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SME

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Everything posted by SME

  1. Is the room drawn to scale? It looks like a small room, and unfortunately it looks like your seats will fall right in the lengthwise null. I didn't mean to say that modal/standing-wave resonances aren't important. They usually do come into play in small rooms. It's just that other considerations are also important, and that study largely ignored those things. How response affects how the bass "sounds" is anything but simple. Putting a sub in a corner without other sub(s) in opposing corners to "cancel out" the standing waves does in fact increase the coupling of the sub to the room's standing wave resonances compared to placements away from those corners. The advantage, IMO, is that such resonances can be attenuated using EQ. Placement away from walls tends to cause suck-outs that aren't easily repaired using EQ, and these response issues may be harder to discern from measurements as well. Either way, EQing a system for the best sound is much easier said than done. It's hard to say why you were not successful applying DSP to "fix" problems heard when a sub is placed in a corner. My first suggestion (1) here is to not obsess too much over what your in-room "frequency response" shows. It's not "wrong", it's just not all that consistent with what you actually hear. Instead it offers hints. Second (2), always look at measurements from multiple locations. Features that appear in several locations within the room are more likely to be audible. Spatial averaging can help somewhat with selecting for these, but rule (1) still applies. Third (3), use EQ judiciously. You're trying to EQ out problematic features, after you've identified them, not make a curve look prettier. The data you get from measuring is part of a map, not the territory. Fourth (4), listen to the result and use your ears to make the final judgment. A prettier curve is no guarantee of better sound. So for example, you might look at your measurements and notice a narrow peak at the same frequency that appears at most locations. Then you might choose a Q or bandwidth for your filter, based on the shape that appears in a spatial average. You might further select a gain that's something like half of the spatially averaged peak, listen to evaluate the result, and then iterate on the gain until the character of the resonance can no longer be heard but not so much that the sound becomes lifeless in that frequency area. Apart from the obvious resonances, you can experiment with broad shape adjustments to improve octave-to-octave balance. This might be best done by ear, unless you are correcting some aspect of the sub that you know about from outdoor measurements or something. For example, you can add a Linkwitz transform to alter a sealed sub's roll-off/Qts to different values, albeit with very substantial effects on *power* and *excursion* requirements. If you know roughly where your subs exhibit an "inductance hump", then a broad EQ dip might be helpful here. To the extent that you are able to use EQ to reduce or eliminate audible problems, you can make significant improvements to your sound, but you will likely find that other problems still linger. The consequences of those problems are varied and complicated. Your changes might lead to much improved clarity, but you might lose a lot of "slam". Or maybe you get more impact but a "colder, harder" sound. Almost every good sound system gets hand-tweaked at the end of the process, so don't hesitate to experiment, especially with broader scale EQ adjustments. I think you are confusing *gain* with *power capability*. The volume setting on your pre-amp says nothing about how much headroom you have in your amps. It's possible that your "really loud symphonies" are actually clipping and distorting on some of the bigger hits. It can be hard to tell unless you are able to A/B the difference unless you're really overloading things badly. With that said, 83 dB is very insensitive. For a "reference" home theater experience, where you are sitting ~10 feet away, you'll probably want speakers closer to 93 dB sensitive. Mind you, a lot depends on details --- are we talking 2.83V or 1W? What do the speaker's impedance and FR look like? IMO, speaker sensitivity (assuming use of subs) is most important in the 120-700ish Hz range where you're most likely to see big suckouts due to baffle and boundary problems, especially where they coincide. Either way, I don't think 83 dB will not cut it unless you plan to listen at lower volume. The other thing you need to keep in mind is that each +3 dB "costs you" twice as much power. So while 700W sounds impressive, it's only 8.5 dB higher than 100W. You could easily see that by putting the sub in a corner vs. mid-wall. Or put another way, moving the sub to gain 6 dB would be similar to *upgrading* your 700W amp to 2800W, *and* you'd have less distortion. Do you have any way to take measurements before you design the room? That would help a lot. Number 3 sounds sketchy. They are essentially IB subs if their back-ends fire into open space that is isolated from the room. However, I have no idea what your giant MDF contraptions are going to do mechanically, and you could end up losing a lot of efficiency among other serious problems that way. It's definitely best for the sub drivers to be mounted rigidly. I lean towards your option 2.,putting the subs in the corners under the bass traps. If you are sitting more toward the back of the room, then the subs will be closer to you than the front wall, so hopefully that lengthwise null with be a fair ways in front of you. The opposing corners placement will cancel out the major width-wise standing wave. Is there any reason why both (1) and (2) only involve using 2 subs? Why not use all 4? Why not do (2) and also put two subs at the front of the hall, at the midpoint and on the ceiling if need be.
