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SME

My living room "make over" (aka the "surrounded by bass" project)

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Yeah.  One of the things I want to implement and use in my custom software is self-test and diagnostics.  I really want my system to know within a reasonable margin of error how my system should respond in each channel in at least one key measurement location and to be able to give me a quick sanity check with a quick set of sweeps taken from that location.  Right now, I have no way to easily assess whether the subs are properly balanced, or if a knob got bumped (damn cats) or simply lost its memory because it's just a cheap POS.  Of course better quality equipment will help there too, but there's always opportunities for things to get knocked and nudged ... for faulty inputs to drop output in one speaker by 6 dB at random times, and so on.  For large immersive systems with many channels, I believe this kind of functionality will be essential to the sanity of whoever has to maintain such a system.

 

Apart from design work on surrounds, I also spent some time this weekend coming up with a scheme to use the last 2 outputs on my Motu A16.  I'm thinking I want two sub channels up front as I have now and a total of four subwoofer channels behind my sofa.  Part of my overall plan is to eventually use some or all of my subs to improve in-room accuracy of sounds above 100 Hz produced in other channels.  What hadn't occurred to me until I gave it more thought is that subwoofers are large enough that the box shape matters a lot, especially for higher frequencies.  And if there are multiple woofers with sufficient spacing relative to the wavelength being played, then the dispersion may become very directional and/or lobed.  But wait, that's an opportunity in disguise.  Imagine varying the phase difference between a pair of opposing woofers with respect to frequency.  The consequence is a speaker whose directionality is made to vary with frequency in a way that's optimal for the room.  In a sense, the sound can be more accurately "beamed" to where it's needed.  Now try doing this with subs in multiple locations, and you can imagine how powerful this concept can be.  It makes me really want to do 4 channels up front too, but I don't I'll have the 2U to spare.  And of course, the cost of amps for 6 channels of sub is already astronomical.  What would people think if I told them I had 48 kW of amp power (plus some spare change for the mains) in my living room?

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So Motu actually overnighted me my third A16 and made good on their promise to send a return shipping label.

 

I hooked up the new unit and ran through my tests, using a patch cable to check for a signal on each output and the ability to sense a signal on each input.  The display worked fine, and it looked like everything was great.  Then I turned on my amp.

 

The music I was playing sounded heavily distorted.  The distortion sounded a bit like the digital noise you hear on crappy computer sound card outputs, but it was modulated by the signal and got louder as I increased the level of the signal input.  More disturbingly, my center speaker was emitting this noise, even though I was playing music in 2 channel stereo.  More testing confirmed that the center emitted this noise even with signal sent to my subs, which were turned off at this point in time.  After playing with the patch cable, I was able to confirm that at least one output was producing a strong signal even when no signal was being routed to it.

 

It gets even more fun.  I turned it off and disconnected the cables in order to switch it out with unit #2, which at least has a working audio section.  While working on getting all the cables situated inside my rack, I felt a sudden sharp prick in one of my fingers.  Being a bit confused and thinking some kind of sharp piece of debris had found its way into my cabinet, I looked more closely and fished around with my fingers again.  I felt another prick.  Then I smiled, got up and fetched my multimeter.  I measured like 2V across the hot and shield of one of the cables.  Whoa!  Never before have I felt an electric shock while handling line-level cables.  Mind you, these were plugged into the *line inputs* of my OpenDRC units.

 

After thinking for a bit, it occurred to me that I probably bridged the connection between the ring (cold) and shield with the edge of my finger while I was working and discharged DC blocking capacitors in my OpenDRC units that had been charged to *at least* 2V.  So whatever electrical flaw exists in my Motu A16 #3 caused DC output at substantial levels.  Awesome!

