Ask an Acoustic Engineer (me)

I haven't tried making an air pump like this, but I think you'll be blasting bass into whatever is at the end of the air pump if you use 30hz. That sounds uncomfortable. Maybe try really low frequencies, like 0.25hz.

For frequencies less than 1/5 of free-air resonant frequency fs, and for small displacements:

Xvc,peak = (Vrms * BL / Re) * Cms * (1.414/1000)

Example:
BL=14.8 tesla*meter
Vrms = 2 volts ac rms
Re = 4.03Ω
Cms = 0.126 mm/N

Xvc = (2 * 14.8 / 4.03) * 0.126 * (1.414/1000)
Xvc = 0.0013 meters
Xvc = 1.3mm peak, one-way

This assumes no restrictions from air pressure in the mechanism you make, and it assumes mounted infinite baffle. The actual amount of air pumped will decrease with those realities.

You can turn this into air displaced per second by adding frequency and surface area of the cone. For an air pump, I think you need to multiply the result by two since a full stroke is peak-to-peak, whereas Xvc is peak one-way.

If you do something amazing, please share?
 
Hey Justin, thanks for doing this. I asked this in a separate post but would love your thoughts; what are people missing out on when they only use a single USB microphone when using a RTA for tuning? Assuming they follow a good process of starting with each driver, then driver pairs, etc and are moving the microphone to average out measurements? What information is being missed that could be helpful, if any? And what sort of things can be done to overcome these shortcomings? Just wondering what some of the technical aspects are with these sort of measurements.


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Hello mauian, and you're welcome!
Thank you for the question.

An RTA that uses pink noise will not capture any time-related information such as phase, impulse response, group delay, waterfall, and so on. Time information adds another layer of complexity to what you're doing, but also another layer of information you can use to your advantage. The thing I use time information for the most is setting time delays and polarity and ensure each speaker is summing together correctly.

You can use an RTA to see phase interaction near a crossover point but you can only see if two speakers sum together nicely or if they fight eachother. If they fight eachother, you cannot see which one has a phase response you prefer so you don't know which one to try and adjust.

Time information can be acquired using the acoustic timing reference in REW if you have a USB microphone, however I do not have much experience with that technique. I prefer to use a two-channel sound card with a loop-back cable for the timing reference which seems more reliable. You'll need to use the "measure" button to do a sweep to get time information.

A relatively new feature in REW is the ability to vector-average measurements to reduce the influence of reflections and background noise. This method requires phase/timing information so you must use the measure/sweep method with some kind of timing reference, and that could be more challenging with a USB mic.

Something else you miss out on with a simple USB mic is tons of headache trying to connect everything and settings that are initially confusing and you generally not being productive for a while. The learning curve is steeper, but it comes with more potential benefits too. Very double-edged sword kind of thing.
 
Hello mauian, and you're welcome!
Thank you for the question.

An RTA that uses pink noise will not capture any time-related information such as phase, impulse response, group delay, waterfall, and so on. Time information adds another layer of complexity to what you're doing, but also another layer of information you can use to your advantage. The thing I use time information for the most is setting time delays and polarity and ensure each speaker is summing together correctly.

You can use an RTA to see phase interaction near a crossover point but you can only see if two speakers sum together nicely or if they fight eachother. If they fight eachother, you cannot see which one has a phase response you prefer so you don't know which one to try and adjust.

Time information can be acquired using the acoustic timing reference in REW if you have a USB microphone, however I do not have much experience with that technique. I prefer to use a two-channel sound card with a loop-back cable for the timing reference which seems more reliable. You'll need to use the "measure" button to do a sweep to get time information.

A relatively new feature in REW is the ability to vector-average measurements to reduce the influence of reflections and background noise. This method requires phase/timing information so you must use the measure/sweep method with some kind of timing reference, and that could be more challenging with a USB mic.

Something else you miss out on with a simple USB mic is tons of headache trying to connect everything and settings that are initially confusing and you generally not being productive for a while. The learning curve is steeper, but it comes with more potential benefits too. Very double-edged sword kind of thing.

I did not expect this sentence... “Something else you miss out on with a simple USB mic is tons of headache.” It really made me LOL. Thanks for that and all the other good info!


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First off thanks for sharing your knowledge, pretty cool of you. I have a couple questions that aren't near as technical, at least I don't think they are.

I have a question about sensitivity and xmas. I've heard, well read multiple post online about certain drivers having low excursion but that it was okay due to the high sensitivity, mostly about some midbass drivers. Is there any merit to this? I've always been under the impression that if you had 2 drivers of the same SD but different sensitivity that they would have to move the same amount to produce the same spl at the same frequency and the only difference would be the power used. Just curious if there is something I'm missing because the people that stated this seemed knowledgeable and I can't understand while the xmax wouldn't matter simply because of high sensitivity.