  2. Good question! Originally, yes. Although first I should clarify that the shape of the spatially averaged response depends quite a bit on what measurements you include in the average. IIRC, I was flat in the MLP where there was a kind of mid-bass "power ally" effect, but had rise toward the bottom in the outer seats. I'm pretty sure I went back and made broad-shape tweaks to get the sound to "my liking". And by "flat", I only meant for the sub anyway. I totally agree that flat in-room FR will almost always sound too thin and bright. What you're doing for your room is essentially what Harman et.al. recommend these days, and seems to be pretty standard for "state-of-the-art" DIY system as well. These days, I believe Harman regards their latest target curve to be "secret sauce"---essentially proprietary. That's just bizarre to me. I mean, it's probably only subtly different from previous target curves they've recommended which are publicly known. But more importantly, a target curve is, at best, a fuzzy indication of where things should end up. The X-curve in cinema is essentially the same (e.g. +/- 3 dB tolerances, which are huge for SQ purposes, and pretty much gave the projectionist the lee-way to set the EQ to whatever sounds best for him/her, until we replaced them with robots that don't "hear" like humans do). The same system will "measure" very different, even in the same room, depending on where choose to measure. Does the average include more or fewer off-axis seats? Are the extra rows included? How close are the seats to the speakers, relative to the room size? Those things and have a huge impact on where your broad response shapes will fall, so the target curve doesn't end up being useful IMO. Another thing is that the content itself is not all roughly +/- 2 dB or something, broadly. Instead, there are huge *differences* in spectral balance, not just between programs, but between tracks and/or segment of the same program. In movies for example, a popular technique seems to be "mixing big", which among other things seems to using a huge "smile" EQ curve to up the impact of "big" on-screen events. These differences largely overwhelm target curve differences. What matters far more is that trying to fit in-room response (even with spatial averaging) to a smooth curve doesn't really lead to optimal results. It might clean up some of the most obvious modal/room resonances, but it tends to make a mess of the innate/anechoic FR of the source and may actually make transients sound worse than without the EQ processing. I don't really know, but I have often experienced more satisfying bass from speakers without EQ than with some kind of "room correction".
  3. Sure! But your picture is missing a whole lot of *braces*. Other things to consider: material use is inefficient vs. a more square-like shape. And, the enclosure will develop internal standing wave resonances that are particularly low. You'll definitely want more depth of stuffing/fill at each end. Also, see my replies to your other thread about preferring corner placement for higher efficiency among other things.
  4. I also want to add that the article and sound people in general take "frequency response" measurements captured in a room way too literally. The study seeks to minimize "mean spatial variation" while ignoring the crucial question of whether that should even be the objective. My experience suggests otherwise. The first problem is that bass perception isn't merely a function of hearing through the ears. Bone conduction and mechano-reception (via nerves throughout your skin and body) likely play major roles too, and they are able to pick up signals from a wider variety of locations and via different transmission mechanisms. Second, the brain possesses highly advanced cognitive processing that is well adapted to listening in small rooms and inferring the nature of the sound source independently from the "acoustical context". Floyd Toole and Harman have provided ample evidence that, at least for mid and high frequencies, the listener is capable of "hearing through the acoustics" to a great extent in order to ascertain the original sound source, i.e. the speaker or sub. They've shown that anechoic chamber measurements, on and off axis, correlate better with blinded listener preference than an in-room measurement (taken "literally"). I believe this remains true for bass, except that as you go lower, the speaker drivers interact with more than just the cabinet. They begin to interact with nearby boundaries as well. These boundaries effectively become part of the speaker, and their influence alters the sound of the speaker, just as different cabinet and baffle shapes do. Go low enough, and the speaker/sub will even interact with the modal /standing wave resonances in the room. Nevertheless, the Harman people seem to ignore these observations and instead argue that optimizing in-room frequency response is the best course for bass, despite it being sub-optimal for mids and highs. They believe that essentially all the usual rules "go out the window" below the "Schroeder transition frequency", and that below this point, standing wave behavior dominates the room response. I disagree with them on all these counts. First, Schroeder transition frequency is a theoretical construct that assumes large rooms with diffuse reverberant fields, not small residential listening rooms. They describe this transition as typically happening around 500 Hz, probably because that's the rough point below which in-room frequency response measurements start to look a lot "messier" with peaks and