peniku8

That's good to know! I wonder how the SKHorn type of cab would behave (since my design is similar). Probably the same as a BR would? Hornresp shows much more gain in the lower octave thou; the horn contributes to the output down to about 45Hz (where the horn part crosses 95db voltage sensitivity).

Would be pretty easy if you happen to have a mixing console with a WSG server. The Waves F6 could act as your frequency dependent limiter, say a quick attack limiter for over excursion set at the 2 frequencies you'll need it at using a BR enclosure, and a slow attack long release limiter at the excursion minimum which sits a little lower than the amp's long term power limiter to prevent heat buildup from missing ventilation.

20Hz at roughly 85dbC. That's my guideline. Headroom is always good, but we have never mixed at a higher level than the 85db. Ideally ~18Hz 100db. It must be at least 4 cabs since I want to have a smooth room response to begin with and the FR should be decent anywhere in the control room. The space itself is about 3000cuft. It'll be DIY so the cabs are basically free, finish will be rattle can paint. Budget for the drivers is 1000€ max, but preferably within half of that. I have two B&C 12BG100 on order for a PA project, but might also test these in a sealed cab in the room to see how loud it would play. I've got plenty of leftover MDF anyways. That'll be something like a 14" cube. We're only doing stereo mixes and the room is basically a bunker with a single door and a tiny window. We're currently running a vented 8" sub, which is good down to about 25Hz. The single vented sub was more expensive than 4 of the aforementioned drivers, but back then I didn't know any better.

So I guess you're trying to say that this feature in hornresp should be pretty accurate assuming perfect conditions like single horn mouths stacked together etc? My cab would probably behave differently as it is a mirrored design and has two horn sections, so if you stack two you'll end up with only stacking half the horn mouths together. It also got decent baffle area so there should be some gains to be made in the upper register, which hornresp does not calculate.

Didn't @lukeamdman do something like that?

I've been asking myself if any of you have ever used this feature of hornresp and if it is any good. Here is a comparison of a sub design I'm working on; 1 vs 2 cabs. That improvement looks really good to me and with 3 cabs the response is ± 1db from 30-200Hz within this simulated environment, which would be stellar. I'm wondering if cabs do perform similarly when stacked and if this feature could be used (trusted) to get a remote impression of what an array of speakers would measure like.

I would certainly be up for that, but we'd be running into space issues very quickly 😅

That's pretty much the IPAL system. https://www.powersoft-audio.com/en/docman/922-powersoft-bac-ipalmod-system-introduction/file You get what you pay for, if you want to avoid burnt voice coils, using more cabs is the solution. Even the craziest DJ would probably not burn through something like 10 SKHorns in a club.

AES_Recommended_Practice_Specification_o.pdf

The fact that Josh ran full power (23s long) sine wave sweeps of a SP-6000 into two IPALs and said that they were not complaining shows how bad a worst case scenario can really be. Run several tracks with focus on the excursion minimum and voilà, melted VCs. That being said, I've run my FP13000 in bridged configuration at full power into two paralleled 21DS115-8 for several minutes to see how quickly they'd give out, but they didn't even complain. I had to clip the amp by a few db to actually get them to smell somewhat unpleasant. I turned it down and both drivers are still in excellent working condition. During these tests the amp was drawing a solid 3KW average power from the wall.

Can't speak for the cine guys (which usually work on the audio production as a team), but when I do a mix I usually do most of the part in the studio, but keep a pair of headphones close to review the mix there. Some extreme panning effects can sound good on a pair of speakers, but very distracting on headphones. Especially obvious when multi-tracking vocals and panning some to the extremes. I've done productions with some 20 vocal tracks layered, and when you get back into a segment with only 3 vocal tracks or so you definitely don't wanna have two sitting at 100% L and 100% R. I do the entire mix in the studio and when I think I'm mostly done I'll listen to it at home and take notes. I'll then again make adjustments in the studio and either master there or make stereo exports and master at home. Haven't done anything in the car yet and I'm not planning to. I know what that system sounds like, very well (and it's not too shabby with a sealed 15 in the trunk) but I haven't seen any benefit in doing that yet. Often you do that because your ears get used to the sound and you start noticing bad things less. That's where comparing your work to others helps alot to get back on track. I'm much more aware of sub 30Hz content thou since I got this monster sub here. Just recently I've checked out a new album of a band I like and noticed huge 17Hz spikes during a piano section in the song. Very funny that they didn't notice in the studio.