 

I'm just a bit shocked (pardon the pun, but this really does describe my emotional state) right now and don't even know how to react.  Should I demand a refund and start shopping for product from another vendor?  This thing is/was to be the heart of my custom build, starting with the speakers I made.  I don't even know of any comparable products on the market.  If they won't give me a refund, I'll at least keep the unit with the broken display, but I am starting to worry about other problems cropping up with the unit that could damage other equipment connected to it.  At least my OpenDRC units, despite their seemingly shoddy design, used DC blocking caps.  Otherwise, I might have been smelling some smoke or something.

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I thought this was a feature of the motu interfaces? It has DC coupled analogue outputs which enables it to be used with software like http://www.motu.com/products/software/volta/path.html

 

I am not sure if there is any particular requirement on the cables when using such devices (http://www.expert-sleepers.co.uk/siwacompatibility.html suggests there might be, nothing immediately obvious on the Motu site) or if this is something your custom software has triggered in some way.

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Yes.  DC-coupled outputs is a feature of the Motu, and I should expect the same phenomenon (energizing of DC-blocking caps in downstream equipment) when intentionally outputting DC.  However, there should have never been DC on the output.  I'm not using any custom software thus far.  All I'm doing is routing analog inputs from my Denon AVR to analog outputs.  I believe the Motu inputs do have DC blocking, and I am almost certain that my Denon AVR filters DC from its outputs.  The DC is most certainly due to an electrical fault in the unit.  I could try doing more testing to better characterize the problem, but unless Motu wants to pay me for doing their quality testing and troubleshooting for them, I'm not really interested messing with the unit anymore.  And just to emphasize, the distortion I heard on music playback was severe and was always at least as loud as the actual music.  Thus far, this third unit is the least useable of the three. 

 

I did send them an email last night describing the problems in detail and asking if I could return the third faulty unit instead of the second faulty unit.  I also gave them the benefit of the doubt asked them *not* to rush ship me a fourth unit and instead to take their time to figure out the problems on their end.  Receiving one defective unit is unlucky.  Receiving two defective units in a row is *very unlucky*.  Receiving three units in a row where the third unit was shipped to me with very expensive overnight shipping is a sign of something seriously wrong.  Maybe someone just swapped the "good" units stack with the "bad" units stack by accident.  I hope that's all it is, because I still really want to use this product and like this product. 

 

Nevertheless, I'm losing patience and confidence.  I expect them to get this sorted out and send me a fully working unit within a week or two.  Otherwise, I will be demanding a partial refund discount for unit #2 with the bad display and will actively warn others away from this product.  It's just sad.  I would happily pay $2000 instead of $1500 for the product if that's what it cost for them to implement some proper quality control.  But I guess cheap reigns supreme in the marketplace, and in the end, we all pay for this unfortunate fact.

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Wow...Sorry about your string of bad luck.

You are correct there should not be voltage across your outputs with no signal. DC coupled in marketing literature means that the unit has the ability to pass a DC current from the outputs but there should never be voltage present on all of those outputs even with no signal present. Something is funky somewhere.

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The capabilities I need from the Motu include the analog inputs and outputs, the USB digital audio interface, and the AVB audio support.  The front panel display is not essential, but it is very nice to have because it displays both continuous average and peak hold levels for every analog input and output as well as the current sampling rate.

 

Motu asked me to verify that the second unit is not exhibiting any of the symptoms seen in the third unit.  I will probably test it more thoroughly this weekend, ensuring I've listened to sound from every output as well as operating with the AVR outputting a channel through each input.  I'll test the cables with my multimeter to ensure the caps don't get energized again.  I may also try to verify that a DC voltage presents itself on the third unit with it powered on.

 

And hopefully I'll get time to do other stuff this weekend.  I've got some design issues to think through before I can write more code, but after that, I hope to be able to implement the bi-quad and delay processing ASAP.  With that, I can start experimenting with crossovers until I figure out what I want to do for FIRs.