dips. In reality, this is simply a consequence of lower directivity, meaning more and stronger reflections are contributing to the measured response than for higher frequencies. Second, I don't believe most rooms exhibit standing waves until much lower in frequency, and that behavior may not even predominate for subwoofer frequencies. unless you are in a room with stiff walls. Third, the presence of standing waves does not necessarily preclude the listener's ability to "hear-through" the local acoustic effects. Where standing waves are involved, a null in SPL response coincides with a peak in particle velocity level (PVL) response. It's possible that listeners may be able to perceive aspects of PVL independently from SPL, perhaps through indirect mechanisms like hair or clothing movement. And fourth, I've tried optimizing for minimum seat-to-seat variation and then flattest possible response in my own room using practically unlimited custom DSP capability. The result looked beautiful in the REW plots, but the sound quality was far from ideal. This was most evident when I auditioned one of Harman's own speakers, the (Revel Salon 2), playing (in another room) with no EQ whatsoever. Its musical bass quality blew mine out of the water. For my purposes, this essentially disproved the claim that flat in-room frequency response is ideal, and it inspired me to develop a novel and more clever approach to assess the sound of the speaker independent of the localized acoustics. My current method seems to work just as well at 25 Hz as it does at (say) 1500 Hz, and so I'm not aware of any sort of transition frequency where the rules of perception change. The only thing that changes is how *complete* anechoic/ground plane measurements are for describing sound quality, where for bass those measurements are still relevant but room effects become at least as important. Anyway, sorry I can't give simple answers, but that's the nature of the subject! I'd suggest you blame your brain, but actually if it weren't for our brains conspiring to make everything sound better than it really does, none of us would probably have this hobby.
  5. @kipman725: I'm pretty sure he's not talking about a DBA. Hi @arcsabre, I think I know what paper you are referring to. It's also covered here: https://www.audioholics.com/room-acoustics/optimum-locations-for-subwoofers-in-rectangular-rooms The study is a theoretical investigation of standing wave room resonance behavior, and the opposing mid-wall placement recommendation arises as the optimal placement for two subs (run together as one, with no differential delay in the signal going to each) that minimizes seat-to-seat variation of frequency response in the room interior I want to highlight a number of problems with taking that conclusion from the study and running with it. First, the study is theoretical and does not consider absorption and losses in the walls, floor, and ceiling. Unless you're inside a bunker, those losses are not trivial . The lack of symmetry due to different kinds of construction on different walls is one thing. If your walls aren't particularly rigid, you may not have much of a problem with standing waves anyway. That's because standing waves require multiple successive reflections to build up. Second, mid-wall placements typically have poor output efficiency and suffer more from boundary interference effects from the adjacent walls. My own experience is that these problems can be severe. Depending on your room, such interference problems may predominate over standing waves for much of the response. In general, subs are most efficient in a corner because the corner allows more pressure to build near the cone, which improves energy transfer. This efficiency boost is very important because it reduces the amount of equipment and cost required to reach a particular output goal. Also because the sound energy spreads through a smaller section of the unit sphere ("pi/8 space" vs "pi/4 space", the directivity is effectively higher, meaning you get high SPL for the same power output too. Even the paper notes that mid-wall placements tend to have poor output vs. corner placements, but they are focused on minimizing seat-to-seat variation regardless of cost or compromise. Third, the study treats the subwoofer system completely independently of the mains speakers. The trouble here is that the subs do eventually cross over with the mains, and that crossover region is quite broad even with fairly steep high order. For an 80 Hz crossover 4th order LR, it might be 60-110 Hz, which is nearly an octave, and covers frequencies that are crucial for enjoying all bass content. IMO, a good crossover with the mains is likely to be *more important* than avoiding standing waves at lower frequencies, especially if your focus is music. My suggestion if you are building two subs is to put one in each front corner of the room. This puts them relatively close to the mains, so they will be more likely to combine with the front channels at multiple room locations. These placements will also cancel out some of the standing waves along the width of the room, if that's a problem for you. It will not cancel out standing waves across the length of the room. This is most likely to be a problem for in the very middle of the room (length-wise) or a little further back, and if it's possible to move those seats even a couple feet, it might make a big difference. If you can build 4 subs, then you can put one in each corner. Though this might make integrating with the front mains channels a bit trickier---or not. A lot depends on specifics.
