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Ask an Acoustic Engineer (me)
This is an experiment where you can ask questions and an acoustic engineer will try to answer them (me).
I do not mean I am the best person to answer your question ... there are many talented people here who can help! What I mean is, I wish I knew someone like me earlier. I want to be that resource for you. You can read my professional bio over here.
I welcome questions about working in acoustics as a career, or how to go to school to prepare for this field, what enrolling in a master's degree for acoustics is like, or anything similar. I can also speak a little bit about the tools available to professionals.
I also hope this can be a place to ask those challenging questions you've been unable to find an answer to, or perhaps it is hard to find a definitive answer to something. I also hope that I can learn from you since every time I teach something, I learn something too. I'm inspired some by Andy Wehmeyer and his relentless engagement and helping people, and I am inspired by my audio mentor who has guided me along the way.
What would you like to learn?
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Re: Ask an Acoustic Engineer (me)
Why do speaker drivers use a voice coil and permanent magnet instead of opposing electromagnets?
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Re: Ask an Acoustic Engineer (me)
I'm curious about the typical speaker and subwoofer design. If sound is caused by vibrations, why are the devices producing those mostly round and slightly concave? Is this optimal for acoustics, or perhaps a compromise of strength, power handling, material cost, etc...
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Re: Ask an Acoustic Engineer (me)
Chithead, if you mean round vs say the kicker square cones then the issue is boundaries. A round sub has one boundary to keep on the same plane for smooth waves to develop from. If you look at Kicker’s square cones, they have to keep excursion down since it is harder to keep the square cone in the same plane, which would develop the cleanest waves. Plus a square potentially has 4 boundaries corners with each having the opportunity to be slightly out of sync if not perfectly in the same plane, which means a less clean audio wave since it isn’t one clean wave but potentially 4 waves slightly distorted from each other (corners). That’s the audio enthusiast answer and I’d be interested in how Justin can correct me, not that correction is the point. But that is a good question and I’d be interested to know how much further down the deep audio hole Justin could take it.
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Re: Ask an Acoustic Engineer (me)
I would also think the concavity of a speaker is due to space limitations. If it was flat or convex, this would then mean that the speaker needs more space in front of it to be able to move. Ultimately it is about how much air can be pushed/moved and how much space it needs.
I would also think the concavity of a speaker has to be important in some way to the way it is moving the air, concave would mean that all of the air is being sent straight out, convex would send the air out in more drastic angles, and flat would match the in/out angles, but I could be wrong about that reasoning since I am basing that off of more how light works.
https://image.slidesharecdn.com/curv...?cb=1390695812https://lh3.googleusercontent.com/pr...JiL-g7g8qYfGw9
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
BigAl205
Why do speaker drivers use a voice coil and permanent magnet instead of opposing electromagnets?
I've wondered this too! There might be other reasons but the ones I can think of are:
You would add another source of heat by having a second electromagnet, and speakers don't usually need more heat.
Also creating a magnetic field from a second electromagnet would be another source of energy loss, making the inefficient loudspeaker even less efficient.
A second voice coil would also be a second source of inductance (Le) so it would be challenging to get higher frequencies out of the design. It might be similar to adding a 1st-order low-pass filter onto an existing design, but there might be room to optimize it too.
You would also lose the electrical damping that is present from the magnetic field that the permanent magnet creates. The electrical damping of a speaker is substantially dominant compared to the mechanical damping. You can see this by comparing the Qes vs the Qms of a speaker:
Qms is related to the mechanical damping and this is where the "m" in Qms comes from.
Qes is related to the electrical damping and this is where the "e" in Qes comes from.
Qts is the combination of mechanical and electrical Q or the total Q, which is where the "t" in Qts comes from.
The total Q, which is related to the total damping of a speaker, is usually very similar to the electrical Qes of the speaker. A random speaker might have a Qms (mechanical) of 10 and a Qes (electrical) of 0.50, then the Qts (total) would be 0.476. So the electrical damping is twenty times more dominant compared to the mechanical damping.
So if you lose the electrical damping from not having a permanent magnetic field anymore, then the cone would tend to ring for 10-20x longer. This is very easy to test by tapping on a subwoofer cone when the amplifier is powered off and listening to how the cone rings, then turning on the amplifier and tapping on the cone again. You will hear a tremendous difference and you can even feel the presence of the electrical damping if you push on the cone a little bit.
