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Analytical Analysis - Displacement vs Sound Pressure

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René - Acculution.com

René - Acculution.com

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Yikes. Some Gefell measurement mics had flipped polarity for years and that has created a lot of confusion. It became standard procedure for me to test mic chain polarity by popping a ballon (my breakfast paper bag ;-)

As for misconceptions, a very annoying one is the "inverse square law" for sound pressure vs. distance, when it's actually only 1/r. Even renowned text books have this wrong, confusing pressure with intensity.

And for resources, the website of Eberhard Sengpiel (RIP) is a classic -- really a jewel -- with very accurate and condensed info without bloating math. It's main focus is on recording techniques and microphones but of course much of this also directly transfers to speakers.

Best entry point for non-german readers is via the google search presets: http://www.sengpielaudio.com/Searchengine.htm
Also there are many online calculators: http://www.sengpielaudio.com/Calculations03.htm
Yes, that is actually a good topic; inverse laws, far-field vs near-field "stuff". Thanks.

Also, https://www.acs.psu.edu/drussell/demos.html is a good resource.

And Erin and I will probably do a video soon on Beaming/Radiation from a source.
 
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Snarfie

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You should always be careful when carrying around knowledge that came out of intuition or “it is well-known in this industry that…”-thinking, where you trust what is being said within the company that you are in. Once in a while, it can affect the engineering decisions in ways that can cost a lot of time and money to course correct once the problems show up. Try and work from first principles whenever you can.


We are discussing on ASR that phase/time alignment can be resolved with DSP/room correction. Speaker brands like Thiel, Vandersteen an others came up with hardware solutions to solve the Phase/time alignment problem using specific designed drivers, slope designed baffles an first order crossovers which are a pain to get it wright.

I don't know what to belief. Thing is i own Vandersteen who are from design Phase/time aligned on top of that they are corrected for Phase/time by my DPS. When i listen to same sort of far-field speakers who are not Phase/time aligned by design i expect after DSP correction that both sound more or less the same for instance regarding depth/staging. Still after DSP correction i hear a big difference between the two different designed speakers.
An no it is not subjective the difference is IMO day an night. Could be that I did not have the opportunity to listen to the latest speakers who could sound differently.

What is your opinion possible empirical/theoretical experience on both.
 

storing

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Vandersteen who are from design Phase/time aligned on top of that they are corrected for Phase/time by my DPS
The topic is on (air) pressure in relationship to the position of the speaker membrane; simply speaking time aligning something doesn't change the shape/amount or pressure of that driver's membrane, it just shifts the complete wave in time (which is what you are hearing i.e. yes fixing alignment between multiple speakers which is what I assume you mean can sound better because reproduced sound is closer to original), but that doesn't change that at a given point in time when the membrane is at max. displacement the pressure is negative.
 
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René - Acculution.com

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We are discussing on ASR that phase/time alignment can be resolved with DSP/room correction. Speaker brands like Thiel, Vandersteen an others came up with hardware solutions to solve the Phase/time alignment problem using specific designed drivers, slope designed baffles an first order crossovers which are a pain to get it wright.

I don't know what to belief. Thing is i own Vandersteen who are from design Phase/time aligned on top of that they are corrected for Phase/time by my DPS. When i listen to same sort of far-field speakers who are not Phase/time aligned by design i expect after DSP correction that both sound more or less the same for instance regarding depth/staging. Still after DSP correction i hear a big difference between the two different designed speakers.
An no it is not subjective the difference is IMO day an night. Could be that I did not have the opportunity to listen to the latest speakers who could sound differently.

What is your opinion possible empirical/theoretical experience on both.
Theoretically, with first order filters combined with "wide passband" drivers that can handle the power and excursion, you can set up a two-way system, that combines acoustically to a flat response with zero phase. You then need to align the drivers physically so that there is no phase difference due to distance. I don't know how well this can be done, but do remember some old measurements, where seemingly the time domain aspects of such speakers were quite good (impulse response, square wave response).

For a general speaker, you should not "distance align" willy-nilly, without considering the phase response of the drivers and the cross-over first.

Since a speaker is generally a non-minimum phase system (it is a sum of several (generally) minimum-phase systems; each driver with its crossover part), it is not directly 'equalizable', since we run into trouble with stability and causality, when we try to invert such system. Causality issues imply that we need to be able look forward in time, and we typically think of systems as being causal. But with DSP, we can take in inputs for a while and hold them in a buffer, and thereby form a new timeline from which we start, and have the output be dependent on both previous input ("new zero time") and the recent data ("future signals"), and correct, albeit with an overall time delay as a result, as can be seen in the measurements on e.g. KEF LF50 Active and Kii Three.

Erin showed recently that the amplitude response is also affected (https://www.erinsaudiocorner.com/loudspeakers/kii_three/), which makes it a little difficult to compare even a single speaker with itself in different settings, when you are only interested in the temporal difference. So comparing two speakers with each other, I would probably not expect them to sound the same, since they have different amplitude responses(?) and raditation patterns.