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I got busy today.  With help from the wife, I got my second XPA-5 installed into the rack above the first one.  It involved plenty of swearing, but we managed to get it in a lot quicker than we expected.  Then I made some new speaker cables and got the new amp connected to the Motu A16.  When I was done, I dragged out the new speakers, set them up, and routed the left in to the left woofer and horn and routed the right in to the right woofer and horn.  I don't have any kind of crossover yet, except for the protective 40 uF in series with each horn.  I do have the speakers crossed to the subs at 110 Hz (same XO as used currently by the old speakers).  I used the A16 web app to set the gains of the four channels by ear.  I'm not using any room EQ.

 

I am absolutely amazed at the sound quality despite the lack of crossovers and the fact that the horn CDs are mismatched.  Yes, the sound is heavy on the upper mids due to the extended bandwidth of the AE woofers, and the left horn has quite a bit more top end than the right horn.  I also don't have any delays set between the drivers much less room EQ.  Nevertheless, the imaging is remarkably good and in many respects, it exceeds the performance of my current speakers in their fully optimized/calibrated configuration.  I'm sure the imaging will improve a ton once I start getting some filters on these guys.  :)

 

Oh yeah, on another positive note, the noise floor with the MiniDSP OpenDRC-AN units removed from my chain is much lower.  The speakers are hooked up to the amp, which is hooked up to the Motu, which is hooked up to the Denon.  All connections are balanced except between the Denon and Motu.  The noise floor is still audible, particularly in the horns, but it's not very annoying at all.  With the MiniDSP units in the chain, we could hear the horns hissing from the bedroom of the house.  So it does appear that the noise performance of the Motu is quite good.  If need be, I can still pad down the sensitivity of the horns a bit, but I may be able to reduce it a lot by improving my cable hygiene.

 

This second Motu unit seems to be doing fine with audio.  The display actually worked really well for about 24 hours in a row, but it's back to being blank again.  There's hope that I can make this thing work.  :)

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Just a quick update.  I'm still using the Motu with the broken display and haven't heard back from Motu for a couple days now.  I hope they are working finding me a fully working unit, but I will ping them tomorrow if I haven't heard anything.

 

I feel slightly silly for listening to these speakers without a proper crossover, but they do sound quite good even without.

 

I've been working on the PC that will run the DSP code.  It's set up with the Motu on USB and Jack audio server on a PC running the Linux Mint 17.3.  It took a bit of tweaking, but I got things up and running without too much hassle.  This stuff is way easy compared to 15 years ago.  My informal tests suggest I can reliably get latency down to 2 ms with the kernel I'm running.  I bet I could go lower with a different kernel, but I'll probably opt for a built-in latency of at least 10 ms, probably more to improve computational efficiency and make room for some non-minimum phase correction.  I can also confirm that the system (without my software added) successfully passes audio.  So it looks like the Motu A16 and Linux work fine together via USB.  That simply rocks.

 

So now I'm back to just code.  We'll see how I progress, but I hope to be working on a crossover by the end of the weekend.

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Are you at least running a cap to protect the CD?

 

Yes, absolutely!  I have 40 uF in series with each CD, but that's all.

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I hadn't read those threads yet.  Thanks!  I get the impression that bass addict wasn't so much dissatisfied with the 2 X TD6  speakers he built for "heights" as he was dissatisfied with the overall result, particularly the TD-15M + SEOS-12 + BMS-4550 builds he did.  I can try to guess at what he might of done wrong, but it's largely speculation without some good measurement data.

 

Another important point is that the design uses TD6M drivers.  I don't know why anyone would try to use TD6Ms as woofer because they really are mid-range drivers.  Notes from mtg90 on the DIYSG forum indicated that the mid-range was brought way down to deal with the bass roll-off resulting in a sensitivity around 92 dB/2.83V and 4 ohm impedance.  That's worse than what I have now.  For my design, I planned to use TD6Hs instead.  They have a lot more excursion capability as well as more mass and motor strength for use as actual woofers.

