SME

I forgot that Ultimax come in 12s and 15s. I so rarely see the smaller versions get used in builds. Most probably don't see a point when 18s only cost a bit more, unless they really need something small. Is this a mixed-use room? If not, why the concern about looks? Very small subs generally provide poor performance to cost ratio, so you give up a lot for looks. Given your budget, I'd suggest going with vented enclosures. A lot of details would depend on your room behavior. If your room is like mine (which is slightly bigger, albeit open to a hall), you'll have a huge amount of room gain in the 20-60 Hz area, in which case you could probably get away something fairly small (but limited in output at the low end). Using a smaller 12" or 15" driver and a "tower" form-factor you might be able to keep the footprint especially small.

Is that 0.2 mm with or without the weather stripping? I would assume the tolerance would be less important with the "weather stripping" in place. BTW, is that actually weather stripping? Or is that gasket tape? I looks more like gasket tape in the picture, which is fine because that seems like the better choice. I'm also more inclined to put the gasket tape on the port blocks for better aesthetics and to avoid obstructing the vents or cause unwanted flow separation. I'm not sure if typical gasket tape will be durable enough for frequent addition and removal of the plugs. I'll keep thinking on it. Thanks everyone for the input.

I'm not familiar with the feature, so I am not inclined to comment on it. However in general, Hornresp reports output in terms of *acoustic power* and not *SPL*, which is important where directivity is not omni-directional (or the 2pi equivalent, as it were). When you start stacking large cabs, a few things are going to happen regardless of the type of cabinets. First is that directivity will typically increase for progressively lower frequencies as the size of the array grows. This means that at a particular location and frequency, SPL may be quite a bit higher or lower than the dB value indicated by Hornresp, for better or worse. When indoors and in smaller rooms, the acoustic power may be more important than the anechoic SPL output anyway. Second is that the co-location of multiple sources will tend to boost power output and efficiency, but this depends on the distance(s) between sources and the wavelength. Because lower frequencies have longer wavelengths, arraying will tend to boost lower frequencies more than higher frequencies. In short, don't confuse SPL output, which depends on directivity, with acoustic power output, which does not and which is what Hornresp generally shows. I'm inclined to doubt that a horn stack will increase in LF extension beyond what a single horn can do. Can it increase output and efficiency down load? Yes, absolutely, but that's not the same thing.

Can you clarify what you mean by 85 dBC? Is this a calibration target? And if so, using what kind of test signal? Also, have you considered that a more neutral sounding "house curve" may require in-room SPL that's quite a bit higher than a flat in-room response? This "house curve" should arise naturally when using an anechoic flat monitor. As calibrated/optimized and while listening at an overall level appropriate for monitoring typical "loud" music tracks, my system can theoretically hit ~106-108 dB RMS (at the seats) in the (roughly) 20-50 Hz range if full-scale sines are playing in both channels. With more dynamic content like movies and tracks that haven't been crushed, those numbers can be a lot higher. Anyway, the good news is that small sealed rooms have a lot of LF gain, so you'll get plenty of mileage out of small sealed subs. However, your room is not tiny, and that budget won't go very far to get good performance out of cabinets that are so small. To do so requires a lot of motor strength, which usually means expensive magnets and extra shorting rings to keep the upper part of the response clean. One fit for purpose sub would be this one: https://stereointegrity.com/product/hst12-12-subwoofer/, but I don't know that you could afford even one of those on your budget. You also need a lot of amp power to use the excursion capability of a sub like that, and to overcome its very high mass which gives it only ~80 dB sensitivity. A pro sub like the 12BG100 is kind of the opposite situation. The mid bass capability is *much* better (93 dB sensitivity), but it only has maybe 1/3 the excursion, meaning -10 dB max capability in the deep and ULF bass compared to the HST12. The HST12 also has more motor strength, and so will output more deep/ULF frequency using the same power. Is the need for a 14" cube form factor a limitation of width, depth, height, footprint, or all of the above? If you could step up to an 18" driver, you'd have a lot more good options and get a lot more value for your money. I personally think pro-style 21" is a very nice sweet spot. Compared to four HST-12s, four B&C 21DS-115s stuffed into "tiny" 575x575x350mm (ish) cabinets should cost less, have more output at the bottom, and provide *way better* overall efficiency above 30-40 Hz or so. The Lavoce SAN214.50 gives similar performance to the DS115 and may be less expensive. Anyway, it's something to consider. I've seen a few conversations around here wishing for better 12" sub options, for either pro or small room usage. If you have to stick with 12", then the driver you really want probably doesn't exist. :(