This is also related to the "damping factor" wars of amplifiers from a while ago. If the amplifier had a poor damping factor then the electrical damping would be less effective and the cone's motion would be different than intended.
One example of a speaker that uses two electrical sources instead of a permanent magnet is an electrostatic, however they work a little differently than two voice coils like I think you are asking.
So all in all, the moving coil loudspeaker hasn't changed a whole lot in the decades it has been in use! It's one of the most amazing things that we still haven't made it obsolete with some new technology.
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3 Attachment(s)
Re: Ask an Acoustic Engineer (me)
I think this one is a little bit of "form follows function" and a lot of "form follows practicality".
Quote:
Originally Posted by
chithead
I'm curious about the typical speaker and subwoofer design. If sound is caused by vibrations, why are the devices producing those mostly round and slightly concave? Is this optimal for acoustics, or perhaps a compromise of strength, power handling, material cost, etc...
The round shape has many advantages in manufacturing like being able to use a lathe to turn metal parts to size, being able to apply adhesives with a stationary applicator while the speaker spins around on a bearing like a lazy-susan, and not having to control the orientation of many parts since a round magnet mates to a round steel plate and a round basket no matter how you spin them. It is convenient to take a sheet of material and roll it into a round shape like the former of a voice coil, or the copper windings of the voice coil itself.
The motion of the voice coil is also very critical to control, especially since the clearance in the magnetic gap is very tight (often less than a fraction of a millimeter). A round shape for things like the spider and surround apply a uniform force in every direction which keeps the voice coil centered in the magnetic gap really well.
A round shape also has no stress concentrations whereas a square shape would have a stress concentration at each of the four corners. This means the stress or friction or stretching of a material is exaggerated in a few specific spots which will lead those areas to always fail first. Think of a ketchup packet. You can press and pull and squeeze along the smooth edges of those things all you want and they will never open. But if you tear really gently on the edge with ruffles you can rip them apart easily. That is a stress concentration, which is a great thing for ketchup but a bad thing for speakers (like square ones). Actually anything with a "tear here" is a stress concentration on purpose.
Attachment 9770
Round shapes are also super easy to cut into cabinets compared to square holes. Sure you can get some more surface area with a square driver but the raw convenience of making round holes is huge.
The acoustic performance off-axis of a round shape is also very easy to predict. Having the same polar pattern up and down vs side to side is not always desired, but it is super predictable and relatively simple methods can be used to design with round speakers.
Quote:
Originally Posted by
dgage
Chithead, if you mean round vs say the kicker square cones then the issue is boundaries. A round sub has one boundary to keep on the same plane for smooth waves to develop from. If you look at Kicker’s square cones, they have to keep excursion down since it is harder to keep the square cone in the same plane, which would develop the cleanest waves. Plus a square potentially has 4 boundaries corners with each having the opportunity to be slightly out of sync if not perfectly in the same plane, which means a less clean audio wave since it isn’t one clean wave but potentially 4 waves slightly distorted from each other (corners). That’s the audio enthusiast answer and I’d be interested in how Justin can correct me, not that correction is the point. But that is a good question and I’d be interested to know how much further down the deep audio hole Justin could take it.
Interesting idea. I think you're talking about what happens when a cone stops moving as a single rigid body and starts to deform. Or maybe you're asking about how the corners would have a higher stress/pull on them kinda like the stress concentrations I mention above.
A square cone and a circular cone both have standing vibration modes (or breakup modes) at a high enough frequency. They are really neat to visualize and this was the entire subject of my first class at Penn State with Dr. Russell. He has an amazing website with tons of fantastic animated pictures of vibrations: everything from the textbook-style that I'm borrowing below to actual measurements he's made of hockey sticks and baseball bats and acoustic guitars.
If you have a moment, PLEASE go browse the few pages Dr. Russell has. It's mostly like a picture book of really fascinating animations that don't need any science to understand. He's one of the best teachers I've ever seen!
Attachment 9771
Image from Dr. Russell at Penn State, at his website:
https://www.acs.psu.edu/drussell/demos.html
https://www.acs.psu.edu/drussell/Dem...le/Circle.html
https://www.acs.psu.edu/drussell/Dem.../rect-mem.html
Quote:
Originally Posted by
Jdunk54nl
I would also think the concavity of a speaker is due to space limitations. If it was flat or convex, this would then mean that the speaker needs more space in front of it to be able to move. Ultimately it is about how much air can be pushed/moved and how much space it needs.
