3ll3d00d

Also is there any thought being given to other considerations such as the total area of the high speed particles? For example a 4" port firing into the back of a couch versus a 21" driver surface area. One is a higher intensity but much more localized effect versus a more diffuse effect over a larger area.

I have been thinking about this in the last day or two, both with respect to port positioning (relative to the body) and how many ports to use.

The downside of a ported box for nearfield is the box is bigger so you can fit fewer boxes in a given seating area. The upside is they have much greater PVL so you need fewer of them to achieve a given level of TR or you can run them at a lower level (to avoid negative audible impact from the NF). Most NF subs appear to be simple sealed designs (often using an HT18 or a cheap car sub like the infinity 1260) and the 12 inch designs are commonly used in multiples, 2 per seat seems pretty common so you're looking at a bank of 8 subs spread out behind the seats. This sort of output seems to be enough to deliver plenty of TR across those seats. Therefore I've been thinking that running multiple ports (3-4 for the rear chamber, 2 for the front chamber) is a good idea so as to spread the output out across the area. I could also then more at chest height as people seem to report that this is more effective for a (lower) mid bass tune

I thought this might also let me play with the tune of the box a little bit (though whether that is practically useful remains to be seen).

The other thing I've considered is a 4th order bandpass box which comes out quite a bit smaller & still provides significant PVL through most of the range. It seems, on paper, one could use this design quite effectively if using multiple subs NF to get both the low end and the upper end. 

Is that supposed to display an image? It doesn't work for me.

for completeness, a high Q ULF LS has some ripple in jriver. It equates to a minimum corner frequency of ~8Hz for completely predictable behaviour. I'm not sure if this is to be expected or not, probably irrelevant anyway but I thought I'd mention it (can't attach a pic due to space issues, not sure if http://yabb.jriver.com/interact/index.php?action=dlattach;topic=106708.0;attach=22139;imageis visible either)

Ok so practically speaking we are back to rules of thumb (aka experience) as it doesn't seem amenable to modelling without being a fluid dynamics expert.

as subject really, I might be blind but I haven't found good info on this so far. I am aware of rules of thumb like "keep it under x% of speed of sound" but this seems distinct from whether a particular port will compress or not.

if the answer is "yes, if you use akabak" then any example would be appreciated as I've never used it but have been planning to try it out

context is I'm planning on building a dual reflex bandpass sub to test out this PVL hypothesis, design details in http://www.avsforum.com/forum/155-diy-speakers-subs/2539913-ported-nf-uxl-18-build.html#post46292769in case anyone is interested

I actually generate the BEQ curves with S, and calculate Q to post here.  I can post both on future BEQ.  I sometimes use steeper S than 1, though.  JRiver should allow values greater than 1, as the filters are not 'unstable' as they claim.

 

JSS

the next build of MC22 allows S<=5 (which equates to Q~=1.73), there might be some instability in the output if the corner frequency is set to 5Hz but it seems ok above this (not sure if measurement issue on my part)

I get the reason why you want to push the XO lower, it's a good reason for sure. I'm just saying it might be a risky strategy and that dynamically compressing the content to deal with it is not, IMV, a good trade off. I'm not saying what you're doing won't work btw or that it's an inherently bad idea, testing (using your ears) is the best and quickest way to verify.

FWIW I have found there are a few tracks from Adele 21 to be perfect for stress testing this aspect of a crossover, she can really belt it out and her voice has a broad range (which you can see if you watch a live spectrum analyser alongside listening).

Yes I did, and I realized the speaker's max volume will be limited with this low of a crossover. However, the speaker won't be playing all that loud for the tweeter to struggle. Eventually I want to find a way to make a limiter that changes the tweeter's crossover frequency from 1500Hz to 3000Hz above a level where the tweeter will start distorting. One forum member from another forum says the DSP tools he's working on could help me do this. But one step at a time. 

I don't understand why you'd want to design something that requires you to deliberately introduce distortion in a critical part of the frequency range, it seems completely contrary to some of your goals too. At what distance will you listening btw?

FWIW I made a speaker recently for which the 1st iteration of the crossover pushed the tweeter too low, I found this introduced a harshness/roughness that was really obvious with certain content even at lower levels. It was quite unpleasant, almost unlistenable in fact, but only on that specific content. I didn't think I'd even pushed it that far beyond and the measurements didn't look bad but it was still rough (this was roughly an LR6 at 1500Hz vs an LR4 at 1900Hz). Different speaker obviously so ymmv. 

A 1500 crossover sounds optimistic to me. Have you looked at http://medleysmusings.com/scan-speak-illuminator-d3004602000-tweeter-testing/ ?

1st half of http://www.aes-media.org/sections/pnw/ppt/jj/room_correction.pptis a good summary, comes from http://www.aes-media.org/sections/pnw/pnwrecaps/2008/jj_jan08/

Actually there was a post on this subject in the last day or so comparing the various target curves (b&k, ITU-R BS.1116-3, EBU 3276, the harmon/olive paper from 2010 & the toole paper from 2015) and they're all v similar. The specs (EBU/ITU) go into more detail on the room & speaker parameters, I don't recall whether the B&K & the Harmon papers gave specific guidance on that point though.

In 1994 (updated in 2015), the ITU produced a "Recommendation ITU-R BS.1116-3

 

Room2 spl and velocity 0-90-h directions:

Rom2 spl v 0 90 h.png

 

 

The velocity spl plots reveal that the cause for the peak around 800hz is likely to be ceiling or floor.

