3ll3d00d

For most of the measurements, there is a pretty nasty looking dip somewhere in the 600-700 Hz range, but it does shift slightly between locations.  Somewhat less obvious is that there are peaks, one at round-about 850 Hz and another broader one at 350 Hz.  Below there, the responses roll off due to a combination of baffle step effect and 110 Hz crossover.  

I wouldn't trust the data from a 4ms window here, at least certainly not for the 350Hz and below part and the 600Hz might be questionable as well.

there were some polars linked from http://www.audioheritage.org/vbulletin/showthread.php?33972-JBL-Master-Reference-Monitor&p=381408&viewfull=1#post381408which seem to come from http://www.soundandrecording.musikmachen.de/Magazine/SOUND-RECORDING/2014/12/JBL-M2-Master-Reference-Monitor-Testbericht

no baffle step compensation as the woofer is loaded by the floor so I knew that that would add something but I found it hard to get a reliable reading on exactly what it would add. My best guess is that it would add more than baffle step under 150-200Hz hence I decided to ignore it and rely on EQ to compensate instead.

I know what you mean re letting it sink in over time. Personally I find back to back A/B a bit of a fruitless exercise, I find that something either sounds wrong pretty immediately or an itch develops over the course of a few weeks. If either happens then it's a signal to drill into what is going on, if not then I leave things alone. This tends to lead to bursts of activity followed by long periods of stasis (with respect to the configuration), last time was after the subs changed ... took ~3months to dial in them in but then that configuration was then left unaltered til now (~18months later).

I mean anechoic flat, there is the usual mess caused in room though I haven't had time to measure except to quickly verify levels. I thought I'd live with it as is for a bit and then get into EQ'ing to give me a better handle on how it actually sounds in itself.

What are the diffusers made out of btw? also why place some above the speaker, I didn't catch what the rationale was there.

btw I have just got my speakers up and running, atm I'm running them flat as I haven't had time to do any sort of in room measurements. I was slightly surprised to find that I don't find them forward/bright at all. What this says about the room and/or my hearing is TBD

Anyway, from what I've heard, it sounds very good, but the one thing that bothers me is that the dialog seems to originate more distinctly from above than I'd like.  That surprises me because the horn is only a few degrees higher than the horns on the left and right are.  Perhaps elevation is easier to hear after it passes some threshold?   

have you considered that this is what your eyes are telling you? I don't recall the details off hand but IIRC the chapter on localisation in https://auditoryneuroscience.com/topics/book-previewwent into some detail on how the visual system influences this.

I haven't looked at other platforms tbh, not many have the ability to set something that low down.

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)

what are your main subs going to be? 

Are you concerned about cancellation of the main subwoofers output at the seats from the nearfield subs at / near the high pass?

I would think he's using an appropriate FIR filter to deal with that, at least that's how I handle it

I would have thought the local environment is a major contributing factor here. I can't realistically use all of the capability I have in the house because I live in a densely populated city, have neighbours who I respect (and who haven't minded on the occasions when it does get somewhat feisty in here!) and have ears I don't want to damage further (tinnitus which I would assume is derived from years of standing too close to PA speakers in clubs).

Normal viewing levels for me are ~ -10 (bass has a ~4-5dB rise by the bottom end), -5 probably about as loud as it ever goes for normal film viewing). No idea what that equates to on the scale referenced in the original post.  

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

I've seen the capability and flexibility of using the Motu/JRiver combination from mojave and it's really impressive, but I don't want to rely on running absolutely everything through JRiver.

I would say your choices, for video playback, are;

jriver as the source

or

normal consumer equipment as the source + some opendrc model OR a PC running as a custom audio processor fed by an appropriate set of pro audio interfaces 

There are a few options for generating/designing FIR filters

I do use acourate generated filters in jriver, there is a cutdown version that runs on opendrc hardware

some others (like mojave) use audiolense

rephase is a free option, this one is a largely manual filter designer

I came across http://aplaudio.com/conc2/products/tda-eq/ recently, no idea if it's any good

I have a motu now (the 1248) as well as the PSU on my RME died. The motu mixer app is served by the hardware via a built in web server so is accessible via any device on the network (assuming you have a network cable plugged into it of course).

I don't think there are many options outside of a PC or a minidsp for running arbitrary FIR filters though. You can get devices like a DEQX but they're 1) expensive, and 2) often limited to a preselected set of steep crossover slopes.

Does the OpenDRC have the same issues as the 10x10? It's a different hardware platform after all.

 

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