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Why I Rejected Ported/Passive

11K views 74 replies 15 participants last post by  davepete 
#1 ·
Sorry for hijacking Ilkka's thread. terry j was asking why I rejected a ported design for a sealed design. The short answer is my tuning frequency is too low. I've copied the hijack posts here so the discussion can continue as necessary. I'm in italics. terry j in between.

I rejected going LLT in favor of sealed, with IB not an option. For my design goals, namely flat in-room response to single-digits, the ported and passive radiator models didn't turn out very well. Particularly because of the port volume needed to be so large coupled with the rapidly increasing group delay. It would have been extremely difficult to build an enclosure with a port that was larger than the enclosure, so a passive radiator would be more feasible. But the group delay gets even worse with a PR. And it would be harder to find a PR that would provide the necessary mass to fit by itself on a sonotube endcap; I originally resigned myself to a box enclosure, but the calculated weight would have been like 200 pounds per unit, or something like that, for a PR design.

The trade off is I need to pump in much more power and the overall cost is increased, but I am going with sealed sonotubes now.


Joshua, have you written up your sub anyplace, if so I'd love to be able to have a look. I will readily admit that I'm not technically 'savvy' enough to really follow your reasoning above, but can't you use eq on the LLT to get the same in room response as a sealed?? Presumably you came to your conclusions via a modelling program, which one did you use? Could your decision have been different if you'd modelled with different drivers?

I'm still in the processing of building the subs (someone tell me why it is so difficult to get an endcap to fit into a sonotube?!) and plan to write things up and post photos when it's done. Probably next weekend.

I use WinISD Pro (alpha) to do my modeling. I wish there was a decent Mac OS X program to do it, but doesn't seem to exist. My overall conclusion wouldn't depend a lot on the drivers, because all you'd be doing is moving the max SPL level up and down and moving the driver Fs left or right.

So I just modeled a 15" TC-2000 with two SA-PR15-1400 passive radiators. Maximum cone excursion is reached at 7Hz with 120W of power. So you see, applying a boost here would blow everything up. Again, ported won't work because the port volume is just too great for such a low tuning. Ported and PR designs fall off at 18dB/octave and 24dB/octave (IIRC) at the tuning frequency. The tuning frequency of the design I just modeled is 16.55Hz. I can use EQ cuts to flatten the response up to the tuning frequency (SPL at 30Hz is about 30dB greater than SPL at 7Hz), but I can't do anything to boost the low end.

Sealed designs roll off much more gradually, so I need more power but I can use EQ boosts to make it flat without sacrificing the higher frequencies. My sealed design, using two 15" drivers instead of a 15" driver and two 15" PRs, and boosting the low end, reaches maximum excursion at 7Hz with about 2200W of power, but is about 20dB stronger at 7Hz, and the high end is much closer to that point; e.g. SPL at 30Hz is about 10dB greater than SPL at 7Hz.

Also, I'm not talking about max SPL. Just the flatness of the SPL across the spectrum. Max SPL >20Hz is still greater with less watts with a ported or passive design.

As a side note, group delay is not so bad in the PR I just designed. Only a little worse than the sealed and probably negligible.


Just so I understand, you mention the port volume required at the frequencies you're talking about as 'just too great'. Does that mean to compensate by increasing the size of your tube it gets too big, or are you practically saying the port is HUGE and simply not practical. Again, from vague recall of the LLT philosophy, most are knida tuned around 15 hz or so, is yours different in being tuned much lower which then leads to the problems you've encountered??

sorry Josh, just re-read your post, and see that you are tuning to 16.55 hz, so you're not tuning lower. Will have to re-read the LLT again to work out for myself why most can use an LLT and you can't, went and got myself all confused again!!
 
