SME

Excellent, thanks for the update! Edit: It looks like your BMS driver may be exhibiting diaphragm break-up around 17 kHz just like the DNA-360 as evidenced by the considerable beaming there. I used to think this kind of feature was due to a mismatch between the CD exit and horn throat, but better measurements indicate this is break-up, at least for the DNA-360. I'm curious about how much output you can get before that break-up goes non-linear. If you are up to it, you might try doing the sine-sweep tests I did. Obviously, care should be taken when doing the higher level sweeps. I looked for evidence of power compression or excessive distortion after each sweep before going forward, just so I didn't damage anything. I finally got a bunch more measurements done including a full redo of my DNA-360 / SEOS-15 measurements. My hunch was correct in that the compass app on the phone did not give me the most accurate angles. As the horn was rotated, the error in the reported angle from the compass app accumulated. Here's before: Here's after: Both of these are normalized to 30 degrees instead of 37.5 degrees (as was in my original post) because 30 degrees looks better for the MLP now that I have good data. Note that the horn horizontal dispersion appears to be 90 degrees instead of 120 degrees, as it should be. After doing the new measurements, I discovered an unwanted filter in-line, so I'm only updating the normalized maps until I have time to apply the inverse filter to all the data I just collected. Whoops. I also did some measurements of the woofer in its box, which are posted here. While I was at it, I also did a set of polars at 10 degrees elevation. I am hoping that I can assume that the response is approximately separable, i.e.: P(theta, phi) = P(theta)*P(phi) for pressure P and azimuth and elevation angles theta and phi. In terms of SPL, that can be written as SPL(theta,phi) = SPL(theta) + SPL(phi) + C. where C is some constant. If the response is separable, then I can estimate the polar response anywhere on the unit-sphere using only the data along each axis. I'll use the elevation 10 degrees data to try to verify this assumption. If this works out, I can use the data I have to model the response at each listening location with any particular crossover as well as to calculate the power response and true directivity with that crossover.

If you can make it work with your system, the nanoAVR is the best choice I'm aware of that is more-or-less plug-n-play for this kind of thing. Others can speak for whether JRiver and a good computer audio interface can be made to do this well, but I understand there are some issues getting proper shelf filters implemented.

Just a quick update. Progress has been slow due to lots of other real life stuff getting in the way. I have been doing some more pre-design work on the subwoofer subsystem. I hope to be able to order subwoofer drivers very soon. Last weekend, I attempted to do polars of the SEOS-15 with the alternate compression driver, but I couldn't pull it off using the compass method at all. The magnet was just too strong even with the phone something like 3 feet away. Worse still, the magnetic field seemed to cause the phone's magnetic directional sensor to progressively lose accuracy as the phone was rotated. It was only after I got to about 45 degrees that I realized the compass was off. As such, I'm very suspicious that this problem may have fouled up the horizontal polars I posted for the DNA-360 + SEOS-15. As such, I am going to re-measure it using a different method that I hope will be much more precise. Hopefully I will be able to do that this weekend.

First of all, the DVD and the BD may or may not be from the same master. I don't think anyone here knows for sure as no one has replied to your question. However, even if the mixes are the same, your AVR may play back the DD track on the DVD with attenuation called dialnorm. Typically, dialnorm reduces the signal by -4 dB, but it may be more or less, if whoever produced the digital bits overrode the dialnorm value to something other than the default. My Denon AVR always flashes the dialnorm gain applied at the start of a bitstream. Second of all, if you use Audyssey with Dynamic EQ, you may be getting different for the playback of each one because Dynamic EQ seems to operate based on the MV control without taking into account dialnorm. If you turn up the MV to play the "softer" DD track, then Dynamic EQ is providing less correction than it is for playing the BD. Dynamic EQ tends to boost the deep bass much more than the mid or upper bass, so the BD probably plays with stronger deep bass relative to the rest of the spectrum. Of course, it's also possible that they are separate mixes. As you would be right to point out, until someone does a comparison, we won't know.

OK, that makes way more sense! The impulse responses are very interesting to me. I'm not talking about the part after 2.5 ms or so, where I expect to see some reflections. The significant difference in impulse response, from roughly 0.5 to 1.0 ms is interesting to me. Sound moves roughly 1 foot in a millisecond (ms), so whatever is causing the indoor measurement to differ from the outdoor one is very near the horn or the mic. Any thoughts as to the difference here?

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?

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.

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.