The dips may be cause by reflections from the back, but down at 200hz there may be a sideways reflection, and up closer to 600hz there may be a vertical reflection.

 

What is the h direction? what in that plot tells you the peak is the ceiling or the floor?

The direct sound arrival for that bass may cancel at the listening position, but it will be returned to the listening position in later reflections.

What do you mean by that exactly? The difference between the transient leading edge of a signal vs steady state or something else?

looks like the same yes

How is this done? is there a guide that was written?

it is described in the link in my earlier post http://www.avsforum.com/forum/113-subwoofers-bass-transducers/1333462-new-master-list-bass-movies-frequency-charts-post23468771.html#post23468771

 

the second one was done by extracting all channels then adding them to Audacity and exporting as mono wav in the same 24bit as the original movie files.

 

I can't seem to add the dts file, what program do you use to convert it into a wav file? Sorry I should of asked this from the beginning.

 jriver as per http://www.avsforum.com/forum/113-subwoofers-bass-transducers/1333462-new-master-list-bass-movies-frequency-charts-post23468771.html#post23468771

I made my boxes a bit like that. I wasn't sure whether it would have a significant effect on diffraction or not as I couldn't find any tool to model that shape. Once I get the boxes finished then I could compare against my test box (which is just a box) to see how it looks.

  For my center channel, I'm planning on installing plywood at an angle between each edge of the speaker and the wall to maximize beneficial boundary gain.  The speakers are only 225 mm or so deep, for this express purpose.

 

do you mean so that the shape presented is like this or something else? i.e. the square is your cabinet and the slashes are the angled plywood

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Have you guys every graphed Rush?

DATS sorta worked with W7, but it was squirrley. Now that I'm on W10, I can't get the DATS to calibrate properly or even measure the same driver (or resistor) repeatably. I'm not impressed with it at all. 

My settings are correct. I've tried updating to the latest software (for even more $$$), still no joy. At this point, I'm done wasting time and money on it. 

 

FWIW (probably not much as you have multiple other ways to measure!) there was a thread on techtalk about that, various users had similar problems but settings were worked out that fixed the problems. 

the soundtrack is pretty lame and it is rather long, not that that is news to anyone who has watched any LoTR film (especially those directors cut versions), but I don't think it is *that* bad. There are certainly many far far far worse movies out there anyway.

transient response may be made to disappear from the FR if too much smoothing is used. I believe 1/24th octave is the absolute minimum for bass. For < 120 Hz, I usually prefer to review my measurements responses un-smoothed and a very long time window.

i don't follow you here, what is the connection between steady state fr with effectively no gate (or smoothing) and transient response?

Vibsensor is a useful tool for seeing what is going on on a particular surface under ~40Hz. I don't see this as having anything to do with transient response nor music playback. I find that that effect, since adding a NF sub, accentuates feelings of dread/tension during certain scenes and that running it too hot makes it a bit too much of a rollercoaster. If I sense the NF during music playback then that is just wrong for my preference.

My preference sounds closer to bosso with respect to FR BTW, I run it with a small (~3dB) lift from 120Hz down to 40Hz (roughly) and flat from there. The NF response is broadly flat from 15-45Hz and rolls off steeply either side using 2nd order NT filters, this means the NF is really completely gone by 60-70Hz (and the filter is setup so that combines nicely with the main sub to avoid any artefacts as we move firmly into the audible range).

Personally I think it was a great vfm upgrade, not essential but nice to have for sure.

 

It looks like JRiver's 1.0 limitation for Q (or really S, but Q in the UI) means many of these BassEQ settings can't be properly implemented.  I had previously created all of mine in JRiver 19, before they implemented notification that values over 1.0 are ignored.  I've recently upgraded to JRiver 20, and while it imported my old BEQ filters, including ones with a Q>1, if I try to edit it, I get an error message notifying me of the problem.  

 

One of the best features in JRiver is how easily it allowed me to implement BEQ.  This is a blow.

the problem is just that you can't copy the params straight into jriver and go for it, you'd have to spend time porting the published values to something that jriver can handle. Here's an example

the red line (in acourate) is the BEQ filter for the Pacific Rim LFE channel, you can see it's basically a fairly steep shelf filter that adds 28dB by about 6Hz and starts ramping up at ~40Hz

the green line is a LS with S=1 (Q=0.707) at 15Hz with +14dB

brown is the result of stacking 2 greens (to give 28dB total boost)

the REW trace is showing what happens when you run 1 and then 2 of those LS filters through jriver which shows that it can implement that filter

IMV that's a pretty close approximation to the intended curve and a much simpler filter to enter. If you want to fine tune it to hit the original BEQ filter more closely then you would just stack up some small notch filters along the curve to push it up/down as appropriate. It's not that hard to do if you can loopback jriver into REW as you can just iterate over it until you get a result that matches the target. Obviously it's a lot easier if you have a feel for how different filter shapes will sum up but that's just practice.

The only stumbling block is working out what the actual intended filter shape was. You could do this using the minidsp spreadsheet by plugging in the values and then copying out the values for dB by freq into another worksheet, summing those values up to get the final filter shape and then working out a jriver compatible filter. In fact you could do this all on paper without the loopback at all by plugging such values into a spreadsheet. I imagine this would be quite easy to do actually (albeit probably rather tedious work).

I find excel, i.e. me writing equations in excel, terrible at that sort of equation unless you break it down into many cells. Funnily enough though I used what you posted to calculate that post earlier and it worked OK