You have a point, but from my experience I can tell you that many mixing engineers also mix like they're not only deaf in a certain region due to their system's capabilities... You wouldn't wanna hear some of the mixes I've already recieved to master. When 90% of the mastering work is correcting errors made in the mix, you just sometimes wonder if they didn't actually didn't hear it, didn't bother or just didn't care. Some wierd things in mixes are also requests of band members. Once I had a mix where there was a very narrow EQ placed on the kick drum at 13khz. I had to pull it down by around 15db. Did so via M/S processing to not affect the cymbals too much. Might've also been a mistake when a compressor made the tube distortion from a preamp get out of hand, you can get some nasty spikes doing so, but I can't imagine that the mixing engineer was "deaf" to not notice this.

I'm very happy that I came across AVS and data-bass. The knowledge here is invaluable for me as a studio and live sound engineer. I'm a techie and I'm also obsessed with getting the absolute best out of my gear, so I could not live without a sub which isn't properly calibrated anymore. It's absolutely crazy that most of the guys on AVS (for example) do know more about the tech side and have better sounding systems than probably most (home) studios do. Like, if you have a good system to enjoy music on, you do profit. If the studio has a good system which they can mix on, everybody from the band over the studio engineer to the entire audience which will ever come across their tracks will profit from having a better mix (probably). Most of the time it's just lacking knowledge in the sense of "I don't know what I don't know", but a little research could quickly uncover many of the important topics. SME, about the plugins he is talking about, I think it's more of a budget allocation thing. Like, "rather buy plugin XY instead of the sub, it'll be a bigger benefit to your production quality". I could be wrong thou, I didn't read through his entire document again. I did the mistake of buying a Genelec studio sub for 1 grand. 1 grand for an 8" sub which can't reach below 25Hz properly (oh wonder). I'll probably build like 4 sealed 12" subs to replace it, just don't know which driver to use. Our control room is the worst thou, and I have a ~40db spike at 30Hz ANYWHERE in the room (after doing a sub crawl with about 50 different measurements I gave up). Equalizer APO has to take care of that until I get a decent hardware solution (maybe a 10x10HD??). I'm often doing masterings at home, where I can trust my SKHorn. In the lowest tuning it reaches down to about 14Hz where distortion gets out of hand. I just need to replace my Klipsch speakers with DIY ones and my home setup is complete (apart from the 7.x.6 system in the currently non-existing dedicated theater room), although I gotta say that I'm not unhappy with the Klipsch towers. Well, I'm gear obsessed, but I think I have all the reason to. It's my (dream) job after all 😊

I wonder how it would affect rining if one EQ'd the sub flat to say 5Hz and recreated the sub's rolloff in a dsp. Probably not in a good way. I only noticed ringing while using a heavy linear phase EQ on a snare drum (pre-ringing that was) once and never had any issues with it again, but I'll pay more attention to the phenomenon in the future and see if I can get a clearer picture for myself. Ringing is a very interesting topic which I only once read about. Did so when trying to figure out how to deal with phase issues when EQ'ing and most of the info I gathered was 'was really bad in the analog world, but is not an issue anymore with digital EQs). Guess I could do some tests to visualize how something like a FabFilter pro Q-3 handles ringing.

Cascading limiters is only possible with some of the more costly amps I know (Powersoft, the new Crowns, Linea, Lab PLM etc.). My Sanway D10 has a variable attack time, you'll be able to limit the signal to say 50% of the RMS rating after it has been anywhere above that level. By saying anyware, you'll already know that you might be sending 2KW into your sub for the duration of the attack time, which might already burn some smaller drivers. If you have different limiters available, I'd set one with a quick attack (say 5ms) to cut off peaks (somewhere in the ballpark of 2-4x the rated AES power), one with a moderate attack time of around a second or two to cut off anything above 1-2x the rated AES power and one long term average which gets active after 5s and limits the driver to less than half the rated AES power. I'm not an expert on this either, but I'm sure you get the point. Ideally you'd set up limiters to be transparent, and that's more or less possible with a peak limiter, but the long term average limiter might be quite noticable if you have a DJ pushing the absolute max with sine waves. You'll just notice it drop off at some but, but you will know that your limiters work as well 😉 As Ricci already said, the worst cooling a driver can get is at its excursion minimum. And when the amp is heavily clipping that's basically every frequency since the clipped since wave (aka square wave) means no movement at the peaks. If you take a 1KW sine wave and clip it to the point of getting a square wave with the same peaks, you'll end up with a 2KW signal (the peak voltage of a sine wave is the rms voltage multiplied with the square root of two and every voltage increase squared is the power increase, so that's twice the power).

That would've been something around F30 +2db Q0.6 I doubt that you'd notice much ringing with a filter like that, but you're correct of course. Also, I think modern processing equipment should be accurate enough to reduce rining to a minimum. There are mixing consoles that have up to like 48bit 192khz processing. Mixing desks are not loudspeaker management systems, but I'm sure these are also up to modern standards. I'd prefer having the subs and tops at the same volume at the crossover frequency (you can easily measure that by sending them a sine wave at that frequency and measuring the SPL of the speakers individually), then applying an EQ to shape the response of the entire system. You can experiment with different low shelves. On my home system I have the sub some 9-10db hot and have a low shelf of +6db 30Hz Q0.7 on top of that. PA systems often start rising up to like 10db from 1khz down to 100hz and some more in the sub region. At least that's what I've seen in an l.acoustics paper.