  6. In the end, you absolutely have to build and test. I generally agree with most of what you say here, but I would not agree that the issue has been figured out for decades. It'd be great if all we had to worry about was making the flow area big enough, but we often have to balance many competing concerns in the overall design, most especially cabinet size. And for any given design, there may still be 1 or 2 dB of performance that can be eked out with a tweak here or there. Of course a lot depends on what you're trying to do. Right now I'm experimenting with a dirt cheap 12" mid-woofer driver with a kind of whimpy motor compensated for by a very low Mms and soft suspension. It's got quite a bit of Xmax for what it is though, so it can make some bass in an appropriate cabinet. That cabinet needs to be relatively big to do it though, such that it's trivial to find a vent with a large enough flow area. This can make a decent, very affordable living room speaker, but the output density is poor. In contrast to this consider a Skhorn loaded with two 21" IPALs. With enough power, it's easy to overload the vents on that thing, and there's no easy way to expand the vents without making the cabinet bigger --- probably a lot bigger. The Skram, relying on a single 21", has proportionately more flow area than the Skhorn, but even it can be overloaded. Pretty much any kind of vent-based design in which one's trying to optimize output density is going to require difficult compromise involving flow area. Often this involves doing stuff like adding 90 degree bends to make the pipe or slot fit into the cabinet, and it's helpful to know how those features affect compression and whether the trade-off is worth it vs. shrinking the flow area to get the length to fit.
  7. Good guessing can save a lot of time and money that would be spent building physical things that don't work.
  8. 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!
  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. 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.
  11. 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.
  12. 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.
  13. 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.
  14. I'm relieved to know you set it on fire on purpose. With all the nasty wildfires fires around, I was imagining something much more grim.
  15. Hello also from Denver here! You started in on one of the Othorns already? I'm not sure how far along you are, but you may want to consider the Skhorn or Skram instead. They have much cleaner response to well above 100 Hz, and I think most people here regard those two designs to largely obsolete the Othorn. If you have a choice, I very strongly suggest you go with one of these newer BP6 designs. If you already have Othorns you want to use, then I'd say the situation is a bit of a mixed bag. The 80-100 Hz region is pretty narrow---essentially 4 semitones (or a major 3rd) apart. However, crossovers are not an all-or-nothing thing but involve blending over a pretty wide range, even with e.g. 4th order slopes. IMO, the Othorns will probably sound even better crossed even lower at like 60 Hz, at which point, a dedicated mid-bass section (for say 60-150 Hz) starts to make more sense. Looking at Ricci's compression sweeps for the Othorn, I don't expect EQ applied above 100 Hz will be very help helpful because the response above 100 Hz actually changes a lot with the signal level at medium-to-high levels. EQ which sounds good at low signal levels might actually make things worse at high signal levels.
  16. Can you show a picture with both the before and after overlaid? Also, the levels look very different between those two measurements. Did you measure both with the mic in exactly the same location? A tripod can help with this. Also, did you apply the same "smoothing" to each? How much smoothing did you apply? If you aren't sure, I suggest 1/24th octave to get a lot of resolution. I'm not familiar with the Architech software, so I don't know how it's used to configure EQ. Can you share the settings you used?
  17. The Wikipedia page might be helpful: https://en.wikipedia.org/wiki/Thiele/Small_parameters. It's useful to think of the physical system in terms of what they call the "Fundamental Parameters". Note that this is only one of many possible sets of parameters that must be specified to complete the model. A background in physics will help a lot there.