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#32 ·
Ilkka said:
I'll list some enclosure volumes and internal pressures:
600 liters: ~1.5 mbar
300 liters: ~6.0 mbar
150 liters: ~25 mbar
75 liters: ~100 mbar

1 mbar can be defined as ~10 kg per 1 square meter or ~2 lbs per 1 square foot. So for example 6 mbar would equal with 60 kg per 1 m^2. Well how about our 100 kg guy sitting on a tube (area around 30 cm by 30 cm)? That's around 1100 kg per 1 m^2, meaning around 18 times more pressure than what the woofer can produce. Although in a 75 liter box they would pretty much equal.

Based on this, I'd say the internal pressure isn't a big problem, especially with large enclosures. The smaller the box gets, the more thicker (and braced) the walls need to be.
Indeed.

Bosso - here's an example you might like. Take a piece of sonotube - let's say any diameter larger than 6". Now cut it into two pieces, one that's only 4" long, and one that's 6' long. Now try with your bare hands to compress the 4" length. I don't know if you have sonotube laying around, but I do, so I can tell you it's very easy. When I try the same with the 6' length, it's nearly impossible, it won't give at all. This is because the force I am creating is being spread over a larger surface area.
 
#33 ·
Yes, now...try the same experiment and increase the diameter of only the 6' long tube, with end caps installed on both examples. Which tube gives more? The thought that a smaller box or tube is weaker is incorrect.

The first incorrect assumption here is that the tubes are perfectly round to begin with, which they aren't.

The second incorrect assumption is that I'm somehow talking about huge flexing.

The third misconception seems to be that either method of enclosure design is flex-free. We're talking about the least amount of flex, not which one has zero flex. And, we're talking about very small distances of flex.

Ilk...when you say that the tube expert says that if a 200 pound guy sits on a tube and it only would flex 'a bit', what specifically would that mean in mm? If he has that info, we could extrapolate the amount of flex with 2 opposing drivers, which would stay on topic with the OP's design. I would imagine that what the guy means as 'a bit', in terms of this discussion would actually be quite a bit.

Then again, if the same 200 pound guy would sit on a braced box of the same internal volume, would it not flex even less than the tube?

From Dan Wiggins from back when his company was called Avatar:

: Now, when a driver begins backward motion, into the box, it will create a
: compression zone (pressure above ambient). This zone will travel as a wave.
: When the driver moves forward, it will create a rarefraction zone (pressure
: below ambient). Likewise, this zone will travel as a wave. This is a basic
: fact of wave mechanics and physics, and will occur for all cases.

My argument is that when the pressure zone wave travels through the tube, it, at the very least, would correct the tube to perfectly round where it isn't so to begin with. When the rarefraction zone wave is traveling through the tube, it would, at the very least, return the tube to it's imperfect shape.

I would also say that my argument goes beyond these least case scenarios to a case where the rarefraction zone distorts the shape of the tube beyond it's original shape.

This thought is what Dan seems to be saying in the same conversation:

"While this is true theoretically [that a tube cannot flex], in practice, one will often have flexing
in a tubular enclosure, simply because the enclosure is not a perfect tube."

He also brings up a point that goes with the very large tube enclosure that most have not addressed:

" Well, you forgot about internal reflections. Realize that when the
wavelength is equal to 4 times the length of the tube, one will have
cancellations from reflections inside the tube."

Using a 6' tube, that would be around 45Hz, no?

Bosso
 
#34 ·
bosso said:
Yes, now...try the same experiment and increase the diameter of only the 6' long tube, with end caps installed on both examples. Which tube gives more? The thought that a smaller box or tube is weaker is incorrect.
It's not that a small tube or box is weaker, the strength - let's say resistance to deformation - per surface area is actually higher than that of a larger tube or box with walls or endcaps of the same thickness. However, because the larger enclosure has much more surface area, the force applied per surface area isn't as much. If you really want to get detailed, Ilkka's numbers are only one half of the equation. We'd also need to know the resistance of each enclosure type to deformation. Then you can compare the force per surface area of each enclosure to its resistance to deformation. I'm not particularly interested in calculating that :R