Second quick question because I suck at searching, can't find relevant info. I've read before that everytime you double woofers you lower distortion but I can't remember by how much was thinking 6 or 12dbs. Have 4 6.5 subs that I intend to use, a pair up front and a pair in the back both in manifold configurations. Just wondering if they will be up to par distortion wise with the rest of my system as I will probably lowpass around 150hz at 24db.
 
First off thanks for sharing your knowledge, pretty cool of you. I have a couple questions that aren't near as technical, at least I don't think they are.

Hi there Sandman, and welcome to the forums!

I have a question about sensitivity and xmas. I've heard, well read multiple post online about certain drivers having low excursion but that it was okay due to the high sensitivity, mostly about some midbass drivers. Is there any merit to this? I've always been under the impression that if you had 2 drivers of the same SD but different sensitivity that they would have to move the same amount to produce the same spl at the same frequency and the only difference would be the power used. Just curious if there is something I'm missing because the people that stated this seemed knowledgeable and I can't understand while the xmax wouldn't matter simply because of high sensitivity.

This is a tough one to answer without some context. Can you show where you found these claims so we can read the conversation too? There are a few ideas tangled together and I want to make sure I understand the context.

Second quick question because I suck at searching, can't find relevant info. I've read before that everytime you double woofers you lower distortion but I can't remember by how much was thinking 6 or 12dbs. Have 4 6.5 subs that I intend to use, a pair up front and a pair in the back both in manifold configurations. Just wondering if they will be up to par distortion wise with the rest of my system as I will probably lowpass around 150hz at 24db.

I haven't heard that rule of thumb yet. It might work if you add more woofers and then play the same overall volume level as before. This way each woofer is moving a little less so each woofer will have a little less non-linear distortion since it is staying closer to the rest position. I don't think the amount of distortion reduction can be estimated simply like you suggest since there are so many variables involved.

Are you maybe confusing the other rule of thumb where doubling cone area and also doubling power can result in +6dB acoustic? This one would also be easier if you share where you read this claim so we can read it too.

Distortion in a subwoofer is usually caused be making the cone move too far. Cone movement increases with loudness and also with lower frequencies. If you install those subs and you think there is too much distortion, try adding a high-pass filter at 25hz, 30hz, or 35hz to limit the excursion at the lowest frequencies and then they can play the other frequencies a little louder while still behaving. This is also called a sub-sonic or infra-sonic filter and many amplifiers have them built-in.
 
99% its a dumb question but would you hear a difference if you were listening to a speaker that was spinning? Spinning as in you were still seeing the front of the speaker at all times.
 
I wonder if you could help me with this question...

https://www.caraudiojunkies.com/showthread.php?4171-A-pillar-mounting-for-Aurasound-Whisper

...seeing as I haven't made any practical progress yet, due to work

Thanks in advance
You could rout some baffles with maybe a 10mm depth out of 18mm wood, oversize the outside and them cut some 9mm ply or mdf to a ring that is 1mm smaller to allow for grill cloth and make a press fit grill to suit (if trimming the pillars in vinyl allow 2.5mm) if trim painting the 1mm is good, spray the baffle area with paint forst so it colours it, then mask off the recess and trim paint the rest of the pillar, remove masking, fit speaker, fit trimmed grill

i would round the inner edges if you can do so safely and put a small round on them, however you are dealing with tiny work pieces, I think I’d cut the outer edge of the grill nearly all the way through leaving 0.5mm of material, then cut the inner all the way through and then while still attached to the bigger piece of material use that as a handle to cut a small radius on the inside, Then cut the outer edge with a Stanley knife and remove any paper thin excess with sandpaper

get a grill cloth that matches the interior and a vinyl or trim paint that matches also
 
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99% its a dumb question but would you hear a difference if you were listening to a speaker that was spinning? Spinning as in you were still seeing the front of the speaker at all times.

I like this question!

I don't think there would be any difference if the speaker is spinning at a slow speed like 1 revolution per second, and if you are directly on-axis. If you were off axis by quite a bit then you might (maybe?) hear a kind of warbling in the higher frequencies where cone breakup occurs. I think this would be true since cone breakup modes are not always symmetric with respect to rotation.

If you were listening to a horn then you would certainly hear a difference if you're off-axis since horns are usually designed with a very specific vertical and horizontal off-axis response that are different. For example horns for live sound can have a wide horizontal response and a narrow vertical response.
 
I like this question!

I don't think there would be any difference if the speaker is spinning at a slow speed like 1 revolution per second, and if you are directly on-axis. If you were off axis by quite a bit then you might (maybe?) hear a kind of warbling in the higher frequencies where cone breakup occurs. I think this would be true since cone breakup modes are not always symmetric with respect to rotation.

