peniku8
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Posts posted by peniku8
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On 8/5/2019 at 3:01 PM, kipman725 said:
As long as the mouths are within 1/4 wavelength they should act as if they are stacked together.
That's good to know!
1 hour ago, Infrasonic said:From what I have read, I thought it was that tapped horns will not benefit with additional extension when used in stacks. However, front-loaded horn bass systems will.
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).
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On 8/7/2019 at 3:06 AM, radulescu_paul_mircea said:
There is something I would love to see more in the processing units: a limiter with a frequency dependant Threshold. A limiter with an output Threshold and given attack and release constants that are not engaged ,even if passed, if the frequency is not in the right band. Powersoft does have this, I think Linea Research has something similar and I've seen this behavior in Eminence D-Fend.
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.
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1 hour ago, SME said:
You mentioned you are considering 12"s. Is that really as large as you can go? How big is the room? How well contained? How low do you want to go? Do you need to monitor multichannel content with LFE? Or just 2 channel content?
If you want low, loud and small, be prepared to spend some serious money.
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.
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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.
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Didn't @lukeamdman do something like that?
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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.
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35 minutes ago, dgage said:
I think studios should standardize on the Mariana 24 just so they wouldn’t miss those sorts of issues but I might be biased.
I would certainly be up for that, but we'd be running into space issues very quickly 😅
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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.
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Quote
4.5 Power-Handling
4.5.1 Test Conditions and Equipment.
The lf driver shall be mounted in free air so that the direction of motion of the diaphragm is in a horizontal
plane and so that there is no appreciable air loading from adjacent structures. The driver shall be excited with a
band of pink noise extending one decade upward from the manufacturer’s stated lf limit of the device. The
noise shall be bandpass filtered at 12dB per octave with Butterworth filter reponse characteristics, and the peakto-
rms voltage ratio of the noise signal supplied to the lf driver shall be 2:1 (6 dB). Refer to Appendix C for
recommended method. The manufacturer shall state the upper and lower cutoff frequencies (– 3 dB) of the
noise signal.
4.5.2 Test Procedure.
The device under test shall be subjected to successively higher powers and allowed to reach thermal equilibrium
at each increment (approximately 2 h). Power shall be determined as the square of applied rms voltage, as
measured with a “true rms” voltmeter, divided by Zmin. The rated power of the device shall be that power the
device can withstand for 2 h without permanent change in acoustical, mechanical, or electrical characteristics
greater than 10%.-
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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. -
59 minutes ago, SME said:
Likewise, I doubt most cinema mixes get heard on more than one system before they are finalized. I believe a wide assumption in that field is that because all the different cinemas are calibrated to the same standard, the mixes will translate. In contrast, I believe music engineers often test on a few different systems (such as their car), and I think this drastically increases the chance of hearing problems hidden by their main set of monitors.
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.
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15 hours ago, SME said:
That's an excellent point, thanks!
And now that I think about it, my post may have come off as a bit judgmental about use of plugins without the ability to monitor the result. In practice, I believe this is done all the time and works out OK and sometimes can sound really good. It's probably the case with most popular "unfiltered" movie soundtracks including those that have been subject to high quality custom BEQ to make them "unfiltered". However, embarrassments do happen. They are more likely to happen when processing is applied by the mixer blindly, or deafly as it were.
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.
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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 😊
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22 hours ago, SME said:
The ringing is already there, as a consequence of the HPFs (both electronic and acoustic) at/near the vent resonance. It can't be avoided because it's a fundamental property of the cabinet alignment. One would have to EQ it to not roll-off at its lower limit, which obviously doesn't work in practice.
If the PEQ you suggest does not precisely cancel a comparable existing dip there, it will also contribute its own resonance. At Q 0.6, this is likely to substantially affect transient sounds, in addition to directing listener perception disproportionately toward the ringing that's already there.
While many DSP systems have problems with filters at very low frequencies, I'm not discussing that here. The ringing I speak of arises from the final response shape, not necessarily the processing that was applied to get it to that shape.
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.
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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).
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1 hour ago, SME said:
I would not recommend PEQ boost of any kind at 30 Hz. The vent tune will already be contributing to ringing there, and any boost, no matter the bandwidth, will likely accentuate that ringing. If instead of cutting from the top, one wishes to boost the bottom, I'd suggest trying a low-shelf filter if that's available. A low-shelf centered around 75 Hz (looking again at the sims) could be helpful. However, I think the PEQ I suggested above may be a better place to start.
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.
