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  2. Will post some measurements on Sunday cause I’m out of town until than.Thanks a lot for your time and help, I really appreciate it;I really had invested a lot of time and money in my theater, learned a lot and have so much more to learn and I don’t want to give up before the bass ,that is so important to be done right , is not so how I imagine to be to my ears. Very through and very interesting topic what you said about FR in room vs what you hear, but I think it depends from one room to another; A very nice guy, AVS member, that helped me to understand REW , crate filters in rew for my inukes that I applied and the frequency response was perfectly flat from port tuning to 120Hz, but”paradoxically “ how you say, it sound very, very bad in room, the abundance of ULF was so present, I couldn’t turn the volume more than-10db on Wotw, pod scene for example, I think the house was falling apart.Since I sealed provisional the subs and applied just a little EQ-3 points, it sounds better but thhis loud bass from the ported, that was sometimes fun too, disappeared. Although I put a two feet thick absorber on the whole back wall, I have this problems in the ULF, that hope to solve.
  3. Can you post your EQ settings? Also, can you post any measurement data you have (preferably before EQ is applied)? Data for measurements taken at multiple locations is even better. In my experience "frequency response" measured in-room, especially at only a single location, is not a reliable direct indicator of subjective sound quality. It's not that frequency response isn't important. In fact, FR matters a lot! The problem is that the in-room FR at the seat is not really the FR that you hear. EQ that makes in-room FR appear "flatter" can actually degrade sound quality. Measurements at multiple locations can help reveal and distinguish modal room resonances from interference effects. Modal resonances are audible and likely to benefit from EQ treatment. Interference effects are less audible and much more difficult to treat using EQ. Other than that, I suggest experimenting only with broad (low-Q) changes, mostly cuts rather than boosts. Often times, cutting bass in the right place (i.e. where cut is actually needed) can paradoxically make it seem louder and more intense. Your 5-10 dB peaking filter at 60 Hz is likely to just emphasize whatever FR problems exists in that region, which may make things better or worse. Applying a broad dip centered at the "inductance hump" for your sub may be helpful if you have any outdoor measurement data to refer to, but this hump is hard to see in an in-room measurement. At the end of the day, the sound quality of the result is what's important, not how the FR picture looks. Always listen to the result of any EQ you apply and consider playing with the gain to see if you can find a balance point. ==== Sealed subs don't necessarily need a HPF like vented subs do. Below their tuning frequency, vented subs essentially behave like leaky boxes and excursion tends to rise very rapidly for the driver, even though it's not making much useful sound. Driver motion in sealed subs is constrained by the air cushion. As for what you get with pro-style drivers vs. UM18s, the answer is complicated. The differences are as follows: (1) Headroom/output capability: This is important if you are running the system hard and are constrained by output capability. The pro drivers will have quite a bit more mid-bass output than the UM18s. In sealed cabinets, they will have less deep bass and ULF output. With that said, you seem to have a lot of output capability in there, so unless you're really hardcore, you're probably not constrained by output capability. (2) Innate/baseline frequency response differences: This affects the sound without EQ and also with EQ to the extent that you don't completely compensate for the difference in response shapes. Pro style drivers will typically give an FR with more mid-bass and less deep bass. A sealed sub will give up even more deep bass, but may allow ULF output to lower frequencies than your current vented subs are capable of. Pro style subs often (but not always) have better management of inductance, which reduces "humping" that often shows up in the 40-60 Hz range and can overemphasize this region at the expense of everything else. Realize that, in principle, innate frequency differences can be compensated for using EQ, but in practice this can be difficult. So even when using EQ, it might be advantageous to start with a baseline that has more mid-bass emphasis and maybe less severe inductance hump centered at a higher frequency. (3) Non-linear effects, in terms of distortion and compression: Different types of drivers exhibit different non-linear behaviors. The is more of a driver-by-driver thing, but inductance non-linearity may be quite important here. So drivers that manage inductance better are also less likely to exhibit inductance non-linearity. Note that non-linear compression is not necessarily the same at all frequencies, so non-linear effects can involve frequency response changing with driver level. For example, the "inductance hump" might be diminished at high drive levels relative to low drive levels. EQ (which acts linearly) cannot compensate for these non-linear effects. So to the extent these things matter (and unfortunately, there's no simple answer here), drivers with lower distortion and better managed inductance are likely to sound better. ==== So probably the first thing you want to figure out is whether you are constrained by output capability. With 6 ported 18s in there, I rather doubt it, but if you are in fact constrained in the mid-bass and have extra headroom in the deep bass and ULF, then moving to more pro-style subs is likely to help. If you have enough headroom, then you can try experimenting with EQ (or with turning it off! ) to see if you can improve things that way. If not, then a move to pro-style subs could still help for reasons (2) and/or (3) above. FWIW, I do lean toward recommending pro-style subs in general because small listening rooms tend to provide substantial boundary gain to compensate for the relative deficiency of bottom-end output with such subs and because they almost always handle the higher mid-bass frequencies better. Pro-style subs have also become quite popular in the DIY community, and a lot of people swear that they sound better or "slam" harder than traditional home-theater style subs. This is likely due to some combination of (2) and (3) above, and I mostly lean toward (2). Of course, you've already invested a lot of time and money in your existing setup, so IMO it's worth trying to get the best out of it before changing it. Good luck with whatever you do!