This Article is the HiFi version of this: https://joeysturgistones.com/blogs/learn/why-your-studio-doesn-t-need-a-subwoofer A very well respected studio engineer, known for mixing big Metalcore acts, such as Asking Alexandria. And I'm sure the causes for writing such an article are the same in both cases: bad integration which lead to the descision of discarding subwoofers entirely. Those are the people who cause the necessity for BEQ, by just filtering out the low-end which their system can't reproduce.

I'd put a broad PEQ at 30Hz to bring that up to equal the 70Hz SPL (or bring down the 60Hz alternatively). The 85Hz notch might need some experimentation, it may or may not be good trying to even that out. It is so narrow so you might not even notice a difference. I found that positive EQ'ing with a very high Q often doesn't provide any desirable changes. When EQ'ing out phase problems I encountered a very wierd pumping phenomenon once. When playing back a sine wave, the level would sway, as if it was controlled by some kind of LFO. You can't recover the 150Hz dip as that it caused by internal phase cancellations. More energy also means more cancellation so you end up without any improvements.

Scaling it up would drop the tuning quite a bit. Making the cab only wider wouldn't make much of a difference, but if you also increased the height you'll end up with a pretty big spike around 30Hz so you'd want to EQ that out. Here is a response with the wider cab (guesswork):

You'd want the tuning frequency of the mains to be way below your Xover freq, due to the phase swap at the tuning frequency (that will cause cancellations with the sub). I'd advise on going with a ~70Hz tuning frequency (or lower if possible) when going with a 100Hz crossover at 24db/octave or greater, that would leave you with half an octave of constructive interference. I would also not cross over that PD driver at much higher than 2k to the tweeter. I prefer using networked dsp amps. That way you have quick access to all your amp channels and settings from your PC, which allows for efficient system tuning. When using the FP10kQ for two subs, either use it in bridged config (with 8Ohm drivers) or one cab on channel A and one on channel B to prevent bus pumping. Depending on how big the dance floor (?) is, it might be advisable to put two cabs low and hang two up high, so you can individually adjust the SPL of the tops to get an even coverage.

Another thing that confuses me is the fact that you did use the lossy inductance setting, but did not use the additional parameters you measured, or am I confusing things now? When I enter values there and turn them on, the Le will light up in green. The SKHorn measurement looks closest to the simulation without these parameters (Le in red), so should I model my design (using the IPALs atm) with Le in red? I'm trying to understand this setting, am I correct in saying that lossy inductance kinda does what the shorting ring is doing in the driver?

I would advise against the combination of hardwood and plywood, as both have a different CLTE. If you're really limited on weight, you may try to use Banova Ply. You could do the outside panels with regular BB ply and use the Banova for all inside panels (and be my guinea pig at the same time 😉) Other than that, I'd personally prefer to keep the speaker count at 2 (i.e. one or multiples of the same sub playing from 20/30Hz to 60/100Hz depending on taste and a (line) array of the same capable tops). I had a quick look at the tops you're using and I don't see the need for a kick bin here. The dual 10" tops should capable enought to cross them over between 80 and 100Hz, and the Skrams are capable enough to be used in that configuration as well.

I got into HR today and finally understand why you chose to correct the parameters like this. How much of a difference did this make?

That's cool, I was thinking about boosting the bottom end (via a larget back chamber?) to get a flatter FR. Or would multiples (open air) do the trick similar than horn stacking? I know you get the biggest benefits by making a cluster, but I'm not a fan of that. I'd either go with one per side for smaller shows or a row of 4/6/8 along the stage front. Ya know, beamforming n stuff I mixed a show for around 1600 guests a while ago and there was a center bass cluster of 4 SB218. When mixing so that the FOH had a decent bass level 20m from the stage, the first row near the subs was in for a hell of a ride. When hearing complaints about the kickdrum, my question 'where did you stand?' was always answered by 'right in front of the stage'. I don't wanna do that when I'm in charge of the PA.

@Ricci, how do you expect the new Eminence driver to perform in the SKHorn? Its stats looks pretty close to the IPAL, with the benefit of being a good fit for 2Ohm stable amps (with 3Ohm per Cab per channel), but the efficiency might be a little lower? Since the LaVoce dropped in price here (to 350€/400$, can probably get it even cheaper) it would also be interesting how one would modify the cab to fit that driver better (slightly larger back chamber?).