I look forward to trying this out! I'll be able to program it into my prototype Motu/PC-based DSP. All I really need to do is move the bass management from my AVR to the Motu/PC and then I can just program in the filters. I'm curious as to how the mid bass bump will affect things for me. I find myself quite satisfied already with the slam I get from my system with most movies. In fact, I recall thinking in the movie theater: "this is going to slam nicely when I hear it at home if they don't screw up the mix". Do you think the BD mix was altered to have less mid bass? Or do you think the theater you went to used a mid-bass bump for effect? Or do you think there's some kind of acoustic secret sauce in large rooms that makes mid bass hit harder than it does at home without more SPL? Or do you have no clue and just like the sound of the BD with more mid bass? Don't mind all my questions. I'm just curious about your thoughts and experience. FWIW, I run a full-time house curve of about +0 dB @ 20 Hz to -5 dB @ 300 Hz and then flat out to 12 kHz or so. It just sounds more natural to me that way than running flat or using a treble roll-off. I'm sure you've noticed one of the nice thing about having good bass absorption in the room is that the extra bass doesn't really add in the way of "bloat" or heavy sound. It mainly just makes stuff hit harder.

Wow, I'm real excited for this one and the potential for BEQ to make it even better.

I bought a WT2 to use for my builds. It does have a pretty clunky 1990s interface that runs only on Windows, but it did basically work out of the box without much fiddling. I found it very helpful to just hook it up and go when I needed to verify my recent speaker builds. I would recommend it to him to at least get a good grasp of the parameters of his subs. It would also be useful to observe any changes occur over the course of break-in.

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.

Thanks for the reply! I'm thinking of building a very shallow enclosure only 9" or so deep for a 15" driver. The purpose is to squeeze it between the rear wall and sofa. The face opposite the driver can't accommodate the amp because of the extra clearance needed for cables and connections. So I have to install the amp on the side, which is only 9" wide. The bigger, taller, but narrower amps work fine. Will that 1200W amp be rated at 1200W into 4 ohm? Or will that be a 2 ohm rating? I should probably not base my design on an amp that isn't on the market yet, but this 1200W amp sounds like a perfect fit.

I admit that I'm also sad to see the Speakon connectors go, and the DSP features are throw away to me. But I realize not everyone feels this way. Actually, the delay knob will be an excellent feature for more typical home users who don't have or want the complexity of an outboard DSP unit. The drop in price is a big plus to me because what I am contemplating will require a lot of different amps. Now what I'd really like to see is a single channel plate amp to fill the gap between the SP1-700 and the SP1-2400. It appears there is a 2 channel plate amp available with 2 X 700W, the SP2-1400, but unless it's bridgeable or else one can safely run one channel of each voice coil in a DVC driver, it's not really useful to me. Its footprint is also too big for the subs I'm planning. Haha!

YES! That explains what I heard in the "better" theater: hints of < 30 Hz bass, but only hints. I'm glad to see that the gentle slope doesn't roll-off a cliff further down! Can anyone say BEQ?

All this sounds very exciting, and I look forward to more info as it becomes available.

My suspicion lies in the measurement device, which in this case is the scope. In theory, the scope should be able to measure True RMS down to ULF very accurately, being that it does digital sampling. The trouble is that the actual algorithm implemented on the scope may not do this well. I honestly have no idea if this is really the case, but it seems plausible. Can the scope report the peaks instead? If the waveform is undistorted and you are always testing with sine waves, then you can back calculate Vrms from Vpeak by multiplying by 0.707. Edit: I mean multiplying, not dividing. (*head smack*)

Hi Brian, Can you explain what's going on in this picture: Why are there four sets of amplified outputs for a 2 channel amp? Or is this a 4 channel amp?

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.

Thanks! I couldn't find a the link for that anywhere on the rest of the web site. It looks like those specs are for the Pro models. I wonder if there's a specs page specifically for the home theater models? Another bit of information I'm curious about is how many amps are recommended on the circuit to run each amp. There's a picture of the SP1-4000-HT, which shows "120 V / 8A" on the nameplate. Unfortunately, the exact same picture appears for the other plate amps, and no similar picture exists for the rack amps. I'll probably have to email about it or something.

Can you please point me to these updated spec sheets? I can't seem to find them.

Saaweet! Looks like my currently planned build just got a bit cheaper. And I see that the HT models gained the 12V trigger feature as well. Very nice!

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.

Josh, Have you measured the responses of each bass-managed mains channel to see how well the mains integrate with the subs? I remember when I first EQed my subs to flat, I was very happy until I did my sweeps of the bass-managed mains channels. Optimizing sub distance was nowhere near enough to get a good blend. Even with the the ports stuffed and a relatively high crossover, I was still seeing considerable interference from the mains all the way down to 40 Hz. That's a big part of what led me to put OpenDRC-ANs for DSP on my mains (big improvement!), and now, the Motu A16 + PC DSP with full matrix support. And yeah, one should not underestimate the impact of inductance on sound quality, even when using lower XOs. Anywhere there are dips in your room response for any one of your mains in the 80-200 Hz range, there's a chance you'll be hearing more sound from your subs than from that mains speaker, despite the 24 dB/octave roll-off. Having the subs be able to reproduce that sound cleanly does improve sound quality.

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.