If you were listening to a horn then you would certainly hear a difference if you're off-axis since horns are usually designed with a very specific vertical and horizontal off-axis response that are different. For example horns for live sound can have a wide horizontal response and a narrow vertical response.

Neat, thanks for answering.

Another question. Does the shape of a sealed enclosure matter as long as the airspace needed is provided? I know that the amount of air inside is supposed to act like a spring but i feel like the shape of an enclosure could have pressure zones within it. For example, if one were to build a kickpanel enclosure that had enough room to set the speaker into and then the enclosure followed the left of the driver seat and then led to a large box underneath the seat. Technically, you would have enough airspace for a driver but there would be sections of small, tubular, and large air spaces all linked.

And on that note. When it comes to kicks venting outside of the vehicle. Is there a simple way of understanding how big of a hole is needed? I had 10"s drivers in kicks that were vented through the firewall by 2 4in circular holes. I still had a huge rise in the FR that would be reminiscent of an undersized enclosure. I'm curious if there is any math or lamens terms to understanding what realistic venting is needed? The only way i can think of is using a DATS device(https://www.parts-express.com/dayto...er-based-audio-component-test-system--390-806) to test QTC of a driver in free air and then adding more or larger holes in the firewall until the qtc of the enclosure matched.
 
Neat, thanks for answering.

Another question. Does the shape of a sealed enclosure matter as long as the airspace needed is provided? I know that the amount of air inside is supposed to act like a spring but i feel like the shape of an enclosure could have pressure zones within it. For example, if one were to build a kickpanel enclosure that had enough room to set the speaker into and then the enclosure followed the left of the driver seat and then led to a large box underneath the seat. Technically, you would have enough airspace for a driver but there would be sections of small, tubular, and large air spaces all linked.

And on that note. When it comes to kicks venting outside of the vehicle. Is there a simple way of understanding how big of a hole is needed? I had 10"s drivers in kicks that were vented through the firewall by 2 4in circular holes. I still had a huge rise in the FR that would be reminiscent of an undersized enclosure. I'm curious if there is any math or lamens terms to understanding what realistic venting is needed? The only way i can think of is using a DATS device(https://www.parts-express.com/dayto...er-based-audio-component-test-system--390-806) to test QTC of a driver in free air and then adding more or larger holes in the firewall until the qtc of the enclosure matched.
I’m sure he will come along and chime in, but yes if there is a restriction then it will have an effect on the sound of the driver as it will effectively load up the driver and restrict airflow a fair bit

as for the last part, I do know of people who use a dats for that exact purpose to test the Q of enclosures and also IB drivers as well ���� So it’s certainly a thing
 
Ok, I have an easy one for you (I think?). A lot of people seem to look "down" on 6x9 speakers. Personally, I've found that they make fantastic midbass speakers. What are you thoughts on "non-round" speakers? To me, for something like midbass, a 6x9 seems perfectly fine and has the advantage of more cone area so it can provide deeper bass than 6.5" midbass speakers.

Just curious what an engineer things about 6x9s. :-)

Thank you!
 
Ok, I have an easy one for you (I think?). A lot of people seem to look "down" on 6x9 speakers. Personally, I've found that they make fantastic midbass speakers. What are you thoughts on "non-round" speakers? To me, for something like midbass, a 6x9 seems perfectly fine and has the advantage of more cone area so it can provide deeper bass than 6.5" midbass speakers.

Just curious what an engineer things about 6x9s. :-)

Thank you!

The only real downside (that I know of) to a 6x9 is that you have a 6" speaker properties and a 9" speaker properties so things change based on how you orient the speaker in the door.
 
The only real downside (that I know of) to a 6x9 is that you have a 6" speaker properties and a 9" speaker properties so things change based on how you orient the speaker in the door.
But would that really matter for the freqs that the 6x9 speakers play (400hz and under, in my case)? I can't imagine that "turning" the speaker would really change the sound at all?
 
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But would that really matter for the freqs that the 6x9 speakers play (400hz and under, in my case)? I can't imagine that "turning" the speaker would really change the sound at all?

No, because the 1/2 wavelength of the longest dimension is still > 400hz. 13500/2/9 = 750hz. (Speed of Sound ÷ 1/2 wavelength ÷ 9 inches)
 
I like this question!

I don't think there would be any difference if the speaker is spinning at a slow speed like 1 revolution per second, and if you are directly on-axis. If you were off axis by quite a bit then you might (maybe?) hear a kind of warbling in the higher frequencies where cone breakup occurs. I think this would be true since cone breakup modes are not always symmetric with respect to rotation.

If you were listening to a horn then you would certainly hear a difference if you're off-axis since horns are usually designed with a very specific vertical and horizontal off-axis response that are different. For example horns for live sound can have a wide horizontal response and a narrow vertical response.