1 hour ago, jay michael said:Ok thanks for the suggestions. Next time I have it up and running ill tame that peak at 200 a bit. I had the subs pretty close to my fence this time so perhaps that's what was happening around 85. Agreed fully on increasing the sub output. I kept it flat-ish to not completely drive my neighborhood nuts. In your opinion which is the better way to do this? Reading the manual for my venu360 dbx seems to suggest I should turn down the attenuators on my amp powering the Danleys until the desired top vs sub balance is achieved. I have read from other sources to just boost eq for the range controlling the subs.
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.
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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.
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24 minutes ago, medico said:
Very interested in this design, going to build one just to have a listen. and if i like what i hear i'll build 8+ of them for our rig!
one thing, our tops are bit wider than these, so would want to build them 2inch or so wider, and possibly taller, by few inches.
will this effect the sound in anyway? and how so?
also whats the best way to do so? scale everything up?
look forwards to hearing back!
Cheers,
Murdoch
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):
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1 hour ago, Timon said:
before we start building the second skram we want to make oszillating tests with test pieces. These will be like 18mm birch, 12/15mm Banova with Epoxy(and carbon), the same with poplar and birch. we are not sure zet, how the tests will look like, but we are sure to find a way
after that we will make the second skram out of the winner design and then compare it with the orig. skram.
Ok, those are interesting advices, so we will try it without them first. Is it a good idea to make i.e. per side one Top with both, the AMT with the Mids, and a second cab with only the Mids (eighter with the same tuning or again as a 4 way system?)
Another change in the plan is to use not the very expensive wavecore driver, but instead something like the http://www.precision-devices.com/Product-Details/PD103NR1.
Would you recommend a closed box or BR Design for a tuning around 100Hz? (And could you recommend a calculation tool for designing these?)
We want to use a FP10000Q for two skrams, another class A/B or D amp for the L/R mids and one for the L/R AMTs.
Do you prefer a single extern dsp for all amps or the build in dsp (with much more effort)?
Thanks a lot for your help, I think we are on a good way!
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.
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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?
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4 hours ago, Timon said:
Hey guys,
my friends and I are totally fascinated of the Skram and for now we want to build two of them.
At first we'll build it with the suggestion of 18mm birch ply and after we want to do some changes for the second pair.
We want to make it out of 15mm (birch ply and in the core of it one layer of poplar to make it lighter) in addition we would like to add some braces out of carbon and epoxy resin. It should be as stiff as 18mm birch ply.
Do you have any concerns with this plan so far?
We are not sure which kickbass we want to use. Maybe some of you have another recommendation?
We are planing to build one Cubo Kick 12 per Skram with the 18sound 12ND830 and a little bigger cabinet, so it gets down to about 90hz and up to 200hz.The other consideration is to build some sealed boxes as kicks but I worry if this will be very harmonic with the Skram. But it should be the purest and punshiest kick, isn't it?
There would be I guess 2 drivers per Skram (depends on the driver).What do you think? What would you do?
As Mid Tops (from 200hz with a bit different AMT) there will be one Mundorf AMT (with a horn in front of it and an active cooling system) and two Wavecore WF259PA01 per side.. Mostly we are playing Techno, etc. but the system should be able to let you enjoy all kinds of music...
It would be great if you guys could help us with your knowledge!
Thanks a lot,
TimonI 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.
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On 2/20/2018 at 10:10 PM, Ricci said:
Incorporating the air volume in the driver cones rather into S2, instead of using Vtc and Atc, was a much closer match for this design.
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?
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5 hours ago, Ricci said:
NSW6021-6 looks great in either the Skhorn or Skram. No mods needed. Same for the SAN214.50. If you want a larger back chamber use the Skram.
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.
after that we will make the second skram out of the winner design and then compare it with the orig. skram.
The case againts subwoofers
in Bass Content
Posted
85dbC was what I measured during one of our mastering sessions, to get a general idea. This was with the sub measured to flat at the MLP and a house curve bumping it to +10db below 60Hz. We're a studio for music and our content is absolutely crushed.
We're not really limited in space, it's just due to the looks of it. I could probably fit 3 SKHorns behind the desk, but it wouldn't be pretty. I like the 12BG100, the only thing that's bothering me is the Xmax. 21" would get me further with less money, I've realized that a few times already, but I don't think I could justify the footprint.
Keep in mind that there will be 4 sealed cabs and that our single 8" is basically (almost) enough already in terms of SPL. I doubt that 4 sealed 12" would perform worse than a single vented 8"