 

I sit about 3 m away.  You would be correct to say that my build is a bit overkill for reference level playback.  But that hardly puts me in strange company around here.  :)  I want something that'll keep up with my mains and deliver excellent sound quality.  The TD6H pair with help from the corner can probably do about 3 dB less than my TD12M mains.  I know it's typically argued that surrounds need not be as big, but I do want a lot of headroom.  For one thing, I'll be sticking to 5 mains channels in this room, so the surrounds will have to handle sides and rears on 7.1 tracks.  For another, Atmos tracks and tracks originating from Atmos theatrical mixes have a lot more bass in the surrounds because of the surround bass management that theatrical playback systems added for Atmos.  Third, I want to use the surrounds to help with room EQ, even for sounds that originate in the mains speakers.  As such, I really wish I could make a pair of 8s work, but that causes too many other issues in the design.

 

 

Thanks a lot to you, Ricci, and 3ll3d00d for your suggestion.  If I decide to go with a coax, I'll definitely lean heavily in that direction.  I do wish they made a 4 ohm version so I could get a bit more power into them.  Both this and the Eminence have sensitivity of 95 dBish.  I expect a TD6H pair to yield 97 dB.  Of course, the Radian probably has a bit more bass if I end up needing it, but I can't find an Xmax number for it.  Can anyone help?  The other reservation I have is that I'm pretty sure I want the tighter horizontal dispersion of the SEOS-15 horn turn on its side.  I'm fairly certain it will help a lot with front wall reflections I have.  I know because I've done a lot of acoustic measurements already with speakers that are already installed in the planned location.

 

 

Lemme know on the radian if you decide to go that route. I can supply an XO schematic if you want to do that...or you can roll your own :)

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Lemme know on the radian if you decide to go that route. I can supply an XO schematic if you want to do that...or you can roll your own :)

 

Thanks for the offer, but I'm going fully active.  That's partly what the Motu's for ... and the two Emotiva XPA-5s.

 

In the long run, the crossover will be done as part of the room EQ.  I can take measurements from each driver (or sub channel) and then run simulations that accurately predict the responses for whatever arbitrary filters I choose.  Then I can iterate on the filters until I'm happy with the results.  In time, I'll probably experiment with some automation of this process.  I've already done this with remarkably positive results for my subs and MBMs with a MiniDSP 2x4 on the sub channel.  I also use filters via OpenDRC-ANs on each of my mains to improve their integration with the subs and MBMs.  With the Motu up and running and the software improvements I'm working on, I will be able to customize filters on each bass driver in my room for each input channel (i.e., FL, FR, FC, SL, SR, and LFE) independently.  for example, say you're trying to optimize your sub delay for the best crossover response with the mains and you notice that 3 ms looks better for FL but 7 ms looks better for FR.  Well with full-matrix DSP, I can have it both ways.  I have 9 different bass sources in my room right now, and depending on what subs and amps I buy, I may end up with 11.   Each of these 11 sources may use a different set of filters for each of the 5.1 or even 7.1 channels.  (Even though my system will only be "5.1", there's no reason I can't customize the mix-down of the 7.1 channels to the 5 treble and 11 bass sources.  This whole concept really blurs the notion of what a speaker is and how channels should be counted.)

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Tonight, I installed my fourth Motu A16 unit.  Thus far, it's working flawlessly.  That's a big relief!  It looks like this one was a brand new one that they opened up and tested by hand.  As frustrating as this experience was, I'm glad they followed through on getting this solved for me.  They also sent me another return shipping label.  So far, I've been very happy with unit #2 aside from the bad display.

 

Other than that, work has been a real pain lately, and I've managed to lose some of my time to other discussions on this forum.  Nonetheless, I am very close to testing a live version of my custom code.  I think the code that does the actual processing is working fine, but I want to see some good automated test results before I go live.  I don't want to know what 50V to my horns would do to them or my ears in the events I somehow manage to send full-scale white noise to them.  Right now, I'm struggling with the Jack audio library.  It mostly works well, but does have bugs as well as confusing behavior that's not well documented at all.  In any case, it's just a matter of time before I find a work-around to the issue and get the tests fully working.  Hopefully I'll be doing measurements and/or crossovers by the weekend.