I would also think the concavity of a speaker has to be important in some way to the way it is moving the air, concave would mean that all of the air is being sent straight out, convex would send the air out in more drastic angles, and flat would match the in/out angles, but I could be wrong about that reasoning since I am basing that off of more how light works.
The traditional dish style cone shape is a form-follows-function thing where you need a speaker cone to be lightweight and rigid at the same time. If you use a geometric shape that is inherently strong then you can use a little less material and make the cone lighter. Think about a paper water cup that is usually found near water coolers.
Attachment 9772
The paper cup is really fragile when you first pick it up because you are holding it from the sides, and pinching it from the sides. When you fill it with water the cup has no problem holding the weight of the water because a cone is stronger in it's axial direction, or up-and-down in this case. When you're done with the cup and it is empty, it takes almost no effort to crush it from the sides and crumple it into the trash, and it weighs nearly nothing because it's essentially a thick sheet of paper with a little glue (much like a paper speaker cone!).
So the shape of the material makes it stronger in this case, but only for the intended use. If a paper cup is intended to hold water, then a paper speaker cone is intended to push and pull air in much the same way. It is a very efficient shape if you need a strong and lightweight shape that only pushes and pulls.
There are many "non traditional" shapes now that we have more exotic materials to play with. There are lots of tradeoffs in the shape of a cone and how it transmits energy to the air. A convex shape like a dome tweeter or a dome midrange has a different off-axis frequency response than a concave shape like a subwoofer cone, and some of that performance difference is due to the frequencies a tweeter plays vs what a subwoofer plays. A concave shape can also have either a straight wall which looks like a simple paper cup, or the wall can be curvlinear which is a fancy way of saying "a constant curve like the edge of a circle". Both have strength and performance tradeoffs too.
The dust cap on a speaker cone can also contribute to its strength in a big way. I'm working on a thin-mount subwoofer design that relies on the presence of the dust cap to make the structure stronger and the design would not work without the dust cap. It is really interesting to be able to model the stresses and how far a material will bend on the computer, and then making changes to see how you can improve the design without ever building a thing.
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Re: Ask an Acoustic Engineer (me)
Justin do you think we will ever have a single speaker cable of 20-20k? And please explain why or why not. Thanks
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Re: Ask an Acoustic Engineer (me)
Do you know anything about power supplies of amplifiers and the pros and cons of regulated vs non regulated power supplies in car audio amplifiers. An example of a regulated power supply would be JL Audio RIPS (regulated, intelligent power supply) or I think Rockford Fosgate constant power. Most car audio amplifiers have non regulated power supplies and their power changes with voltage and impedance as well as the inductance of a speaker has a great affect on power output, but dynamic power output can also be better with a non regulated power supply vs regulated. Will you hear the difference, I don’t know.
How about some commentary on speaker cone material and the certain attributes or pros and cons of each type. I have experience with some and have done a lot of research on this. I experienced the odd order distortion or breakup of aluminum cones as listening fatigue myself. Here is what I understand as sort of generalizations:
- Paper, the most popular which is light weight, strong, and doesn’t have nasty breakup nodes or odd order distortion like metal cones but isn’t as moisture resistant. Can be used in both three ways and two ways.
- Aluminum, has great detail, is very rigid and strong and has good pistonic action but can suffer from ringing and nasty cone breakup in higher octaves. Best used for three ways.
- Poly, exhibits a smooth response and breaks up gradually for a smooth extended response but can lack detail, is heavier, and temperature can have a great affect on performance. Good for car audio and can be used in both three ways and two ways. Great for two ways.
- Fiberglass and Carbon Fiber. I have a pair of Focal poly glass midwoofers that seem to sound very similar to how my paper cone mids do. Not much is known really. Can be expensive and possibly heavier than paper.
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2 Attachment(s)
Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
DaveG
Justin do you think we will ever have a single speaker cable of 20-20k? And please explain why or why not. Thanks
Ever? I think so. Soon? I'm not sure. I think it depends on how you define a "single speaker".
If you consider headphones to be a single speaker then these exist today already. They are able to get good bass response since they are, more or less, directly coupled to your eardrum. If you're listening to headphones with good bass response and then you lift them a little bit so the air seal is broken, the bass is tremendously reduced. This is because bass wavelengths are much longer than the size of the airspace between the headphone and your eardrum so any bass frequencies are directly passed along. This is a similar effect to "cabin gain" in the inside of a car where below a certain frequency the bass starts to get reinforced, seemingly for free. So headphones can do 20-20khz with a single driver but they do have a unique "enclosure" directly attached to your head that allows it.
