamental is the most important one.

Also, I realize that the measurement I made above was with microphone setup movement between runs.
I redid more carefully to answer a couple remaining questions lingering in my head on my previous data.

First, the nearfield response of a W26 woofer in a sealed enclosure with no PEQ and no phase compensation:
1703989086935.png

There is a limited window between 400Hz and 1kHz with both flat phase and amplitude response on the raw driver with no phase correction. At ~1.2kHz there is a resonance in the driver with a large distortion spike and a wonky dip in the phase response. Below 400Hz is a steadily increasing difference in phase between a fundamental and the related higher order series in the square waveform. Since the first harmonic of a 400 Hz square wave is at 1.2kHz (right in the middle of the phase dip) we expect poorly rendered square waves. If we lower the fundamental frequency, we also expect poor square wave because the fundamental is in the rising region of phase relative to the higher order harmonics. There is no window at the mic position where the fundamental and the first one or two harmonics can fit without a large relative phase offset, as measured at the mic position.
1703985999030.png

As expected the square wave doesn't look very square on the scope. Below 250Hz they get extremely messy. The 250Hz and 400Hz waveforms look somewhat triangular The triangular shape is verified in simulation when the higher order components are offset by 20 degrees as in the bottom plot:
1703988836814.png


The simulation couldn't align better with reality for the 250Hz square wave. The 400Hz is less clear, but the 1.2kHz harmonic is in a distortion region where the driver is not behaving absolutely linear, so all bets are off!

I applied a new FIR phase correction, this time not bothering about woofer equalization, but instead trying to get a maximal region of flatness. I was able to get 80Hz to just below 3kHz fairly flat. Here are the phase response for the driver with and without phase compensation at top, with the measured square waves for the phase-compensated driver below for various frequencies across the sweet spot.
1703990274592.png

You can see that 60Hz is not square with the large triangular shape expected from phase offsets between the harmonics. 80Hz and above is very good. This is satisfying since that's the region I was able to flatten the phase! Starting at 630Hz the behavior becomes less square, which I attribute to the higher harmonics having wild phase and amplitude variation due to the distortion mechanisms of the woofer. Many of these are cone and surround resonances and breakup, so I am not surprised that the waveform is less convincing. Of course I can clean that up with a crossover and EQ filtering out those higher modes.

I hope this helps visualize. And clear up a couple of confusions on my part in my previous post.
 

Adis

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... like gravity and Q.E.D... and no one thinks those are actually "real". :cool:
I have no problems with squares and no problems with waves, just with square waves. Waves have no corners, squares do.
On a graph, a square wave doesn't look like wave to me, it looks like broken coordinates - anything BUT the wave. Shouldn't appear in nature. And it doesn't.
 

Hayabusa

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I may well be wrong in my view, I am not an expert in such physical details.
The difference to your view is difficult to describe and very probably formulated "unphysical" or simply wrongly by me. Will try anyway ;)

You are only looking at the time offset of the "wave front" at the temperature-related different sound velocities c1 (343 m/s at 20°C) and c2 (349 m/s at 30°C).

What also changes is the wavelength at a given frequency:
⁁1 = c1/f
⁁2 = c2/f
So something else is happening.

With the change in speed of sound, the time it takes for the sound wave to reach the microphone changes. This means that the "phasor for each frequency" is at a different "position" when the measuring microphone is reached after 5m at a different air temperature. This then causes the frequency dependent phase shift.

The phase shift ΔΦ (caused by change in propagation time) is then:

ΔΦ = 360° * Δt * f

Δt = d/c1 - d/c2 (d = distance)


Here again how Klippel describes the phenomenon:

View attachment 338310
Source: Klippel GmbH

If an expert like @René - Acculution.com is reading this, it would be nice if you could set the record straight about how wrong I am (before I write any more BS? and have to correct countless posts ;)).

I think Zapper made the right conclusion:

My calculations are all based on let's say an ideal point source. In that case the distance traveled for all frequencies is the same and so if the speed of sound changes by temperature that is also the same change for all frequencies. So all frequency components of the signal coming from the point source will arrive at the same time: waveform will remain the same.

But! In a not ideal sound source each frequency can originate from a slightly different position and can have a different distance traveled to the observer.
In that case if the speed of sound changes this will results in a different delay per frequency component and will distort the waveform.
 

Hayabusa

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I have no problems with squares and no problems with waves, just with square waves. Waves have no corners, squares do.
On a graph, a square wave doesn't look like wave to me, it looks like broken coordinates - anything BUT the wave. Shouldn't appear in nature. And it doesn't.
Indeed a real square wave is not really square
 

Adis

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Indeed a real square wave is not really square
What confuses me equally, if we look at the graph, is that at the same time point (X-coordinate) there are several points of different intensity (Y-coordinate). How to understand that? Apparition? :)
 

Mnyb

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I think Zapper made the right conclusion:

My calculations are all based on let's say an ideal point source. In that case the distance traveled for all frequencies is the same and so if the speed of sound changes by temperature that is also the same change for all frequencies. So all frequency components of the signal coming from the point source will arrive at the same time: waveform will remain the same.