But to get back to your earlier comment about pipe, and the wall thickness increasing when diameter increases, is that not because the pressure in the smaller tube and larger tube is going to be the same? Moving x amount of water at some rate in tube of y diameter = some pressure (pressure = F/A). In the larger tube, with a larger cross sectional area, in the same amount of time, you're moving more water, so the force is much higher. If we assume the cross sectional area is multiplied by a factor of 3, and the force is multiplied by a factor of 3, then yes, the wall thickness of the larger tube needs to be increased compared to the smaller tube, because the relationship of tube wall thickness to tube diameter would have to be the same, as the pressure is the same. HOWEVER, in a large and a small subwoofer enclosure using the same driver, the force is a constant, not a variable, and thus, if enclosure surface area of a tube increases, the pressure is reduced, and wall thickness does not have to increase.
 
#35 ·
As for your comments on sealed vs ported and giving up headroom, we've had these discussions before, and you seem to keep forgetting the outcome. Keep in mind that with a LLT you only need to be amp limited to some point above tuning, usually ~20hz. With sealed, you need to amp or enclosure limit based on how low you want to reach. If we assume you want something close to a flat response, then headroom in the higher bass frequencies is going to take a big hit. Assuming you want to delve deeper than ~20hz, how exactly is the LLT giving up headroom to it? An LLT is only giving up headroom to a higher tuned ported sub with a highpass, not a sealed with as much flat extension. The sealed will NEVER be able to keep up assuming the same driver. If you use a highpass on a sealed as opposed to trying to get relatively flat, usable output to say 5hz, you can gain more output capability in higher frequencies.
 
#46 ·
Headroom for dynamics from 30Hz on up does not take a hit in the sealed sub, but it does in the LLT. Since you're applying more power to the sealed sub, you have it in reserve, unlike the LLT, which sacrifices 30-50Hz for a lower tune and is amp limited, which translates at 30-50Hz as less headroom.

When you calculate the EQ/excursion headroom/amplification for a L/T sealed sub, you end up with a flat anechoic FR to the desired new F3, amp limited. This does not change the headroom from 30Hz and up.

Music isn't static, it's dynamic. Headroom in the 30Hz and up region for transients is probably the most important factor in accurate reproduction.

You seem to keep forgetting that room gain typically kicks in at 30Hz and the assumption of a flat response is not the simulated anechoic one.

Sticking to the point, just look at the anechoic response of the LLT from 30-50Hz, where typically there is no room gain. If you choose to EQ that dip, you lose even more headroom.

The main point is that in a sealed, EQ'd sub you are not 'chasing single digits'. In fact, very little L/T boost is required to couple the sealed sub to the typical room to result in a flat response to single digits.

You're also far too hung up on the 'driver for driver' argument. I design a subwoofer to meet the design goals, not to compete with anything else. If you choose to use a single driver to compete with a multi-driver sealed sub that has more available power, good for you. The problem is that you say that the only price is size, when it's also less headroom in the 30-50Hz range and no output below tune. Having the idea that you don't need lower response is fine, it just isn't a law that others need to observe or to suffer anyone's wrath over.
 
#38 · (Edited)
A bit more.

bosso said:
The first incorrect assumption here is that the tubes are perfectly round to begin with, which they aren't.
You stick a perfectly circular endcap in the top and a perfectly circular endcap in the bottom, and assuming a tight fit, you're basically forcing the tube to take the shape of the endcaps. Could there be some imperfection along the way? I guess, but I'd have to think it's negligible.

bosso said:
The second incorrect assumption is that I'm somehow talking about huge flexing.
bosso said:
The third misconception seems to be that either method of enclosure design is flex-free. We're talking about the least amount of flex, not which one has zero flex. And, we're talking about very small distances of flex.
I didn't think we're even talking about the discrete amount of flexing, we're comparing tendency to flex from one enclosure to the next.

bosso said:
My argument is that when the pressure zone wave travels through the tube, it, at the very least, would correct the tube to perfectly round where it isn't so to begin with. When the rarefraction zone wave is traveling through the tube, it would, at the very least, return the tube to it's imperfect shape.
Why do you assume the tube would change shape as opposed to the air just getting compressed? In a small sealed, when a driver is displacing 2 liters of air, the enclosure walls don't flex on a 1:1 ratio with the driver, correct? In fact, they move very little. And that's working with a pretty small effective volume, so the resistance to allowing the air to compress is high. So what do you think is gonna happen with a 300 liter + enclosure?
 