A lot of journal papers show that having optimized transient/temporal behavior is not worth pursuing, and that was what I just assumed. But listening to the Kii Three's myself when visiting Purify at some point in time where Bruno was visiting, I changed my mind, as I in a blind test could quickly pick out which setting it was in, without having ever listened to them before (and generally starting out thinking how to put in a nice way, that I could not hear a difference). Granted, the track (that I also had never heard) had a certain characteristic, where a lot of instruments had to fall in place at the same time, and somehow I picked up on that particular second or less, and right there was a difference.

Time alignment via displacing drivers will lead to big differences in diffraction and so it is best for comparison purpose to do these things with DSP.

Don't know if that answered anything...
 
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charleski

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When you measure the pressure phasor, its phase relates back to the phasor phase of the input voltage. So we typically have zero phase voltage as reference and whatever pressure phasor phase we happen to have; the one shown in data sheets (if it is even shown). Some microphone/preamp combos flip polarity, and I imagine that this is compensated for. The driver being a minimum-phase component, you know that it should be zero phase somewhere in the pass-band region. Comparing the pressure phase to the phase of the driver requires you to take out the part of the overall phase coming from the distance travelled by the sound to the mic, but this is typically also done in the software. And there you will see that pressure is in-phase with acceleration, not displacement (under the conditions stated). So if the microphone retains phase (positive pressure->positive electrical output) then you will have no polarity flip there, and for the microphone you have the membrane moving into a small chamber, so there pressure will relate to displacement(!) typically, and so that is probably what you are asking for. But the internal mic setup and the preamp will determine if there is an overall sign flip, or not.
Wouldn't there be a difference between using a ribbon mike (which is sensitive to air velocity) compared to a condenser mike (which is sensitive to pressure)?
 
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Wouldn't there be a difference between using a ribbon mike (which is sensitive to air velocity) compared to a condenser mike (which is sensitive to pressure)?
You should always look in the data sheet for the microphone if there is a phase change between the pressure and the electrical output.
 
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I don't think so, rather the contrary. Care to elaborate where you think he's mixing up things?
Let's take that in a separate thread, as this is a topic, where people are not in agreement, and it could blow up :)
 

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Hi Rene I have to skew on your reply I'm by far not a audio engineer so bear in mind:facepalm:.

But what I understand reading your reply is:

“Theoretically, with first order filters combined with "wide passband" drivers that can handle the power and excursion, you can set up a two-way system, that combines acoustically to a flat response with zero phase. You then need to align the drivers physically so that there is no phase difference due to distance. I don't know how well this can be done, but do remember some old measurements, where seemingly the time domain aspects of such speakers were quite good (impulse response, square wave response).“

So it is difficult to get the job done from a hardware point of view. I saw in stereophile the Step response graphs from a Thiel CS3.7 which were quite good compared to speakers that are by design not phase time coherent. https://www.stereophile.com/content/thiel-cs37-loudspeaker-measurements

"Time alignment via displacing drivers will lead to big differences in diffraction and so it is best for comparison purpose to do this things with DSP."

Does that mean when other speakers who use DSP end up with the same step response results as speakers that are by design phase/time coherent build?

If thats the case by measuring the step response you prove that DSP can make a speaker phase/time coherent. Question that is left over: To experience soundstage/Imaging does speakers need to be not only phase/time coherent but need also other values/properties.




.
 
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charleski

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Wouldn't there be a difference between using a ribbon mike (which is sensitive to air velocity) compared to a condenser mike (which is sensitive to pressure)?
Actually, to answer my own question, it would probably depend on where the measurement was taken, as in the far-field the wavefront tends towards a planar wave in which the pressure depends directly on particle velocity.
 
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René - Acculution.com

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Hi Rene I have to skew on your reply I'm by far not a audio engineer so bear in mind:facepalm:.

But what I understand reading your reply is:

“Theoretically, with first order filters combined with "wide passband" drivers that can handle the power and excursion, you can set up a two-way system, that combines acoustically to a flat response with zero phase. You then need to align the drivers physically so that there is no phase difference due to distance. I don't know how well this can be done, but do remember some old measurements, where seemingly the time domain aspects of such speakers were quite good (impulse response, square wave response).“

So it is difficult to get the job done from a hardware point of view. I saw in stereophile the Step response graphs from a Thiel CS3.7 which were quite good compared to speakers that are by design not phase time coherent. https://www.stereophile.com/content/thiel-cs37-loudspeaker-measurements

"Time alignment via displacing drivers will lead to big differences in diffraction and so it is best for comparison purpose to do this things with DSP."

Does that mean when other speakers who use DSP end up with the same step response results as speakers that are by design phase/time coherent build?

If thats the case by measuring the step response you prove that DSP can make a speaker phase/time coherent. Question that is left over: To experience soundstage/Imaging does speakers need to be not only phase/time coherent but need also other values/properties.




.
Yes, some have managed to do it without DSP. But one has to wonder if it comes at a cost; are the drivers being strained too much and so on. It all comes down to how much you value the temporal behavior.