You have a K10 which has a "true power" limiter, which I would definitely use. I would configure it to use a fairly long attack time (10 seconds?) and an even longer release time (maybe a minute if possible?). I would also choose a quite conservative power value, like 350 W per driver. The idea is that you shouldn't normally hit this limiter. It's there to protect the equipment against worst case scenarios of long continuous tones at full blast or a DJ pushing the sound into severe clipping. Couldn't these drivers take more like 700W a piece? Maybe. But if one is frequently pushing long-term true power beyond 350W, adding more cabs might be the better solution. The drivers are certainly likely to last longer if they aren't being driven so hard as to heat the cabs up.

One risk of high long-term average power that I don't really see discussed in speaker forums is the fact that Nd magnets lose magnetic strength at a rate that depends on temperature. While they have to get pretty hot (like 350 deg C) to demagnetize rapidly, slower losses can occur with much lower temperatures. I believe details depend on the particular grade of material and the shape, but I believe there's reason to be concern about wear if operating with magnet temperatures above 150 deg C for sustained periods.

I noticed earlier in this thread you discussed making and improving vent block to adjust the tuning frequency. Can you provide any details of the design and how it attaches to the front-end? I did look at the pictures and noticed that you don't appear to use any round-overs at the vent exits. I'm hoping I can find a way to securely block rectangular vents with round-overs. Good luck, ehh?

You mentioned you are considering 12"s. Is that really as large as you can go? How big is the room? How well contained? How low do you want to go? Do you need to monitor multichannel content with LFE? Or just 2 channel content? If you want low, loud and small, be prepared to spend some serious money.

This is my first post to this thread, so let me first offer condolences to the OP. That's a major bummer! As is apparent by the posts here, this subject is very complicated. Something I don't think is emphasized enough in this discussion (at least directly) is the importance of impedance, which varies with frequency. Impedance describes the relationship (both magnitude and phase) between voltage and current in the driver. The amount of power turned into heat in the coil depends on the coincidence of voltage and current. That is, instantaneous power equals instantaneous voltage multiplied by instantaneous current, but to the extent voltage (V) and current (I) are out of phase and don't peak at the same time, the real power consumed (or turned to heat) by the coil will be less than Vrms * Irms. A reason why impedance is so important is that it varies substantially vs. frequency in a way that depends on the particular driver and cabinet design. Frequency dependence means that actual power may be very different when playing two different music or test signals, even if they have the same duty cycle. For example, an efficient vented cabinet may consume very little power throughout its range *except* near its tuning frequency where impedance is very low and power consumption may rocket way up. This means that such a cabinet might do just fine being overloaded with music for many hours until you come across a track that has a strong sustained drone right at the tuning frequency. This is a doubly-bad scenario. It's bad because the coil is taking way more power than it was with other content, and it's bad because (as mentioned earlier here) the tuning frequency is also an excursion minimum which prevents the driver from cooling itself. Now consider that all of the above also applies with AES testing, which is performed with the driver in free air. In fact, the reported "AES power" is not really power at all. It should perhaps be called *nominal power* because it's calculated directly from the *RMS average voltage* used for pink noise test signal by dividing it by the manufacturer specified *nominal impedance*. The pink noise test signal is band-limited to the range 1X to 10X the manufacturer's specified low frequency limit. To estimate the *actual power* consumed by the driver using the AES signal, one can sample the *real part* of impedance curve at *equal octave intervals* (i.e. 1/3rd octave or 1/6th octave, the finer the better the estimate --- because *pink noise has equal energy per octave*) within the test signal bandwidth. Each sample should be divided into (Vrms^2) and the results averaged. This works in principle, but in practice, the *real part* (or "in-phase" part) of the impedance is rarely plotted separately on data sheets. (Perhaps it should be available to plot on Data Bass?) The real part is always less-than-or-equal to what's normally plotted. An alternative method of calculation would be to simulate the driver in a very large sealed enclosure in Hornresp (with proper semi-inductance parameters) and sample and average the "driver power" over the test signal bandwidth. As an example, consider the DS115-4 (4 ohm nominal). The AES power is 1700 W, which simply means that the driver passed the long-term test when operating in free air using a pink noise signal at around 82.5 Vrms average (calculated from Pnom = Vrms_avg^2 / Znom). The specified frequency response lower limit is 30 Hz, so the test signal bandwidth is 30 to 300 Hz. I used Hornresp as described above and sampled at 17 different points. The average I calculated is around 950 W, which is nowhere near the AES power. Keep in mind also that this test signal will cause a fair amount of excursion, which will help push air through the driver vents to cool it, and the free-air mounting allows much more ventilation than is typical inside of a cabinet. Unfortunately, these variables change a lot between drivers and cabinets, so even this number (950W) doesn't tell us much about real world worst-case-scenario power handling. Surely it will be quite a bit lower still. Either way, one must consider that vented and BP6 cabinets (like Skhorn and Skram) will have frequency regions of particularly low impedance, which if hit with a high duty cycle narrow-band signal (as may occur in various EDM), could result in the driver seeing something like 2700W (!!) with the same 82.5Vrms average input used in the AES test. So if one wishes to avoid putting more than 950W average actual power into the driver in a WCS for signal input, one must limit output to a merely 48.7V, which is only 600W nominal power. And as noted above with regard to cooling capability, that 600W nominal may still be too much power.