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I did it!  I am running my first live test with my custom DSP.  I instituted a work-around for the Jack library problem.  Although it's only really a partial work-around because I still intermittently fail.  My problem may simply be that the Python code isn't realtime enough for Jack to not be fussy about it.  The actual DSP code is written to be completely realtime safe.  I did find and fix 3 bugs in my delay code, and I *think* it's flawless now.

 

I'm listening live now with my own delay compensation active, and I confirmed its correct action by measuring a few impulse responses with REW.  Woohoo!  The imaging is definitely a lot better now.  The upper mid-range got a bit stronger, as I expected.  It measures 6 dB hot, which is exactly what I expected.  And in fact, my level matching by ear of the woof and horn got them within 1 dB or so of flat.

 

Tomorrow it will be time to migrate my existing bi-quad code over to my new stuff.  Most of the work is already done for implementing specific filter types like HPF/LPF/peak/dip/shelf once I've migrated the generic bi-quad stuff to the realtime engine.

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No progress on DSP yesterday.  Instead, I discovered that I have a problem with electrostatic discharge (ESD).  It looks like only one of my three racks is actually conductive to earth ground, and it conducts to ground via my amps through my line-level audio cable shields into the Motu, which has a 3 prong plug.  The other racks, including the one with the Motu in it are floating.  (The anodized coating on the Motu rack ears is non-conductive.)  Making matters more frustrating, the racks themselves don't even conduct between the panels because of the paint.

 

I will probably have to wire the racks and their panels together and to a good earth ground.  Until then, I'll need to walk over to a light switch or outlet in another part of the room to discharge myself before doing anything in the racks.  That's a big inconvenience, and it makes doing measurements an even bigger pain because I always need to make sure I'm earthed before touching the mic (or even the stand) to avoid zapping it and everything else connected.  This is only a problem now because got I solid metal racks.  Before I was using a wooden A/V shelf, and I could easily reach in a discharge on a computer chassis.  Now I have to touch one of the racks to even to get to a computer chassis, and discharging into one of these un-grounded racks is enough to cause a significant audio pop.  Not good.

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I got bi-quads working and adequately tested late Sunday evening.  I've hacked together some filters to get a response that looks pretty damn good but is still a long way to achieving my ambitions.  To do this job right, I need to do a ton of measurements at my different locations, and I'm currently lacking the ability to do sine-sweep measurements using the outputs on the Motu directly.  Fortunately, the code to do this is mostly already done.   mostly exists already. 

 

I've got a 2nd order high pass on the horns and a 4th order low pass on the woofers at 1100 Hz.  I also have a healthy dose of BSC on the woofers and a fair bit of shaping on the horns.  After shaping, the crossovers are closer to 1300-1400 Hz or something like that.  Last night, I worked on bass response and sub integration.  Even with the overdamped response that starts rolling off > 100 Hz, the room gain goes crazy down there, and the woofer interferes with the subs down to 40 Hz, even with a 12 dB/octave crossover engaged.  A bad room mode at 62 Hz or so causes the mains to interfere there even with a 24 dB/octave crossover at 100 Hz!  BTW, I expected all of this based on experience with the speakers these are replacing.

 

Anyway, my response now looks half-decent, but I'll probably still tweak it a bit more.  Otherwise, I'd post some pictures.  I'm actually seeing some acoustic stuff in my measurements that I'd like to understand better.  It's apparent that the larger baffle helps a lot with holding directivity down all the way through the midrange.    My impulse response looks very nice from about 200 Hz on up, and I believe the suck-out I see at 150 Hz will mostly disappear with ceiling bass traps.  I've installed these toed-in about 50 degrees as close to the back wall as possible.  There is a 4" OC703 panel on the wall on the inner side of each.  The OC703 absorbs pretty well down to to the mid 100s or so.  Below there, we actually want to let the bass to pass through so it can benefit from the nearby boundary gain.