If you mean a single voice coil and a single cone like a 6.5" driver that is supposed to fill a room with music, then it is much more challenging.
To get 20hz output you need a large cone area. You need a large cone area because it helps to move more air, yes, but you also need a large cone area because a larger cone will couple with the air better at lower frequencies meaning a larger cone will transfer energy into the air more efficiently. There is a crazy formula for acoustic radiation impedance that relies on a factor called "ka". A high ka value means the acoustic energy is more efficiently transferred from the piston to the air, and a low ka value means the energy transfer is less efficient.
The "k" part is essentially frequency. A higher frequency means a larger "k". (k is called the acoustic wave number)
The "a" part is half of the diameter of the speaker cone. A larger speaker cone means a larger "a".
So to get a really large k*a value you either need a high frequency or a large speaker cone or both.
Think about a tweeter. A tweeter has a small diameter so the "a" value is small. The only way to make a tweeter have an efficient energy transfer to the air is to have a large "k" value, which means higher frequencies. Thankfully that is what a tweeter does: high frequencies with a small cone area. So far so good.
Think about a subwoofer. A subwoofer plays low frequencies so the value of "k" must be small. The only way to make a subwoofer have an efficient energy transfer to the air is to have a large "a" value to balance the small "k" value. Because of this, subwoofers usually have a large cone area not only to move more air, but also to couple to the air better and improve the transfer of energy.
This is a really powerful concept and responsible for something I've been chasing for a long freaking time: mutual coupling. If you place one subwoofer outdoors on the left side of a stage, you will get some amount of bass. If you place a second subwoofer on the right side of the stage then you will get about twice as much bass, as expected. But if you place both subwoofers very close together at the center of the stage, you will get more than if you place them apart. This is commonly called mutual coupling. What happens is the effective cone area is increased so the "a" factor is increased so the efficiency that the energy transfers into the air increases. This also might describe why people love to have two huge 15" woofers in their cars even if they don't move much, compared to a bunch of smaller 8" woofers or something like that.
Back to your question: you'll need a large cone area to get any kind of bass energy out of it. A large cone could play treble but it will have tremendously narrow beaming at higher frequencies. A larger cone that can play bass will also have a larger and stronger motor that has a lot of inductance, and inductance is what rolls off treble response. So it's a game of tradeoffs that just doesn't work well. A different approach is needed.
The Synnergy Horn from Danley labs is one possible way to make it happen. Essentially you make a horn (flared tube) with a tweeter at the very back and then place gradually larger speakers along the tube as it gets bigger. You fire energy from every speaker into the same acoustic space (the horn) and it all exits together at the same time and it behaves like a "single speaker". I've heard a few of these and they are simple amazing. I believe one could be designed for 20-20khz range, however they are usually designed in the 50hz and up range for concerts and stadiums.
https://www.danleysoundlabs.com/technology/
Attachment 9805
Attachment 9806
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2 Attachment(s)
Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
JCsAudio
Do you know anything about power supplies of amplifiers and the pros and cons of regulated vs non regulated power supplies in car audio amplifiers. An example of a regulated power supply would be JL Audio RIPS (regulated, intelligent power supply) or I think Rockford Fosgate constant power. Most car audio amplifiers have non regulated power supplies and their power changes with voltage and impedance as well as the inductance of a speaker has a great affect on power output, but dynamic power output can also be better with a non regulated power supply vs regulated. Will you hear the difference, I don’t know.
I know a little bit, but not as much as I'd like. I have to pass on this one.
Quote:
Originally Posted by
JCsAudio
How about some commentary on speaker cone material and the certain attributes or pros and cons of each type. I have experience with some and have done a lot of research on this. I experienced the odd order distortion or breakup of aluminum cones as listening fatigue myself. Here is what I understand as sort of generalizations:
- Paper, the most popular which is light weight, strong, and doesn’t have nasty breakup nodes or odd order distortion like metal cones but isn’t as moisture resistant. Can be used in both three ways and two ways.
- Aluminum, has great detail, is very rigid and strong and has good pistonic action but can suffer from ringing and nasty cone breakup in higher octaves. Best used for three ways.