Thanks for your insightful comments. I don't do work in either mixing or mastering, but my research into audio optimization has me doing a lot of critical listening. I find peaks at 13 kHz to be very irritating (somewhere between piercing and grating), yet its absence can sound very dull indeed. Perhaps that mixer was trying to get the treble to punch through a nasty mess of bass boom they heard on their system? I believe it's often hard not to reach for the wrong control when wanting to fix a perceived imbalance in an audio track. This is a particular ugly example, in part because it is long duration, like ~30 seconds or something? The sad thing is that despite the fact that so many movies are made with budgets that dwarf musical productions, the sound quality isn't necessarily better than what some people can mix and master in their spare bedrooms. Budget seems to have little correlation with overall sound quality in a movie. My perception is that many movie soundtracks are mixed from very high quality recordings but the mixes often suffer from a variety of level and spectral balance problems. Also in general I don't think mastering practices really exist for movies like they do for music. Mastering in cinema is just a technical step, creating digital (and possibly analog) master copies of the end product. Much of the work that's done by music mastering engineers is done by the recording mixer(s) as part of the mixing process. There is no "second opinion" of having a different engineer look at it like there is for music. Likewise, I doubt most cinema mixes get heard on more than one system before they are finalized. I believe a wide assumption in that field is that because all the different cinemas are calibrated to the same standard, the mixes will translate. In contrast, I believe music engineers often test on a few different systems (such as their car), and I think this drastically increases the chance of hearing problems hidden by their main set of monitors.

I think most would probably want to cross the Skram a fair bit lower than the vent resonance at 150 Hz, like maybe 120 Hz or even less.

That's an excellent point, thanks! And now that I think about it, my post may have come off as a bit judgmental about use of plugins without the ability to monitor the result. In practice, I believe this is done all the time and works out OK and sometimes can sound really good. It's probably the case with most popular "unfiltered" movie soundtracks including those that have been subject to high quality custom BEQ to make them "unfiltered". However, embarrassments do happen. They are more likely to happen when processing is applied by the mixer blindly, or deafly as it were.