  4. I use a miniDsp and after EQ,the response is pretty flat from 10 to 120Hz;Kyle mentioned a few posts above that I need to turn off the HPF after sealing the ports, I’m a little confused now, do sealed subs need an HPF? I will try to set a 5 to 10db peak filter in the minidsp in the 60Hz region(??) to see if the um-18s can give me what I’m looking for.How do you think would the Lavoces perform compared to the UMs if I decided to make the switch ( or any other pro driver with equal specs) in sealed cabs? Do I loose completely the ULF and win better midbass?
  5. I don't use a HPF with sealed subs either in testing or my systems. I'm careful during testing to not push the level too high and in my systems I try not to have enough amp on tap to bottom the drivers with the box size used.
  6. I'd echo SME here. The Othorn is still a very good sounding design but I'd consider the Skram instead. It's smaller, lighter, has flexible tuning options and is simpler to build. Output and SQ should be the same if not a bit better due to the more extended top end. Issues don't start till higher in frequency that the Othorn.
  7. That's what I tell people as well. Make the sealed sub as big as you can, until you can reach Xmax at Pmax. Usually the cab will be smaller than what it takes to do so, so it doesn't make sense bothering with a HPF, unless you wanna reduce the load on the sub, like when you HPF it at 15Hz because your sealed 12" sub won't have meaningful output below that anyways. Was it a 5400?
  8. You may want one to remove content in the low region the sub doesn't have useful output in to increase usable output in the region it does. EG. if your sub is -3dB @ 30Hz and you play back the TELAC 1812 recording with 11Hz fundamental cannon fire you would be better off not just going to maximum excursion on the 11Hz and instead reproducing the harmonics*. Only applies if your hitting excursion limits though, if the sub stays in the linear region I don't see an advantage to HPF. *As the harmonics will be distorted by the sub been pushed out of the linear region
  9. Hi @Bobby. Please note my earlier post about EQ and note whether or not you use it, and how you use it if you do. Application of EQ can both make or break the sound of a system. If you use any kind of EQ (auto or manual), try shutting it off (except for essential things like protective high pass filters). If you don't use EQ, you might try adding a gentle shelf filter in the iNukes to attenuate the deep bass and ULF relative to the mid-bass. If you can measure your response in the room using e.g. REW, you might try measuring in a few different spots to see where you might have room resonances that might also benefit from EQ attenuation. All of this can be done before spending lots of time and money replacing equipment, and it might give you insight into what changes will actually benefit you the most.
  10. I don't add one on my sealed HT sub, then again, I made a mistake once with the amp maxed out and now I have a voice coil ring dent on my metal dish cone Still works fine tho!
  11. Thanks, that's been my general thinking as well. While it does make sense to choose to use one if circumstances dictate, this recommendation was for any diy sealed sub and was wondering how many do it or for what reasons they did it for a specific build. Sure isn't much available on the subject even with a general search, either so it doesn't seem to be a popular thought....
  12. I sealed the ports,like you suggested, and the overall result is better than before; the abundance of room shaking Low frequency and TR in the couch is diminished,but still there.Next step is to modify the cabs to have the same internal volume on al six,sealed and see the result; maybe the ULF region and lot of TR is not much of my taste, do you think that six or eight high efficiency 18”, like the Lavoce saf 184.03 in sealed cabs can give me the best of both worlds, enough deep bass and much better punch in the midbass? Think that the small concrete walls room has a lot of gain that made the ported um18 to overkill.I read that the Lavoce is almost as good as the b&c at a lower price.