But! In a not ideal sound source each frequency can originate from a slightly different position and can have a different distance traveled to the observer.
In that case if the speed of sound changes this will results in a different delay per frequency component and will distort the waveform.
Thanks this is great, it might by like this I buy this explanation .

It also another argument for an idea I had ...

Here it goes for others to scrutinise:

Your better have a coaxial driver when trying to build a “ phase corrected “ speaker ( or what to call it ). This will improve real world use cases as it’s mimic a point source better . It’s not perfect in that’s its not a point it has dimensions . But it’s not two different non point sources which is worse .
 

kemmler3D

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Thanks this is great, it might by like this I buy this explanation .

It also another argument for an idea I had ...

Here it goes for others to scrutinise:

Your better have a coaxial driver when trying to build a “ phase corrected “ speaker ( or what to call it ). This will improve real world use cases as it’s mimic a point source better . It’s not perfect in that’s its not a point it has dimensions . But it’s not two different non point sources which is worse .
It definitely doesn't hurt, but physical alignment is not the whole story, since there's usually still a crossover which also introduces phase distortion. This is one reason people say digital crossovers are better.
 

Mnyb

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It definitely doesn't hurt, but physical alignment is not the whole story, since there's usually still a crossover which also introduces phase distortion. This is one reason people say digital crossovers are better.

Ok active coax speakers then :)

I’m personally not convinced that it’s very audible and how much so given other compromises.
But phase correction is cheaper and works better in active speakers and may not demand so many other compromises that it makes it a fools errand . ( like true first order slopes in the xover )

Anecdotally something audible happens when I activate the phase correction in my kef LSX , but I don’t have the means to determine if it’s a side effect or the actual phase correction I hear it could be either or both . These things stick together so something with dispersion or fr response may also change when this is active?

And why is it an option and not on all the time if it was unambiguously better ?
 

Hayabusa

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Ok active coax speakers then :)

I’m personally not convinced that it’s very audible and how much so given other compromises.
But phase correction is cheaper and works better in active speakers and may not demand so many other compromises that it makes it a fools errand . ( like true first order slopes in the xover )

Anecdotally something audible happens when I activate the phase correction in my kef LSX , but I don’t have the means to determine if it’s a side effect or the actual phase correction I hear it could be either or both . These things stick together so something with dispersion or fr response may also change when this is active?

And why is it an option and not on all the time if it was unambiguously better ?
With coax speakers I have always asked myself what is the location of the acoustic source of the larger speaker.
is it at the inside where the cone starts at the coil or at the outside or somewhere in between?
It quite essential to know where to place the tweeter exactly...
 

kemmler3D

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And why is it an option and not on all the time if it was unambiguously better ?
It usually introduces some extra delay, which you might not want if you're watching videos or gaming.

For my part when I turn it on/off on my LS60s I can convince myself I hear a difference, but I wouldn't bet much on being able to pass a blind test. You can definitely hear a difference in the moment before the two speakers are both on the same mode, but I bet that's mostly from a slight difference in delay time.
 

Mnyb

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It usually introduces some extra delay, which you might not want if you're watching videos or gaming.

For my part when I turn it on/off on my LS60s I can convince myself I hear a difference, but I wouldn't bet much on being able to pass a blind test. You can definitely hear a difference in the moment before the two speakers are both on the same mode, but I bet that's mostly from a slight difference in delay time.
Maybe the extra delay can introduce problem when integrating sybwoofers externaly , not using the speakers own sub out ?
 

ctrl

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My calculations are all based on let's say an ideal point source. In that case the distance traveled for all frequencies is the same and so if the speed of sound changes by temperature that is also the same change for all frequencies. So all frequency components of the signal coming from the point source will arrive at the same time: waveform will remain the same.

I searched the web and unfortunately found nothing that supports my point of view. However, with my approach I can directly recalculate Klippel's view of the phase shift as the temperature changes.

I found a web seminar by Kippel where he explains the influence of temperature on the phase frequency response in two minutes.
Please @Hayabusa @Zapper take a look at the video and explain to me exactly what he means (link starts at the topic):

KLIPPEL LIVE Series 1 - Part 2: Standard Acoustical Test Performed in Normal Rooms

In the video, he gives the example of "beam stearing" in speaker arrays. We can probably agree that if driver A is measured at 20°C and driver B on top at 22°C, then there will be a phase error due to the change in the speed of sound which will prevent correct beam stearing especially at high frequencies.

The sound from driver B arrives a little earlier (when assuming same Mic distance as A) at the mic and the higher the frequency, the greater the phase error in degrees - I showed how this can be calculated in post#98 and post#89.
Or do we disagree at this example?
 
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fpitas

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fpitas

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What confuses me equally, if we look at the graph, is that at the same time point (X-coordinate) there are several points of different intensity (Y-coordinate). How to understand that? Apparition? :)
A real square wave is a mathematical abstraction. To actually make one you would need infinite bandwidth.
 
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