In brief correspondance with Brian, he confirmed that the actual power output capability of the SP amps is lower for loads well below 2 ohm. He did not clarify exactly how the amp responds to an overload condition, but I'm pretty sure it handles it without blowing up. The question is whether it throttles back suddenly and how much it does so. A lot of amps probably just cut the sound temporarily under a current overload condition, which is not very cool in a live show. My impression is that SP went out of their way to avoid doing that. I'd certainly be curious as to how the amps handle it and how much power they can actually do. FWIW, I light up the "yellow lights" on my SP2-12k from time to time (albeit on ULF-heavy peaks in movies and with impedance no lower than 2.7 ohm), and the limiter is so smooth that no one would even notice. I can't even really hear any distortion when it activates. Though I imagine the difference would be more obvious if I could do a direct A/B comparison with sufficient power.

The ringing is already there, as a consequence of the HPFs (both electronic and acoustic) at/near the vent resonance. It can't be avoided because it's a fundamental property of the cabinet alignment. One would have to EQ it to not roll-off at its lower limit, which obviously doesn't work in practice. If the PEQ you suggest does not precisely cancel a comparable existing dip there, it will also contribute its own resonance. At Q 0.6, this is likely to substantially affect transient sounds, in addition to directing listener perception disproportionately toward the ringing that's already there. While many DSP systems have problems with filters at very low frequencies, I'm not discussing that here. The ringing I speak of arises from the final response shape, not necessarily the processing that was applied to get it to that shape.

A peak at 30 Hz, yes, but not like a bell-shaped PEQ peak. Rather, I'm thinking of a smooth ramp for the combined response, like a straight line with a slope of maybe around -3 dB/octave from the bottom up to 250-500 Hz or so. When using the subs indoors near boundaries, you may see that kind of shape arise naturally in measurements, albeit with a lot of messy finer details due to the effects of interfering reflections, room resonances, etc. As far as how to do this with EQ, I guess it depends on what's available. Multiple shelf filters, spread out across the frequency range, may be a better choice than PEQ filters. Even better might be to use an FIR-based solution to nail down the desired shape precisely but I don't know that these are commercially available for bass usage. Unfortunately PEQs and shelf filters both contribute new resonances to the sound, to the extent that they don't perfectly cancel out existing problems in the response shape. Used skillfully, these filters can be an effective medicine, but they come with a bitter aftertaste.

Wow! What complete and utter rubbish. I'm not going to question the author's reputation as a mixer, which may or may not be well deserved, but most of what's written there is embarrassingly wrong. It's hardly unique to the author. He's just perpetuating myths that are widespread throughout that community. These myths come about because people are trying to relate their subjective experiences with sound to objective principles, without sufficiently understanding the latter. One of the worst issues I see are that he blames "muddy sound" on rooms being too small for the wavelengths. That's only true in a very ironic and roundabout way. It ignores the fact that mud is at least as serious of a problem for frequencies in the 100-500 Hz range, whose wavelengths are also long enough such that the speaker/monitor interacts with nearby large boundaries from walls and in particular the mixing consoles (!!) in widespread use. The picture even illustrates what appear to be a bunch of narrow EQ cuts, which while much less offensive than equivalent boosts, nevertheless are likely degrade the sound quality of the tracks substantially to the extent that they reflect problems in the monitoring system vs. the actual soundtrack. Worst of all is where he suggests using a sub-harmonic synthesis plug-in to "boost [the sub frequencies]" *as an alternative* to installing a subwoofer into the monitoring system. This is completely confusing the critical distinction between *adding more bass to the monitor system* and *adding more bass to the soundtrack*, which I would expect any competent mixer to understand and appreciate. Yet, this confusion also is not unique to this author and seems to be widespread among mixers. And because the mixers running these plugins can't hear what they are actually doing, this is actually a "worst of both worlds" solution. To emphasize the wrongness there's this sentence at the end: And what exactly do plug-ins, used to alter the soundtrack, have to do with whether *your monitoring system* has a subwoofer? Nuff said. Now with that criticism out of the way I want to point out that I don't think the author is incompetent nor is he trying to deceive people intentionally. He is correct that the mid-range is the most important part of the spectrum, even for "bass" instruments and "bass" music genres. He is writing based on real lived experience of how systems with subwoofers sound and what it's like to work with them. The reality is that good sub integration, with or without "auto EQ" tools, is quite difficult and is beyond the skill of many mixers such as himself. For many such people, monitoring with *no* subs may be better than monitoring with *bad* subs. It's hard enough to deal with the issues in the 100-500 Hz range that are common to every studio using "near-field" monitors and/or a large mixing console. Throwing a sub into the mix is likely to only make things worse. I also agree with his implication that soundtracks often have too much sub boost in them. This is very common for movies which (surprise surprise) are mixed under bass deficient conditions, but it's also becoming quite common with music releases. I get the feeling that excessive sub boost on soundtracks may be the major reason for him to write the article, yet his advice is essentially the opposite of what's likely to help. At the same time, I feel sorry for him and many others like him because it's clear to me that he has probably never heard a good small-room bass system in his life, despite his professional pedigree. They just assume that bass cannot be reproduced in small rooms like it is in large rooms or outdoors because they haven't heard it done before. Perhaps if they knew what was actually possible, they would have a quite different opinion.