  13. Most of the time you probably don't need one, whereas most vented and open-back horn type subs do need one. When designing a sealed system, it's always a good idea to look at excursion vs. the max power (based on amp choice) vs. frequency, which can be viewed in most design programs. Keep in mind that if you think you need a high pass filter, you probably just need to choose a smaller amp (or set a voltage limiter in the amp) or make the box smaller to move the cut-off frequency higher. There are surely exceptions here, especially if you're working with high motor force / low Qtc alignments where excursion can continue to increase for quite a ways as you go below the resonance.
  14. Didn't see anything on this. Came up in discussion on audioholics where someone is suggesting that is something to do to prevent over excursion particularly in combination with eq/boost. Anyone use a protective high pass filter for their sealed subs? @Ricci do you use such when testing?
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  16. Interesting. I'll assume the system updates the model parameters at least as frequently as the graphics do. I imagine it does a good job of correcting thermal-related changes, and the ability to correct for environmental effects on soft parts could be quite useful. These are things that can be approximated to be linear for relatively short intervals of time. Now then, I see something important that I missed. It is mentioned that a system that relies on actual K(x), BL(i,x), Le(i,x), etc. data taken from a "Klippel analyzer" device can theoretically model and cancel out the anticipated distortion from a given input signal. This capability is independent of the ability to adapt to changes of these parameters over longer periods of time. It looks like the purpose of the discussion of "effective" parameter values (averaged with respect to stroke or some short time interval) is to argue that a small signal model with periodically adjusted parameters works OK for identifying and correcting for the longer-term changes described above. So really, there are two different kinds of corrections being made here. AFAICT, the paper is not clear on how these two corrections are applied together, but I would assume they are applied relatively independently such as by multiplying. For example, Kms changes might be modeled as: Kms_now(x) = alpha_kms * Kms(x) where Kms_now(x) is used in the distortion cancellation model, Kms(x) is the original Klippel-measured stiffness, and alpha_kms is a multiplicative factor that descries changes observed by the live impedance monitoring. Something notably absent from consideration here are so-called complex inductance models, which I expect to be important for cancellation of distortion in large, heavy voice coil subs. These papers would generally recommend that one keep Mms as low as possible to maximize efficiency and to compensate for sensitivity problems around F0 by using lower Re (and a suitable high-current-stable amplifier). However, one reason to have a higher Mms in a deep bass sub (aside from the fact that bigger coils also tend to handle more power), is that it acts as a kind of acoustic high-shelf filter that reduces higher frequency distortion harmonics. (Note that high Le does not have this effect because it acts before the main source(s) of distortion are introduced). Higher Mms may also improve system behavior in multiple driver systems or in highly loaded scenarios. IIRC, this was a potentially important factor in @Ricci's M.A.U.L. design. So back to Le. Complex Le is well known to be essential for modeling small signal behavior of large subs, so I think having this be part of a distortion cancellation model is important. ==== Anyway, this work is definitely intriguing. Several years ago, I contemplated experimenting with a feed-forward approach to reduce distortion in my own subwoofers, but I haven't looked into it yet. Instead, I've been focusing intently on linear aspects of response for many years now. Linear response seems to be absolutely critical to sound quality, and even seemingly minute flaws in linear response can profoundly alter the sound. I believe the reason for this is, most ironically, because essentially all real-world content has a lot of harmonic distortion, some of which is naturally occurring but much of which is intentionally synthesized, often very late in the production chain in order to enhance the sound quality. This added "distortion" relates to the original signal in a very regular way, one which listeners seem to adapt to very readily (given that this is also a property of most natural sound sources). However, linear response flaws can disrupt that regularity making it much harder for the listener's brain to correlate the spectral content. This leads to what I would term "perceptual spectral leakage" in which some spectral content becomes disassociated from the rest of the sound. For example, perceptual spectral leakage in the low frequencies can cause muddiness. Perceptual spectral leakage in the high frequencies can cause harshness. IMO, both of these problems are widespread under typical reproduced sound conditions. With that said, non-linearity surely matters at some point. On an indoor system with ample headroom as I'm using to develop my optimization technology, I don't worry much about non-linearity, but I reckon it will be a lot more important for systems being run to their limits, including in most live sound applications. I think I'm less worried harmonic distortion, especially low order HD which is much less likely to sound offensive, and more worried about effective frequency response changes. These papers describe an approach which appears promising for long time-scale changes to sensitivity and frequency response such as from thermal and environmental effects. The papers also describe distortion cancellation, albeit with unclear performance for higher order distortion products which may be more offensive. However, the papers don't appear to discuss short-term frequency response shifts such as when a kick drum hits at 16 dB above the average level. Hypothetically on some system, my optimization methods may yield superb results for everything but those kick drum hits, which would be quite tragic. At this point, I have no idea how important these problems are because my system for R&D has ample headroom for all but the biggest (under 20 Hz) movie effects. We'll see, but it won't surprise me at all if I find that my optimization makes it a lot easier to hear and compare the underlying non-linearities of sub systems.