- Poly, exhibits a smooth response and breaks up gradually for a smooth extended response but can lack detail, is heavier, and temperature can have a great affect on performance. Good for car audio and can be used in both three ways and two ways. Great for two ways.
- Fiberglass and Carbon Fiber. I have a pair of Focal poly glass midwoofers that seem to sound very similar to how my paper cone mids do. Not much is known really. Can be expensive and possibly heavier than paper.
Speaker cone and compression driver diaphragms and tweeter dome materials all have the same goal:
-be lightweight
-be rigid
-be damped
-be durable
-be sexy
-be affordable
Paper is an amazingly good balance of these things with a little weakness on durability, and less sexy factor. Thankfully paper cones can be coated with water-resistant treatments that make them very durable. I am still absolutely amazed that one of the least expensive cone materials is also one of most well-rounded performers too (paper).
All the other materials are basically a mixture of those five qualities above. There are too many materials to analyze here but I think some of the the more exotic materials like beryllium and graphene have higher performance because they have the trifecta: lightweight, rigid, damped. The high price and exotic nature makes them sexy too.
Synthetic materials are great for mobile audio and marine and motorcycles where the environments are harsh. All kinds of plastics and polymers and composites are used because of their chemical and ultraviolet resistance. They don't always have the lightest weight, but they are generally affordable and very durable.
I'm not sure what you mean about odd order harmonic distortion though. When a cone is playing a high enough frequency it starts to distort in shape and has standing waves and bending modes but I don't think they add harmonic distortion to the mix. All the cone breakup and bending modes are the same frequency so you will get some off-axis response changes and some higher frequency roll-off, but I'm not sure there will be extra energy at other frequencies. See my post above with Dr. Russell's website and his animated gifs. They are sensational!
I think the thing you are hearing that is undesirable is the different damping coefficients. Better damping means the breakup modes are more controlled. Also certain material and geometry combinations can lead to dramatic or less dramatic breakup mode shapes. Scanspeak for example does a lot with geometry to control the way the breakup modes form and which standing waves are supressed. I'm going to borrow a picture from Erin's site below.
Attachment 9807
https://www.erinsaudiocorner.com/dri..._4.5_midrange/
Something else to consider: every cone or diaphragm will have a breakup mode at a high enough frequency. The clever speaker designers will either make those breakup behaviors gentle and polite such as the Scanspeak speaker above, or the designers will try to push those breakup behaviors well above the intended frequency range of the speaker.
The Audiofrog GB60 is a great example of this because it is intended to be crossed over at 2khz but the spike in frequency response from bending modes occurs at about twice that frequency.
The Peerless SLS 6" is one of my favorite midbass speakers. The frequency response falls apart above 500hz'ish due to bending modes. I'm guessing the manufacturer didn't worry about that because the specs on that driver scream "strong midbass for a 3-way system". It's all about the intended use in the end!
Attachment 9808
Attachment 9809
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edit: this is fun! thank you for the questions
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Re: Ask an Acoustic Engineer (me)
So we know the advantages of XBL2 motor designs (flat BL curve until the very end). But like everything, there is a con to each pro. What are the cons of an XBL2 motor topology
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1 Attachment(s)
Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
SkizeR
So we know the advantages of XBL2 motor designs (flat BL curve until the very end). But like everything, there is a con to each pro. What are the cons of an XBL2 motor topology
The XBL˛ motor geometry is patented by Dan Wiggins. He is perhaps the most energetic guy I've ever met haha. The design calls for a very short voice coil similar to an underhung design, and a unique top plate geometry. We had a fun discussion about it over on diyma in the thread below. Dan and I both talk about the XBL2 design, distortion measurements with the Klippel analyzer, and especially a lot of talk about the report I created that Adire Audio posted on their website that showed all the performance specs and Klippel measurements. It's a fun read.
https://www.diymobileaudio.com/threads/industry-standard-for-distortion-iec-62458.415015/
Downsides of the XBL2 design are:
The voice coil must be fairly short so you are limited by how much wire you can use in the voice coil, which can result in a lower BL strength at the rest position. Since BL strength affects many of the other Small Signal parameters like Qts, this can make the enclosure design different.
A lower BL at rest also influences the 1w/1m voltage sensitivity rating. If you are trying to design a speaker for use in pro audio like stadiums and concerts, one of the few spec numbers that are critical is the SPL sensitivity. If you have a lower than average sensitivity, that speaker will not sell well in the pro audio world.