Smooth response is more important than flat response. In fact, you may prefer a response with a substantial rise toward the bottom. Also, I believe broad scale features are actually a lot more important than features at narrower scales. As @Ricci recommends, I would not try to "fix" the notch from the port resonance at 150 Hz. I'm not sure about notch at 85 Hz, but I don't see it in your earlier measurements. This suggests to me that it could be caused by interference from one or more reflections. If that is the case, then you should leave that notch alone also. The key thing is that you want to EQ actual problems with the sub and not features caused by acoustic interference. Easier said than done. While Hornresp simulations are not perfectly accurate, I went ahead and used the Filter Wizard in Hornresp to look for a good PEQ that might help smooth the response. The PEQ I came up with is: Frequency 93 Hz; Q 2.1; gain -2.2 dB The result simulates quite flat, albeit with a very slight bump in the low 70s Hz. Without knowing how the subs behave in real life, I'd suggest applying the PEQ while listening to music, and varying the gain until it sounds "balanced". A balanced sound will exhibit the positive characteristics of all these frequencies simultaneously, without letting any one frequency area get in the way of the others. Keep in mind that you'll probably need to turn the subs up a bit more after doing this PEQ to keep them matched with the mains, and you'll want to do this before critically evaluating any adjustments. And yes, do critically listen to the results with your ears and body. Measurements are very helpful to understand what's going on, but i see too many people listen with their eyes, obsessing over how pretty their measurements under the misplaced conviction that the measurements are the absolute truth. (For a number of reasons, it's way more complicated than that.) I would not recommend PEQ boost of any kind at 30 Hz. The vent tune will already be contributing to ringing there, and any boost, no matter the bandwidth, will likely accentuate that ringing. If instead of cutting from the top, one wishes to boost the bottom, I'd suggest trying a low-shelf filter if that's available. A low-shelf centered around 75 Hz (looking again at the sims) could be helpful. However, I think the PEQ I suggested above may be a better place to start.

I haven't heard any Danley speakers, but I hope to some day. I expect they sound quite good. Using EQ you can change the sound of the Skram, for better or worse. The Skram response simulated in Hornresp and exhibited in your measurements has a bit of a hump around 95 Hz, which is likely to dominate the character of the sound. EQ down that hump, and the bottom may come alive. Ideally this can be achieved while retaining (and hopefully enhancing) the punch, but finding the ideal EQ ideal may be tricky. If you apply such EQ to the subs only (instead of the whole signal), you may find that a different crossover frequency works better. For example, crossing at 75 Hz reduces the emphasis from the 95 Hz hump (it's still there, but less) compared to a higher crossover frequency, but if the hump is EQed out, the system might sound good crossed a bit higher, like 90-100 Hz.

You should carefully double-check the inputs you used for the semi-inductance including the names and units. Note that the order of parameters published on Data Bass may not match the order you need to input them into Hornresp. If you aren't sure, can you post a screen shot of the screen with the inputs?

Hey @jay michael , any updates? Have you listened to the Danley + Skram combo yet?