  18. The answer here is a quite complicated and technical. First of all, the parameters are for the Thiele/Small mathematical model of driver behavior. Without a complete set of parameters, the model is incomplete and not usable. Now, it's possible to work with simpler models that require fewer parameters, but will leave you without vital information about the operation of the system. The T/S model is also limited with regard to how it treats things like inductance, which is a particular problem when modeling subs. Hence, it's often useful to use a model with extended inductance parameters, which can be obtained by data fitting to impedance measurements. Now there's a second point. The T/S parameters you see published often are redundant. A complete set required for the model is only a subset of all the possible parameters you might see defined for a driver. For example, if you know both the mechanical compliance constant (Cms) and the effective moving mass (Mms), then you can directly calculate resonance frequency (Fs), meaning that Fs is redundant in that case. Other similar redundancies exist, but to understand them requires delving into a lot of details, and I want to keep this short. The long answer to your question is the subject of many books. Good luck!
  19. Yes---for some definition of "significantly". If you look in B&C's docs, they may describe their methodology for XVAR. Someone else's definition of "Xmax" may yield a different number. The measurements on DataBass don't give a specific number but simply show how compression and distortion increase as the driver is pushed toward its mechanical limits. Realize also that actual distortion depends on more than just excursion. Certain aspects of the cabinet design can suppress or amplify the effects of distortion in the motor.
  20. If they don't have DSP, then latency is not likely to be a problem. You don't need an external soundcard unless you need to determine *absolute* latency. But you don't need *absolute* latency to set up the equipment. You just need to know the latency *differences*. If you use a timing reference, the start of the timing reference is specified to be "time 0", and the the delay/latency in all the other measurements is relative to that zero point. Does this make sense? You can actually figure out most time alignments even without knowledge of relative latency by trial-and-error measurement of the sources (placed next to each other) playing together with different differential delays applied---the time aligned configuration usually gives the most overall output. You don't really need precise time alignment information unless you're doing advanced things like summing responses to see how, e.g. a separately measured left and right channel respond when playing together.
  21. Glad these are a hit. Spread the bass! Have you done any measurements in the room yet? A rudimentary standing wave calculation suggests you're likely to see a significant standing wave resonance at around 31.5 Hz (18 feet). Taming that a bit with some EQ could improve the sound quality further and might also make the lower extension mode sound better.
  22. When using two different amps at the same time, there's two things to be mindful of: (1) Setting gains appropriately. If you don't need the headroom, you can just gain match them such that all the subs put the out the same amount of sound, but this will cause the weaker amp to overload before the stronger amp. Another option is to adjust the gains to compensate for the difference in capability so that hopefully all the amps overload at the same time. That'll get you the most power, but the subs won't be utilized proportionally. (2) Setting delays appropriately. This may not be an issue if the amps don't have DSP. If they do, you may need to delay the signal going to the amp with the lower lag. This may not be a huge issue, if the lag difference isn't too much, but it's best to match the timings if possible. Note that it should be possible to address both of these concerns once and reuse the same settings for other venues unless you need to change them for other reasons (like special configurations).
  23. As long as you never push them into clipping, the sound quality between sub amps should be essentially identical. Likewise, if you never push anything into clipping, then you can probably just keep doing what you're doing right now.
  24. SME

    quieting a pro amp

    I'm not a fan of operating long-term with the chassis open. The chassis may establish necessary air currents for certain components. There's also the risk of foreign objects getting in and shorting things. And also if something goes *poof*, the chassis makes it much less likely that your house will catch fire. Good luck to you whatever you decide.
  25. SME

    quieting a pro amp

    Someone who is familiar with or owns the amp can chime in here if necessary, but my first thought is that a rack-style amp like that is probably designed for ventilation at both the back and front. Your cabinet is probably not providing sufficient ventilation with the doors closed. This might (depending on the amp design) cause the fans to ramp up a lot more than they would. So one option (that you may not like) would be to remove the acrylic doors. Though this might give a louder result, even with the fan speed staying lower, it should help the amp(s) and other equipment cool better. Another option would be to install large/quiet rack fans in a strategic location to better circulate air within the cabinet. This might take careful observation of the airflow and experimentation to get right. And of course, I don't know how much the amp fans are responding to temperature. Keeping electronics cool is always a good idea when you've spent a lot of money on them as heat is one of the primary factors that shortens electronics lifespan.
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