#45 ·
No, my goal is very vague right now. I'm going to try to EQ so that whatever I get, the SPL output can still reach at least 105dB across the majority of the frequencies. I don't have a specific SPL at a specific low frequency as my target. I'm going to see what I've got and make decisions based on that.

What I do know is:

1) The project was a success; I have usable bass below 10Hz to at least 7Hz. The amp was not clipping and it's gain was set at about 1/2. I didn't see any clip lights on the DCX2496 either, but may have missed it.

2) I have the capability of applying up to +45dB of LP filter below 20Hz at either 6dB/octave or 12dB/octave because I can daisy-chain the DCX2496 inputs and outputs. I just need to find out what the right amount is.

3) Room EQ Wizard only measures down to 10Hz. So I'm flattening things between 10-20Hz using the LP filters, and assuming the roll off just happens to be okay <10Hz. Since I have no EQ filters <20Hz, it's not going to be flat all the way down unless it just happens to be. I'll just pick something that works overall and live with it. At least my room is large.

4) I'm not really sure how much to trust Room EQ Wizard. When I used sound card calibration, the measured SPL went way up as it approached 10Hz. I need to do more experimenting.
 
#43 ·
My endcaps are not perfectly round. I'm not that good of a woodworker. I've got 1.5" of endcap inside the sonotube at each end. My tubes at ~4' tall and 20" in diameter. I don't know the thickness of the walls in millimeters. There is no internal bracing.

Anyway, I can drive them hard at some frequency and tell people what I feel from touching the sides of the tubes. Just give me a frequency. (I'll wear earplugs.)

And while on the subject, assuming the walls flex in and out to some degree...what will the end result be in terms of something I can measure? Increased harmonic distortion?
 
#49 ·
bosso said:
Headroom for dynamics from 30Hz on up does not take a hit in the sealed sub, but it does in the LLT. Since you're applying more power to the sealed sub, you have it in reserve, unlike the LLT, which sacrifices 30-50Hz for a lower tune and is amp limited, which translates at 30-50Hz as less headroom.

When you calculate the EQ/excursion headroom/amplification for a L/T sealed sub, you end up with a flat anechoic FR to the desired new F3, amp limited. This does not change the headroom from 30Hz and up.
But you can't apply "more power" to a LT'd sealed sub that tries to stay as flat as a LLT. In order to keep it within excursion limits, by amp/enclosure limiting, the power handling goes way down as compared to what it could be. There is no "reserve" power for a LT'd sealed sub that is relatively flat in FR. You have to decrease the amount of power you feed it to allow reasonably flat extension down low, otherwise you'll exceed excursion much too easily. With the LLT, you only have to limit to a point above tuning, so it will always have more headroom than a sealed with an EQ'd relatively flat response that tries to go as low or lower.

bosso said:
Music isn't static, it's dynamic. Headroom in the 30Hz and up region for transients is probably the most important factor in accurate reproduction.
So in other words, compression is good?

bosso said:
You seem to keep forgetting that room gain typically kicks in at 30Hz and the assumption of a flat response is not the simulated anechoic one.
I've seen dozens of in room FR measurements, and on the whole, room gain isn't nearly as potent as you and Mark like to make it out to be. You can continue to believe it is and enjoy yourself, but I'm a realist.

bosso said:
You're also far too hung up on the 'driver for driver' argument. I design a subwoofer to meet the design goals, not to compete with anything else. If you choose to use a single driver to compete with a multi-driver sealed sub that has more available power, good for you
Lol, you just don't get it. If x design using a single driver can outperform design y, then every time you go with more of x as opposed to y, it exponentially outdoes the equivalent of y. So yeah, 5y will easily beat 1x, just as 5x will easily beat 5y.

bosso said:
Sticking to the point, just look at the anechoic response of the LLT from 30-50Hz, where typically there is no room gain. If you choose to EQ that dip, you lose even more headroom.
What dip?
 