 

With my latest EQ iteration, these are sounding absolutely fantastic!  The imaging in the sweet spot is in full 3D.  Visually, the speakers form an imaginary box, and the sound seems to emanate from that shape as if it were a window.  The phantom surround is also superb.  I can very clearly hear sounds pan across the sides and rear of my head with the right content.  It would be easy to convince just about anyone that they were listening with the center and surrounds turned on.  Of course, the magic falls away when leaving the sweet spot, but the sound stage never collapses completely, and some of the phantom surround effects can be heard in locations well off axis.  That's just damn cool, and I know I'm not the only one here running with an SEOS or something comparable to it.  :)  And the bass?  Nothing short of amazing.  The speakers cross at 110 Hz until I get around to re-doing all the sub EQ, but there is definitely a lot more slam than there use to, even at lower playback levels.  The difference is apparent even when listening in the kitchen.  Kick drum just bumps a bit harder and faster.

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Just a quick update here.  I have been spending more time enjoying the sound of these speakers instead of working on them.  Over the weekend, I worked on integrating them with my existing center and surrounds.  Listening revealed holes in the sound-stage between left, center, and right, and measurements with both the center and left (or right) speaker playing showed a big dip in the 1-4 kHz range.  Sounds like a timbre matching problem, right?

 

Using a short time window (like 2 ms), I compared the phase response of each type of speaker and noticed that the center went through a 180 degree rotation at right around 4 kHz.  My left and right speaker went through a 180 degree rotation near 1200 Hz or so.  This accounted for the nasty 1-4 kHz dip and the front stage hole.  What to do?  I added 2nd order all-pass filters centered at 4 kHz to the left and right so they would have the same phase rotation as the center.  Then I added a 2nd order all-pass filter centered at 1200 Hz to the center and surround channels.  The all-pass filters did the job nicely, and the sound stage is actually quite cohesive in the sweet spot.

 

Of course, this quick fix doesn't address the fact that the speakers have different dispersion characteristics and thus, different power responses.  Nor does it address the fact that the speakers I just built sound a lot better, overall.  But at least I can continue to enjoy multichannel content until I have a center channel built and installed.

 

My other issue is that the bass is no longer integrated as well as it was.  To fix this, I need to get a lot of measurements at multiple seats, but I don't want to use my old code to do this because it won't work with the Motu directly.  I need to write code for a significant prerequisite component, and then I can port my existing sine-sweep measurement code to use the Motu directly.  Once that's done, I can move my subs off of the MiniDSP 2x4 and OpenDRC units and ping each of them separately during a single measurement run.  This last point is a huge deal as it takes me a long time to measure 10-12 or more listening locations and having to cycle the power switches on the plate amps for each location to get individual measurements from each of the four subs.

 

I also need to finish converting the code I wrote to calculate room EQ filters into a generic program.  That's where I simulate the filters applied to the measured room responses and iterate on the filters and filter generation algorithms until I like what I see.  Better still, by having a more direct interface to my DSP than I did through the MiniDSP systems, I'll be able to listen to the results of each iteration with great ease.  I'll also be able to do A/B switches nearly instantly as well as manage an unlimited number of pre-sets.

 

This weekend, I will try to get measurements of the horns themselves so I can settle on their design.  As much promise as the code holds, I want a center channel real bad now that I've heard what these things do for movies.  :)

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Sorry for the lack of posts.  I've unfortunately been occupied with "other" stuff and haven't had much time to work on this project.  Work in particular has been a real bother.

 

I have spent some time thinking about subs again.  Even though I'm probably months away from working on them, I'd like to get the sub drivers ordered well in advance.  I think I'm almost there.

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I just posted some CD + horn measurements I took over the weekend.  I was excited to try 3ll3d00d's idea of using a compass app on my phone for the horizontals.  That turned out to not work as well as I hoped.  The magnetic field from the horn overwhelmed the earth magnetic field out to like 24" vertically.  I imagine I'd needed even more clearance if I were not placing the phone directly above the magnet.  I ended up having to stack a bunch of stuff on top of the horn box to make a place for the phone to lie far enough away from the magnet.