Whatever you end up with, it's probably a good idea to compare the sound you get to no EQ and no Audyssey with the sub level set by ear. For a bass boost can you use a Linkwitz Transform filter? If you know the approximate Fb and Qtc of your enclosure, then you can use LT to extend the cut-off frequency a bit lower and maybe lower the effective Qtc. I suspect this will sound a bit better than a high Q shelf at 30 Hz.

I guess I should have mentioned that the "optimal" filter choice must also depend on the capabilities of the sub(s). Some need steeper roll-offs than others to avoid over excursion. This makes a systematic study rather difficult. For example, if the particular sub system sounds best with a roll-off that's lower order or lower Q, but this is not possible for the system without excessive distortion, then perhaps the listener would prefer a roll-off that starts higher. Another variable is that some DSP architectures allow limiting to be applied between EQ processing steps, and it may be possible to roll-off subs less aggressively at the bottom in exchange for more compression (and maybe milder limiter distortion) at higher levels. Another tricky issue with systematic evaluation is the test content. As I already noted, content itself is rarely neutral with respect to bottom-end. It almost always has one or more ringing filters. That 48 dB BW HPF at 20 Hz you speak of is quite steep, but this my not be audible unless you are listening to content that isn't filtered up a lot higher, which most movies and almost all music is. OTOH, if you play content that is also steeply filtered at 20 Hz (which I encounter often for movies and occasionally for music), then this is likely to *exaggerate* the ringing of your filters. To be precise, a lot of movies use aggressive shelving rather than a HPF at the bottom. This way the ULF content is still in the track but at a lower level and can be perceived if it crosses perceptual thresholds. However, aggressive shelves still ring a lot, and the ringing characteristic tends to contribute substantially to the perception of the bass sounds, whatever they are at the expense of other potentially more interesting details. In any case, there are good and bad reasons for applying filters of various kinds to a soundtrack. In many cases, filters do more good than harm. It's like that with compression and limiting too. Tools are neutral. How they are wielded matters a lot. A wrong reason to filter is to try to remove content that the mixer "cannot monitor". I don't know any mixers who cut their mixes off at 12 kHz or 15 kHz because of the capabilities of their hearing or their speakers. Unfortunately, few mixers can monitor bass with enough quality and extension to hear what these filters are doing.

That's a good description. And on bandwidth limited systems, it may not necessarily be a bad thing to sharpen the cut-off a bit, as long as the ringing doesn't get too excessive. I think it'd be worthwhile to try to do a systematic study of listener preference on this point. It's something I intend to do as I migrate toward testing my low frequency optimization methods in other rooms and on other systems. Not everyone can have 5 Hz extension like I do. Unfortunately this is greatly complicated by the fact that HPFs get applied to audio content also. These are often high Q and can contribute a lot of ringing. This seems to be especially popular with movies these days. It's understandable. Everyone wants "more deep bass" from cinema subs that only play to 30 Hz. Unfortunately though, 30 Hz sub vary quite a bit wrt HPFs used, so the results will be unpredictable and not translate well. Ideally, soundtracks should avoid using HPFs above like 10 Hz. This way, playback systems can be optimized, using a high Q HPF if desired, to get the best compromise of "deep bass sound" vs. "ringing". Alas, this is very wishful thinking on my part.

The first link doesn't work for me. It's great to hear you're able to both upgrade the system and reduce its size and weight. Double win!

Great explanation. Thanks! I do want to note that when doing simulations of BP6 designs, I noticed that the roll-off tends to start a bit higher up, relative to the Helmholtz tuning frequency, than a similar plain vented design. Presumably, the horn is still doing a bit of unloading there, contributing to earlier roll-off. For the data you posted, Hornresp shows 29.5 Hz for the tune (I assume based on Helmholtz). The impedance minimum comes in slightly lower in the sim, around 29 Hz. OTOH compared to the plain vented, the vent in a BP6 doesn't seem to unload as rapidly below tune, so they may have a bit more "usable output" down there. In any case, the sim shows response that's almost around -2.5 dB at 29.5 Hz. The same point in the measurement looks to be around 27 Hz. That's really not too far off. If I understand you right here, shouldn't you include this twice, at least for ground plane because the ground has the same effect as the rear wall in terms of limiting expansion at the inlet/outlet?