Doing it with DSP makes it possible to compare with and without alignment, whereas that is difficult doing it the Thiel way, because you would compare to a very differently built enclosure.

I can only speculate. I think there is more to having an optimized phase, than some companies seem to think, and with more and more of them deploying DSP anyway, the will probably start adding different phase options. Then at least you can compare (provided that the amplitude response is fairly similar for the different settings.)
 
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René - Acculution.com

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Actually, to answer my own question, it would probably depend on where the measurement was taken, as in the far-field the wavefront tends towards a planar wave in which the pressure depends directly on particle velocity.
Yes, so I have conviently taken the phase part coming from the field out of this discussion. There is a phasor phase (the most important one by far), there is a phase dependency from time passing, and then there will be a phase related to the propagation when looking at waves/fields.
 

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I have not come upon a good site or book that explains things concisely without getting too much into the mathematical details, and it is tough to do. I have started making videos on Engineering Mathematics on YouTube, but it is difficult to find the time, and there will be a lot of math there too. Most books or videos that tackle complex topics in a simple manner revert to intuition and analogy, and so get some important details wrong. One topic which should not be an issue at all for example, is Phase vs Polarity; most videos and articles on this topic are wrong
Thank you very much for answering. I study mechanical engineering, so math stuff is not an issue, the issue is people jump too quick without explaination, and bring out unknown physics or math like magic. I think using analogy, graphs and example is really a good way to teach, text along really is not enough, I tend to forget or misunderstand what people write when they have long sentences. What I am lacking now is the simpler manner way of teaching, even if they are not strictly correct, it will actually be fine with me right now. I just need something to grasp to get myself at least going. If you have some of those easier stuff please let me know. I will check out your videos too.
I would like to have more time to post things, but it is difficult with a newly started company. One thing that I would like is something where many people could gather and discuss things on-line, but with one or more moderators. So that topics like the one discussed in this post could be closed once and for all, and the discussions could serve as a repository for these questions that are discussed on forum over and over again (room modes, room gain, polarity, ...).
This is great, I hope we can do more fundamental stuff. There are just too much in acoustic that I can never learn enough! What does your company do? Do you do teaching on the acoustic or making things for other people? Hope your buisness is going great.
 
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Thank you very much for answering. I study mechanical engineering, so math stuff is not an issue, the issue is people jump too quick without explaination, and bring out unknown physics or math like magic. I think using analogy, graphs and example is really a good way to teach, text along really is not enough, I tend to forget or misunderstand what people write when they have long sentences. What I am lacking now is the simpler manner way of teaching, even if they are not strictly correct, it will actually be fine with me right now. I just need something to grasp to get myself at least going. If you have some of those easier stuff please let me know. I will check out your videos too.

This is great, I hope we can do more fundamental stuff. There are just too much in acoustic that I can never learn enough! What does your company do? Do you do teaching on the acoustic or making things for other people? Hope your buisness is going great.
That is fair, we all learn in different ways. There are so many things to consider, and there are always some bottlenecks; the acoustic engineers don't know enough solid mechanics, the mech engineers don't know enough acoustics, and they both should really know a lot of signal processing and also electrical engineering if nothing else than for electroacoustic analogy circuits. One has to look across many different textbooks and perhaps videos, and try and bring it all together somehow...

I do simulations on solid mechanics, acoustics, and electromagnetics for companies as a consultant, looking at electrodynamic drivers, electrostatic headphones, microphones, vinyl pickups, and other products. And then research alone and with students regarding microacoustics, shape optimization, topology optimization, and try to bridge academia and industry work. And then blog post and hopefully some more videos, to have some repository to point my students and clients too. Very busy...
 
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Thank you for the original and very educational post.
Off this topic, but since you're here, may I ask another question? I remember you did simulations about waveguide screws. ... How much discontinuity will be noticeable?
The modal discontinuities discovered w/such screws; brings up a real world application question, based on the following description:
"Turbine Cone Woofer - Our pioneering turbine design delivers a significant increase in cone rigidity without increasing mass. This innovation means the woofer can move with increased accuracy and speed to deliver enhanced sound resolution, enabling you to experience a breathtaking level of detail in your music and movies."
Snag_50a3fff3.png
Would the 7 'dimples' on the surface of this Mid speaker cause similar problems as the "screws" in question?
 

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Thank you for the original and very educational post.

The modal discontinuities discovered w/such screws; brings up a real world application question, based on the following description:
"Turbine Cone Woofer - Our pioneering turbine design delivers a significant increase in cone rigidity without increasing mass. This innovation means the woofer can move with increased accuracy and speed to deliver enhanced sound resolution, enabling you to experience a breathtaking level of detail in your music and movies."
View attachment 171122Would the 7 'dimples' on the surface of this Mid speaker cause similar problems as the "screws" in question?
This would not be an issue as such, as the wavelengths here for the woofer are much longer and it is also not a diffraction problem per say. The added rigidity is well worth it I would imagine.
 
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