 

In any case, I'm quite happy with the data I got, but I wish I had time to do horizontal polars at different elevations.  My horns will be installed above ear level, so I need to optimize both toe-in and vertical angle to get the best seat coverage.

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fwiw (and I might have posted this before but I don't recall off the top of my head) I wrote a v dirty script to turn REW measurements (or rather, a set of frd files) into various directivity sonagrams -> https://github.com/3ll3d00d/directivity-utils

 

it just needs cygwin, gnuplot and awk (or just run it on linux)

 

perhaps you'll find it useful for visualising & comparing directivity (I find it makes anomalies jump out at you more so than viewing all the raw measurements as FR charts)

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Cool, thanks!  I like the 2D color map plots for a quick glance of the overall pattern, but I find the polar plots to be more useful when I'm studying things more closely.  It does help to be running REW and able to zoom in cleanly to see detail where I need to.

 

I just ran the tool, and it didn't generate the polar graph right, but I did get a good 2D color-map filled-contour plot.  I added it to the post.  Feel free to give me shit about normalizing to 37.5 degrees, but that's about where I think it'll land in my space.  As such, I expect the middle region to run a bit hotter because the listeners who will hear that sound will be sitting farther away.

 

BTW, this thing holds pattern to like 600 Hz.  That's a bit wider than advertised isn't it?  Perhaps it's the box that's doing the job?  Even though the box is only about 1" wider at the ends than the horn?  Now I think I will want to get horizontal and polar measurements of the woofer unit as well.

 

Edit: OK.  I just added a 2D color-map for the vertical polar measurements as well.  This time, the tool choked on generating the regular "sonogram" but the "normalized sonogram" turned out ok.

Edited by SME

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one thing to bear in mind with indoor measurements that have such a short gate is that the low end of the CD response is impacted by a small but appreciable amount. Here's a comparison of mine indoor vs outdoor, same driver (bms4550/seos10) horn combo in each instance

 

post-1440-0-84677100-1460533683_thumb.png

 

post-1440-0-00175400-1460533689_thumb.png

 

I suspect you'll find this impacts your reading of directivity on the low end which might be important for considering where to cross.

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Yep.  As the CD rolls off, it exhibits increasing phase delay, so even though the 3 ms should in principle be long enough to assess response down to 333 Hz, the data is probably only accurate to 600 Hz or so because of the added phase delay.  For my own purposes, I can increase the window time quite a bit and accept some minor error/ripple in exchange for better insight into performance below 600 Hz.  Just now, I did exactly this to confirm that the response from 600 Hz and above doesn't change much at all.  The reflections that do occur at 3 ms and later are very weak, even for bass frequencies.  But I opted to stick with a 3 ms window for data posted here because the IR is 100% reflection free up to that point.

 

IMO, the bigger issue with the low frequency parts of the measurements are the baffle effects.  The baffle effects likely have a significant effect on the horizontal polars starting around 1 kHz and on the vertical polars starting around 3-4 kHz.  I believe baffle effects explain the apparent constant vertical pattern at +/- 50 degrees or so all the way from 1500 kHz down to 600 Hz or so as other measurements indicate the vertical pattern continuing to widen from 1500 Hz on down.

 

BTW, I'm not sure what you are trying to demonstrate with your picture.  It looks like your indoor response is about 2 dB lower than outdoor from 500 Hz on up.  This doesn't make any sense.  I would expect the responses to be near equal.  If you used a long time window for the indoor measurement and had very long room decay times, I could see the indoor response being a bit higher than the outdoor response, but not the other way around.  In my room, the response above 3 kHz or so looks almost exactly the same with a 1 ms window or a 100 ms window and a very modest amount of smoothing (say 1/24th octave).  I can't think of any good reason that outdoor measurements should show a 2 dB bump in output like that.  If you were to adjust the green curve down by 2 dB, would it not overlap the pink curve very closely down to 600 Hz?

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