#50 ·
josuah says:
4) I'm not really sure how much to trust Room EQ Wizard. When I used sound card calibration, the measured SPL went way up as it approached 10Hz. I need to do more experimenting.
You may trust it absolutely as long as you have made a proper soundcard.cal file. That's the purpose of the file - to compensate for soundcard response, so as to make it perfect.

To test the file simply short a cable from the line-in to the line-out and do a sweep (with the soundcard file engaged and the c-weight and mic calibration file removed).

This test should result in a perfect flat line.

Here's a couple of graphs of my computer with a loopback on the soundcard to test that REW doesn't modify the response of any measurement.

The first is with the regular vertical axis we normally use and the second is with a large expanded vertical axis to exaggerate any problems.

As you can see, REW measures incredibly flat. I happen to be able to go down to 2Hz with a beta testing version, but the regular version will give a flat response to 10Hz. If it ain't flat, REW isn't at fault.

Text Line Slope Parallel Design

expanded vertical axis.
Text Line Slope Parallel Design


brucek
 
#55 ·
You may trust it absolutely as long as you have made a proper soundcard.cal file. That's the purpose of the file - to compensate for soundcard response, so as to make it perfect.

To test the file simply short a cable from the line-in to the line-out and do a sweep (with the soundcard file engaged and the c-weight and mic calibration file removed).

This test should result in a perfect flat line.
Okay, so if my soundcard calibration does not result in a perfectly flat line? I ran a 1/8" stereo cable from the headphones out to the mic in, and it starts dropping off somewhere around 30Hz, but a lot. I think near 10Hz or so it's like -30dB.
 
#52 ·
I've seen dozens of in room FR measurements, and on the whole, room gain isn't nearly as potent as you and Mark like to make it out to be. You can continue to believe it is and enjoy yourself, but I'm a realist.
I guess that between the two of us, we've set up more than a few systems knowing the anechoic response going in. A realist would tend to notice the results.

Lol, you just don't get it. If x design using a single driver can outperform design y, then every time you go with more of x as opposed to y, it exponentially outdoes the equivalent of y. So yeah, 5y will easily beat 1x, just as 5x will easily beat 5y.
Hah...except for the fact that 4 of the Ys would have to be out in the garage. 5 Ys would need their own zip code. As I said...the design goals matter and industrial design is a big part of that. SVS used to say 'Form follows function' when all they had to sell were the rug covered water heaters...they've since (thankfully) changed their tune. If 5 yards of cloth sock gets it for ya, then I'm happy for you.

BTW, it doesn't take 5 Xs to outdo 1 Y. You don't have to be so dramatic. In fact, at normal listening levels, there's nothing exponential about a 1 to 1 comparo.

Bosso
 
#56 ·
Okay, so if my soundcard calibration does not result in a perfectly flat line? I ran a 1/8" stereo cable from the headphones out to the mic in, and it starts dropping off somewhere around 30Hz, but a lot. I think near 10Hz or so it's like -30dB.
Why are you using the headphone out? Use the main line out, it should have the best naturally flat FR.
 
#58 ·
Okay, that explains things. :p Maybe this should be mentioned in the FAQ/instructions. I also didn't notice taking sound card measurements as something to do in the instructions.
 
#59 ·
If you follow through the HELP files in setting up REW, it mentions line in and out. There's also been quite a bit of discussion about mic/headphone use in laptops along with using USB sound cards for laptops. The REW HELP files definitely has the step of calibrating your sound card. It references using left channel, etc... which there is no left or right on a mic or headphone input or output on a laptop.