  17. I think these area good points I also had some skepticism about the statistical basis for saying that the the small signal values can be used. High crest factor signal is interesting and only reporting 2nd and 3rd harmonic distortion, this suggests some gaming of the results. I think the practical implementation uses voltage and current sensors to inform the model along with reference data obtained from the kipple system of the transducer. There are a few demo videos floating around like this one where parameter variation over time is shown:
  18. Interesting discussion. I'm definitely a fan of the idea of designing transducers and electronics to work together. However I'm skeptical that the described approach will work as well in practice as it does in theory. Even in theory, I don't really agree with the argument given in "5.2 Large Signal Modeling" in the first paper that it is adequate to approximate instantaneous large signal parameters e.g. (Bl(x,i), K(x), Le(x,i)) with their averages with respect to stroke and current. To me this is basically a hand wave. Among other things, it ignores the likelihood that the peaks (from instrument attacks) are likely most relevant to perception of the sound. The practical example given in the second paper isn't all that gratifying to me either. Distortion performance is reported in terms of "averages" with little attention given to distinguishing higher order vs. lower order contributions or the potential perceptual consequences of higher distortion on the peaks. Improvements to 2nd and 3rd order distortion are impressive on paper, but those are likely to be the easiest components to reduce using DSP. Also, the suggested 16 dB crest factor seems way too big for "common audio signals", and if the content really is that dynamic, I'm not sure this whole study would have much relevance. That 16 dB crest means average power is 1/40th of peak. So if we're delivering 4000W at peak, average power will be a mere 100W. A more realistic crest factor is 9 dB, and that means average power is more like 1/8th of peak, or 500W for my example---much more significant! I'll have to dig into the references some time to learn more about the KCS system used here. The first paper suggests that the required DSP can be done in a feed-forward configuration "in theory", but in practice, some kind of "adaptive" control system is "required to cope with unavoidable production variances, aging of the suspension, changing climate conditions and other external influences" as well as for coping with "undesired effects generated by a voice coil offset". OK. The first bucket of things (suspension, climate conditions, etc.) are "slow" changes for which compensation need not necessarily be adapted in real-time. However unless I misunderstand, voice coil offset is an issue that likely requires reacting quickly. Furthermore, if we actually care about distortion levels at the peaks of current and stroke, we probably need a very fast acting *feedback* control system. Is KCS a feedback system? I have no idea, but the lack of detailed discussion on this point in the papers makes it harder for me to believe that this approach is as easy to achieve as the paper suggests it is. Also feedback control systems have a major limitation due to lag in the control loop. This is a problem that affects high frequencies more than low frequencies. So while such a system may do a great job of reducing 2nd and 3rd harmonic distortion, it may be too slow to address higher order distortion problems. Likewise, if high frequency content occurs simultaneous to high excursion from low frequency content, those high frequencies may not be rendered cleanly even with the feedback control. So anyway while I like this idea in principle, I want to see more real world data and a more granular analysis of the resulting distortion artifacts.
  19. I have been reading these papers by Kipple: http://www.klippel.de/fileadmin/klippel/Files/Know_How/Literature/Papers/Green Speaker Design Part 1.pdf http://www.klippel.de/fileadmin/klippel/Files/Know_How/Literature/Papers/Green Speaker Design Part 2.pdf In the second paper they improve the efficiency of an overhung driver by reducing the coil height to the top plate thickness and then compensate for the resulting non linear behavior. The benefit of this in their example is a 39% in voice coil temperature for a given output compared to an overhung driver (~3dB output gain in the non excursion limited region for the same driver). Considering the IPAL driver already is designed to be a transducer with dedicated amplifier and it has a huge coil overhang it would seem that with just shortening the coil a bit and a new version of the IPAL mod that implements the kipple DSP work they could increase the efficiency/output substantially (IPAL2?). There would also seem to be benefits form changing reducing kms but I understand that this is required to have small clearances in the voice coil gap due to rocking modes.