You can see both of the above trends in a comparison I made of the Stereo Integrity TM65mkII vs the Audiofrog GB60. The TM65mkII has an XBL2 motor and it is very interesting to see the design tradeoffs. You can grab that report off my dropbox here:
https://www.dropbox.com/s/x151rliohp...02.11.pdf?dl=0
Simulation and design software for the XBL2 geometry is rare. The only program I've seen that does it is the one Dan Wiggins programmed however that business venture fell through so you have to get his attention and ask him for a copy. He was happy to let us have a copy at Eminence when I worked there and he even gave us a little training so that was nice.
Since it is a patented technology, it requires a fee to use. The fee might be small at the manufacturing level, but it gets multiplied by the time the product reaches the customer. This makes the product more expensive, but the cost is hopefully worth the cool factor and the performance boost.
The motor geometry is very non-standard so a typical speaker manufacturing house might not have experience with it. This could add to design and development costs when getting the first production batch up and going.
The motor also leaves a perfect gap inside the center of the voice coil to place a copper or aluminum shorting ring. The shorting ring is not a requirement of the XBL2 design but the opportunity is so perfect that it is tempting to add one which would increase the speaker cost. I think the ability to add a shorting ring in such a perfect spot is my favorite attribute of the design.
Here is a picture that I borrowed from the May 2009 Voice Coil magazine.
http://audience-av.com/wp-content/uploads/pdf/VC509_TB_Audience.pdf
Attachment 9811
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Justin Zazzi
I know a little bit, but not as much as I'd like. I have to pass on this one.
...
Let's slow up there young bull...
This is is a perfect topic.
Here is what I propose.
1) see if anyone near you has a regulated amp that you could test.
(I'll assume you have a non regulated amp)
2) Get a variac and use it to set the voltage and few capacitors behind a full bridge rectifier to feed DC to a battery.
Then put a current measuring device between then... probably something outputting to a datalogger.
Then get the test speaker producing the same SPL with both amp by adjusting their gains.
Then if one was to run something like 1/2 second of 40-Hz bursts out we can see what the input current is like in each case... both as an impulse and in a steady state sense.
One can also start turning down the variac to do the same test at 13, 12, 11 and 10V.
it is likely that that output SPL, in the non regulated amp, will vary as the voltage decreases.
Of course it is is a lot of hours of work, so it is easy for me to propose. But that is how I would likely start... and change it when I found some problem. At least knowing how to assess the problem would be a great thing to outline.
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Re: Ask an Acoustic Engineer (me)
Thank you Justin for sharing your expertise. This is a rare thing for someone like this to come on here and offer this kind of knowledge so everyone should take advantage and ask but please ask nicely. I’ll try to come up with a few more good relevant questions and post them later.
Holmz, I think what would be most helpful in terms of my question about regulated power supplies vs non regulated would be how these technologies actually matter when used as intended in the automotive environment. Having sagging voltages isn’t really typical unless you have inadequate power supply from the car. I’d like to know how do these technologies affect the sound when used as intended comparatively? Does the greater dynamic capability and faster response of a non regulated power supply make an audible difference comparatively? I think it does somewhat.
Reading a little bit about inductance from the XBL talk above from Justin and Dan Wiggins I realize that subwoofer inductance plays a huge roll in how a subwoofer sounds (please correct me if wrong) so an amplifier with a regulated power supply might be able to compensate somewhat for this, but is it audible? When I had my RF Power ti1500 I ran some basic tests and it made about a 1 db difference compared to my non regulated Infinity K-1000. The Infinity seemed to sound as if it had a little more dynamic headroom, which seemed to manifest itself as more boom in those quick low bass hits. I had to do a long A vs B session to determine this and it was just me doing it so the test isn’t really reliable or very scientific.
Thank you again Justin for sharing your knowledge.
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
JCsAudio
Thank you Justin for sharing your expertise. This is a rare thing for someone like this to come on here and offer this kind of knowledge so everyone should take advantage and ask but please ask nicely. I’ll try to come up with a few more good relevant questions and post them later.
Holmz, I think what would be most helpful in terms of my question about regulated power supplies vs non regulated would be how these technologies actually matter when used as intended in the automotive environment. Having sagging voltages isn’t really typical unless you have inadequate power supply from the car. I’d like to know how do these technologies affect the sound when used as intended comparatively? Does the greater dynamic capability and faster response of a non regulated power supply make an audible difference comparatively? I think it does somewhat.