At any rate, check out the BFD | REW Forum for the Creative SoundBlaster MP3+ USB sound card. It's what I use on my lappy and it works great. Not that expensive either.
 
#60 ·
Okay, so if my soundcard calibration does not result in a perfectly flat line? I ran a 1/8" stereo cable from the headphones out to the mic in
Yeah, I suspected something was amiss. You should definitely get an external usb soundcard as others suggest. The connections you're using aren't really appropriate.

Actually, you can get away with a headphone output in a pinch, but the mic input is a no-no. They're quite noisy, offer poor response, provide too much gain, and usually supply a phantom voltage for cheapy microphones.

You'll need an external usb soundcard with a line in and line out capability - they're quite inexpensive.

If you're serious about measuring responses in the <10Hz area, you may consider a good microphone also.

brucek
 
#65 ·
Interestingly enough, I've been seeing the same problem with REQW that Josuah was describing, and I'm using line in/line out of the Creative USB soundcard on my laptop. I went through the entire setup process exactly as described in the help file, but when I run the automatic measurement sweep for my sealed 2x12 subwoofer, the graphed response doesn't look right. It starts out at 10hz recording a very high level, swinging downward a bit and then back up. This sub's response does not extend down to 10hz, so there's no way that can be right. The strange thing is that when I measure the response curve using the manual measurement method (using an input cable from my digital Rat Shack meter), I get a proper response curve, starting out low at 10hz and building exactly as my ears say it should. Any ideas? Is there a problem with the auto measurement function, or am I doing something wrong?

DP
 
#61 ·
Guess I missed all of that discussion and documentation. :( I can lug my Mac downstairs. It has both line in and line out somewhere on the back. And maybe I'll get a better mic instead of the RS meter once the next version of REW comes out. Thanks!
 
#62 ·
I'd suggest spending a little extra money and getting the M-Audio MobilePre. It's a usb soundcard with line in/out and can also provide phantom power for the Behringer mic. They usually run about $150 (I saw mention of a sale on them through Amazon for even less) and are well worth the money.
 
#64 ·
Thanks. I'll look into it. The line in/out combination on my Mac is about -10dB at 10Hz. But relatively flat until 20Hz.
 
#66 ·
I get nearly perfect looking data using an external Creative Live soundcard, depending on which USB port I use.

My laptop, a Compaq R3000Z (iirc), has two sets of USB ports - one on the right side and two on the left. I think the right side is USB 2.0 and the right is like USB 1.2 or something dumb like that. But, I don't beleive that the left side has enough bandwidth or something to take the data fast enough. The right side works perfectly. I've seen a few other people have this problem. So it isn't enought to have an external soundcard, you have to have a good USB port also - I beleive.
 
#68 ·
Ryan, thanks for the suggestion. I hadn't considered the speed of the USB port. Maybe that's the problem. I have a fairly new Dell Inspiron E1505, and I see in the sytem config there is at least one USB2 host controller, so I'll have to do a little more digging to see which port that corresponds to. Or I suppose I could just try all four ports. I'll run some tests later this afternoon and report the results back.

Dave
 
#67 ·
I went through the entire setup process exactly as described in the help file, but when I run the automatic measurement sweep
Did you create and save a soundcard.cal file with the line-in selected as the input device?

Then when you connect a stereo cable between line-in and line-out and do a measurement (with the microphone cal file cleared) do you get a flat line?

P.S. You should actually start this thread in the REW forum so this thread isn't hijacked....

brucek
 
#69 ·
Bruce, yes I did create a soundcard.cal file and it was loaded when I ran the auto measurements. I did a loopback measurement during the setup process and I did indeed get a flat line that looked exactly like what the docs said it should.

I'll restart this thread over in the REW forum where it should be. I only asked the question here because I noticed Josuah describing almost the exact same thing I was seeing. :scratch:

Dave
 
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