  20. Yeah - for clarification, it was very much a planned burn, and was actually done a bit earlier this year. One of several cabinets that were burned that day. Definitely does not feel right, but I only have so much room to work in. Got rid of a couple of prototype mains, a few unfinished projects, an old car sub cabinet, and the MicroWrecker. It is easier and cheaper to burn them than to dispose of them, though I have hauled a bunch of prototype cabinets down to the local "recycling center" to be used as fuel for the boilers at the neighboring paper mill, I only got charged $20/truckload for those. I've also cut cabinets up smaller and run them through the woodstove in the winter. Wildfires are something else this year. My heart goes out to those affected. We had a small one that was too close for comfort, thankfully the crews snuffed it out before it spread. Fortunately, all I am dealing with is some nasty smoky air.
  21. Muchísimas gracias por tantos consejos. Me han gustado mucho los drivers recomendados. Iré agregando novedades en cuanto las tenga.
  22. Thanks for posting the comparison! About your EQ settings: You reported center frequency (in Hz) and gain (in dB) for each, but there is usually a third parameter called "Q" or "BW" (bandwidth). Is it missing in Audio Architect? Or did you not notice it there? This third parameter affects how broad or narrow to make the EQ peak or dip. I do think you're applying a lot more EQ than you actually need, but before we get into that much, there are other things worth looking at. For my first question: what crossover frequency do you have set in the AVR? It'd be helpful to know where that's happening. Second: would you consider turning the sub around so that the driver is firing into the corner? This is likely to improve things in the upper part (e.g. 60 Hz and up) by helping the sub to better couple Let's just start there, and see if that improves your response before you try EQ.
  23. When doing roomEQ I try to avoid positive EQ points of a Q of higher than 2 or 3, because it starts sounding bad quickly. In my studio I have a point at 31Hz which goes to -35db (almost round room), adjustments of this magnitude are normal. Where are you crossing the sub to the mains? The way the graph looks, you might have the sub on wrong polarity crossing at 60Hz, but it might also be a room issue as well. Try to flip the phase of the sub and see what the new measurement looks like.
  24. The EQ settings I'm currently using in the Crown amp DSP are as follows: 24Hz +9dB 27Hz -9dB 40Hz -11dB 45Hz +4dB 58Hz +15dB 73Hz -10dB 88Hz +11dB 99Hz -3dB It seems to me some of these adjustments are rather large, but what do I know? I made smaller adjustments at first & found myself making larger ones to get results closer to what I thought I wanted. I was attempting to moderate some of the extreme peaks & valleys of my "No EQ" measurements. I would appreciate any advice concerning these EQ settings.
  25. I'm back after working a few hours with REW & the DSP in my Crown amp. I think I was able to generate more accurate measurements. Here is the graph of the performance of the sub without any adjustments made through the amp DSP: and the waterfall: The Crown Cdi1000 has a few DSP functions - delay, limiter, & EQ. I'm allowing my AVR set the delay & I haven't used the limiter. In the DSP section you also choose Stereo/Bridge-Mono. I've set it to Bridge-Mono which produces 1100W @4 ohms. The EQ function allows for 8 different adjustments at whatever frequencies you select. You can make .5 db adjustments up to +/-15 db. The Harman-Kardon software to control the amp functions is called Audio Architect & it only has a Windows version. So, after I took some measurements in REW on my MacBook, I took screen shots of my graphs, transferred them to my iPhone, rebooted into Windows to open Audio Architect, made 8 adjustments (best guess), rebooted back into Apple, opened REW & took new measurements to see how my adjustments effected the performance of the sub. Rinse & repeat. Maybe I should've used the Windows version of REW... Anyway, after multiple attempts to adjust the DSP here is the graph that was produced: and the waterfall: This is the comparison of the 2 graphs: I just realized I don't have the settings I used to produce the graph in blue. I'm going to post this info, so I don't lose it when I reboot into Windows. I'll come back with the EQ settings
  26. Burning subwoofer cabinets; been there, done that. Feels bad man.
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