Reading a little bit about inductance from the XBL talk above from Justin and Dan Wiggins I realize that subwoofer inductance plays a huge roll in how a subwoofer sounds (please correct me if wrong) so an amplifier with a regulated power supply might be able to compensate somewhat for this, but is it audible? When I had my RF Power ti1500 I ran some basic tests and it made about a 1 db difference compared to my non regulated Infinity K-1000. The Infinity seemed to sound as if it had a little more dynamic headroom, which seemed to manifest itself as more boom in those quick low bass hits. I had to do a long A vs B session to determine this and it was just me doing it so the test isn’t really reliable or very scientific.
Thank you again Justin for sharing your knowledge.
Well I took a shot at what I thought might be what one would want.
So in my mind, it would be SPL versus input voltage, and/or output waveform/voltage versus input DC supply voltage... and whether they diminish at the onset of the musical attack, or later in the steady state.
I think it would need need to be something we could measure and quantified(?). If we knew what it was, then we could test it in a calculus f(x) sense...
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Re: Ask an Acoustic Engineer (me)
True Holmz but that someone isn’t going to be me and math isn’t my forte.
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Re: Ask an Acoustic Engineer (me)
Ok Justin I have a question.
how do I compute the x position at DC for a given voltage or current for an IB configuration?
i am thinking of putting together one of the subwoofer boxes as a small sealed box with a 1-way inlet and outlet, and making a respirator in a McGyver fashion.
So I either to pump the sub in a stoke near DC (like 1/2-Hz)
or
I run a small signal at ~30-Hz for ~1-2 seconds and then have it rest for a few seconds.
Thoughts?
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Re: Ask an Acoustic Engineer (me)
This sounds like fun, but I don't understand the question.
Are you asking what the displacement will be for a given voltage in a certain speaker so you can create a resperator and save the world?
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Justin Zazzi
This sounds like fun, but I don't understand the question.
Are you asking what the displacement will be for a given voltage in a certain speaker so you can create a resperator and save the world?
I'll start out small, so just a person or two... :cool:
i have enough subwoofers for 4 of them, which equals what I hear if the sad number that the hospital has.
I think I'll just slap together a seal box with an inlet and outlet, and use a 30-Hz tone and see what it pumps.
It would be nice to if I could send in something around 10-Hz without smoking coils, and the math behind working of the displacement versus current/watts.
But I just do not know how the parameters translate into displacement... hence the question.
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1 Attachment(s)
Re: 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?
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Re: Ask an Acoustic Engineer (me)
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.
Sent from my iPhone using Tapatalk
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Re: Ask an Acoustic Engineer (me)
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.
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Justin Zazzi
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!
Sent from my iPhone using Tapatalk
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Re: Ask an Acoustic Engineer (me)
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.
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Sandman
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!
Quote:
Originally Posted by
Sandman
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.
Quote:
Originally Posted by
Sandman
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.
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Re: Ask an Acoustic Engineer (me)
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.
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Re: Ask an Acoustic Engineer (me)
I wonder if you could help me with this question...
https://www.caraudiojunkies.com/show...asound-Whisper
...seeing as I haven't made any practical progress yet, due to work
Thanks in advance
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
VX220
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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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Jscoyne2
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.
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
VX220
I'll reply to your thread directly.
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Justin Zazzi
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/dayton...ystem--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.
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Jscoyne2
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/dayton...ystem--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
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Re: Ask an Acoustic Engineer (me)
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!
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
jtrosky
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.
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Jdunk54nl
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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1 Attachment(s)
Re: Ask an Acoustic Engineer (me)
You can look at pg 33 here
https://testgear.audiofrog.com/wp-co...y-it-Works.pdf
This is the picture that goes along with it.
Attachment 9976
If we use that and back out 4 x diameter of a 9", that would give us a frequency of ~375 hz to be in the green area. 2x Diameter = ~750hz. This is in air at 20 degrees celsius.
This depends on how the factory oriented the speakers though. Is it behaving like a 6" speaker or a 9" speaker or somewhere in between?
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
jtrosky
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)
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Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Justin Zazzi
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/Tra...er_Effect.html
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2 Attachment(s)
Re: Ask an Acoustic Engineer (me)
Quote:
Originally Posted by
Jscoyne2
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.
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.
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:
Quote:
Originally Posted by
Jscoyne2
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/dayton...ystem--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.