SME

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.

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.

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.

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.

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.

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.

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.

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?

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.

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!

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.

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.

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.

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).

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.

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.

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.

Re-work in QC? I imagine that whatever is going on here, it's probably not good. I would agree unless you think you can develop your general knowledge/skills further by finding new ways to break it.

I didn't see any mention of possible wire/lead slap, though that should be obvious in a free-air evaluation.

This has all been quite interesting to follow. I'm going to take a wild guess that the mixture of transistor types on the old board was "on purpose", albeit maybe as a last-minute hack or tweak for some other problem. Hopefully the new revision is a lot more robust.

Are you using some kind of EQ (manual or auto) right now? Do you have any idea how much output you need vs. what you have in the deep bass? Switching to sealed will give up a lot of it, and you might want to use some EQ to bring response up, at the cost of more headroom. The other issue will be the integration complexity, especially if the Magnums-12s are ported, as noted in earlier posts. Will all the different subs be placed close to one another? That definitely makes it easier. Will sealed vs. vented Magnum 12s give you enough output? If so, this will give you the most flexibility with respect to crossover.

Is it really that simple? My first instinct is that it's different because the energy storage mechanism is a bit different.

That 50% figure sounds impressive, but one question I'd have is how much charge does the capacitor effectively store at that voltage? I assume it's probably a lot less than it stores when fully charged, but the Wiki article doesn't clarify this point. I mean, I fairly routinely discharge tens of thousands of volts by simply touching things around the house, but this static electricity carries so little charge that it's essentially harmless to me (but not necessarily my equipment).

Kinda off topic, but I once hooked up some "jumper" cables between 12V car batteries, and I didn't know that the other car had an electrical short. The insulation started melting off and smoking within a few seconds. Even if the electricity didn't have lethal potential, somebody could have been burned pretty bad.

So the tune is probably at 21-22 Hz then. You'll likely get decent output down to 20 Hz with some audible bass down to ~16-18 Hz, depending on what your room is doing.