Well it may not be heard, but I think it would the same as transverse Doppler.
A fellow I know has tried explaining it to me a few times... I remind him of his dog, with its head cocked over.

http://www.conspiracyoflight.com/Transverse_Doppler_Effect.html
 
Neat, thanks for answering.

Another question. Does the shape of a sealed enclosure matter as long as the airspace needed is provided? I know that the amount of air inside is supposed to act like a spring but i feel like the shape of an enclosure could have pressure zones within it. For example, if one were to build a kickpanel enclosure that had enough room to set the speaker into and then the enclosure followed the left of the driver seat and then led to a large box underneath the seat. Technically, you would have enough airspace for a driver but there would be sections of small, tubular, and large air spaces all linked.

Hey you're welcome!

I've had some debate about this with other engineers and it's fun to think about. I believe you're right that given enough "small" passages and air chambers then a sealed enclosure can start to behave differently. The pressure zones you mention could be thought of like standing waves or resonances in the air chamber. Below a certain frequency the sealed box (and all of the connected passages and whatnot) will behave like a pressure vessel that is very evenly pressurized. Above that certain frequency, the air will be able to slosh around and compress and expand unevenly which I think is the "problem" you're asking about.

What if we built a box that had small air passages and air chambers that were meant to be restrictive like this on purpose? Could we learn something about it? Sure! A 4th-order bandpass box has an air chamber and a narrow passage in front of the cone. A 6th-order bandpass box has more air chambers or ports. More complex enclosure designs have even more chambers and ports.

All of those fancy enclosures have something in common too: multiple humps in the electrical impedance, such as what the Dayton DATS can measure. If you have one, you can easily see if your enclosure is working the way it should be by sweeping the impedance and counting the number of humps in the response.

The graph below is a 4th-order bandpass box. See the two humps? Those are obvious.

View attachment 9982

The impedance graph below is ... I assume a sealed box or maybe a 2-way system. There is one large hump around 150hz and a smaller one around 60hz. Do you see the smaller one? That is an additional smaller hump that, in the context of this question, could be a misbehaving air chamber or small passageway (maybe). I can't say for sure since I haven't done this kind of experiment, but it's what I imagine it could look like.

View attachment 9981

So yeah, your "sealed" box could misbehave but it's really hard to predict or notice unless you have a tool like making an impedance sweep.

Here's a fun experiment I just tried. Model a woofer in a ported box with an enclosure that is very small like 0.01cuft. Then add a port 10in in diameter and 1in long. The frequency response won't make any sense but look at the impedance. You should see one peak in the impedance somewhere around the fs of the driver like 50hz or whatever. This is kind of like having one large air passage in your sealed box. Kinda.

Then reduce the port diameter to something smaller like 3in dia and make it longer like 10in. Look at the impedance graph again. You might see a tiny 2nd peak emerging higher in the frequency rang elike 300hz or something. This is similar to having a smaller air passage in your sealed box that goes somewhere else in the car.

Then reduce the port to a small diameter like 1in and make the length longer like 20in. Now you should certainly see two peaks in the impedance plot. This is like if you have a small restricted passage in the "sealed" box to somewhere else in the car. The extra hump in the impedance plot is showing the other resonance in the system.

The above experiment is not really how things would behave since subwoofer modeling programs aren't designed to model a labyrinth sealed box like we're talking about, but it's close enough to the idea that it's fun to play with.

Are there some rules of thumb we can use to make a "good" sealed enclosure? Maybe. When I was dreaming about installing midbass drivers in my kick panels and venting them to the outside I kept reading people recommending certain things. The common wisdom was something like "keep the air passage no smaller than half the size of the woofer". This seems like a good plan. If you don't choke down the air passage on the rear of the cone too much, then it will be able to breath to the atmosphere and you'll have properly vented kick panels. The same concepts should apply to the narrow air passages and chambers if you want to make a labyrinth sealed enclosure too. See the question below for a more exact formula:

And on that note. When it comes to kicks venting outside of the vehicle. Is there a simple way of understanding how big of a hole is needed? I had 10"s drivers in kicks that were vented through the firewall by 2 4in circular holes. I still had a huge rise in the FR that would be reminiscent of an undersized enclosure. I'm curious if there is any math or lamens terms to understanding what realistic venting is needed? The only way i can think of is using a DATS device(https://www.parts-express.com/dayto...er-based-audio-component-test-system--390-806) to test QTC of a driver in free air and then adding more or larger holes in the firewall until the qtc of the enclosure matched.

I kinda answered it above by happy accident, but yes using something like DATS is immensely powerful. Measure the Qts of the driver in free air or on a flat baffle with no enclosure in front or behind. Then install in the car. If your air passages are large enough then the Qts of the woofer in the car should closely match the Qts when in free-air. The Qts in the car might be a little higher than free-air no matter what you do, but a substantial increase would be easy to detect.
 
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