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Everything posted by 3ll3d00d

  1. haha nf is the only place I have space for more subs! I don't feel the need for more output. I think a ported sub vs the dual I have now will have v similar output capability from ~15-16Hz up. The current sealed makes it to ~9Hz before giving up. I don't think my wooden floor behaves in the way that people in the US describe a slab vs suspended floor, IME a suspended floor in a Victorian house in England is a v different thing (solid but leaky if that makes sense). overall it seems to boil down to; 1) is more output from subs at the front from ~9-16Hz more important than more output from ~10-30Hz NF? 2) are there negative effects from ported in the 15-30Hz range? 3) will a pro style woofer be materially better than a UXL in the 60-100Hz range? my guess is this reduces down to option 1 as the main point of difference
  2. The increase in tactility from NF placement is not remotely debatable in my experience, move more air in close proximity to the listening position and perceived intensity goes up. Achieving an equivalent effect from subs further away requires *much* more output, is a different sensation and requires SPL a fair way above my usual levels to get there. Note that I am talking from practical experience of adding/dialling in NF subs in my room.
  3. I currently have a dual sealed UXL-18s powered by a speakerpower sp1-6000 amp up front with a pair of 12s nearfield behind me. This works well so I wouldn't be changing it except for the fact I want to make a change to the layout up front, this means I need to build a new sub up front and that got me thinking about options. The new space available is ~4' tall x ~15" deep by ~2' wide (probably a few inches extra is ok in this dimension), i.e. something like 200-220L total internal volume. I can see 2 approaches, thought I'd post them here and see if anyone has any thoughts on which way to go. The first one is basically the status quo option, i.e. leave the nearfield as is + build another box of the appropriate dimensions to hold the 2 UXL. The 2nd one is; - move the 2 UXL to nearfield, power these by the speakerpower - build a single ~180-200L ported box tuned to ~18Hz using something like a FaitalPro 18XL1800 or B&C 18sw115 The thinking here is that the ULF wobble might be delivered more effectively by a pair of UXLs nearfield and I can get drivers like the faitalpro or B&C easily (whereas HT sub drivers like the UXL are non existent in the UK). Thoughts?
  4. I think that sort of thing is most commonly packaged as a VST
  5. where do you get the idea from that a PC is limited with respect to what sort of filters it can apply?
  6. now with windows exe -> https://github.com/3ll3d00d/vibe/releases/tag/0.2.0
  7. this is now available for some platforms, see http://vibe.readthedocs.io/en/latest/for details
  8. I use a VBA on my nearfield sub for this purpose - http://www.avsforum.com/forum/155-diy-speakers-subs/1939713-active-bass-trapping-using-spare-subwoofer.html it's equivalent to the signal a real sub would produce so basically the rear woofer has inverted polarity, a delay offset and a reduction in level (in order to get it to "meet" the main wave). A purely passive device wouldn't give you that level of control so I'm not sure how well that will work.
  9. is it possible to approach this from a different angle? i.e. assume that port compression exists and model that effect (on driver excursion and output) this would be analogous to the way you can model the effect of power or excursion so you'd set a port velocity limit of, e.g., 10m/s and then see what happens next.
  10. quick demo of progress so far https://youtu.be/YK-4kOTLzqA
  11. perhaps try it again with your eyeballs wobbling from the 140+dB, it might be quite hypnotic at that point
  12. deeper is always better obviously it sounds like I'm being deliberately obtuse but I'm not, honest! Basically my approach here is model some stuff, see what can be achieved, compare to what can physically fit in my room, compare to what I have already and see whether that is something that makes sense as part of the system, repeat until I find one or more interesting things to build. This means that, atm, I'm asking "what can be built using this approach with 10-12" drivers?" rather than "I want something that can do 110dB at 25Hz" (or whatever).
  13. Perhaps, not sure really, depends what the models come out like. By that I mean the main position available is fairly near field. I don't really have a plan atm tbh, just want to try modelling some different designs and see what happens.
  14. Size is the main constraint, about 2' high x 20" deep and maybe 4' wide. Budget I don't mind.
  15. do you have any suggestions of drivers that would work in a scaled down version? e.g. using 10s or 12s. Just curious to try modelling a smaller version before I decide what to build this year.
  16. Yes that would be nice, lots of posts out there that basically involve people saying "do xyz and post the results (or share your data)".
  17. I thought I'd point this out over here - http://www.avsforum.com/forum/155-diy-speakers-subs/2681865-rpi-based-diy-vibration-meter.html I'm currently working on writing an app that uses an offboard accelerometer to measure triaxis vibration in order to try to get a better, more accurate, view on (what has been dubbed) tactile response. The thread on avs has details on the app(s) itself so I won't repeat that, happy to discuss it here for those who prefer db to avs though. One reason I post it here though is because the analysis portion will actually do what we use speclab for today when it comes to analysing soundtracks, nowhere near the no of options that speclab has but then I don't think they're actually used anyway for the most part. I plan to eventually add support for automated analysis (and playback) using something like a jriver playlist too. This is a nice sidebar for me because I find speclab an extremely irritating application to use. It wouldn't be hard to extend it to capturing from a mic though I doubt I'll get to that for a while (if I ever do... code is freely available of github though should anyone want to contribute!).
  18. acourate does do something that looks like a form of dip limiting along with some sort of perceptual weighting, no idea how it is implemented. The REW version is done via the application of a cubic mean. I don't think it is an unreasonable approach psychoacoustically because we are less sensitive to dips, particularly the extremely high q high(er) frequency dips that can result from an FDW. IIRC the typical filters used to a model the response of the basilar membrane looks more like a 4-6 cycle long window so 2.2 cycles seems a bit short to me if first arrival perception is the target but there aren't fixed rules here, if it works then great. Certainly using a shorter window should result in more robust (less position sensitive) results. The time between first and later arrival needs to be extremely large (relative to reflection times in a typical room) for it to be heard as separate events though.
  19. I don't think anyone debates multeq xt is basically a bit rubbish do they? I use a frequency dependent cycle length, I can't say it is a critical feature though. Perhaps I have been doing this too long but I am quicker to cut to the quick these days. I would like to think this is a nice benefit of spending a good few years doing this so that I can get to a good outcome fairly rapidly. I imagine I am lazier too though Anyway I would say the issue with rew fdw is that it tends to produce a V spiky response when you go past a few cycles which makes it harder to use.
  20. An easy online alternative is to paste it into https://www.mathpapa.com/algebra-calculator.htm i.e. Paste into the box and click evaluate, you can then enter values for gain and q 1/((((1/Q)^2-2)/((10^(d/40))+1/(10^(d/40))))+1)
  21. I mean measure the response of your combined set of filters by running a sweep through jriver, compare against the expected correction.
  22. If you solve for S you get 1/((((1/Q)^2-2)/(A+1/A))+1)
  23. You could run a loopback measurement through those filters to see what the shape looks like.
  24. sounds like these ideas might help http://www.diyaudio.com/forums/pc-based/298248-sensorless-dsp-cone-excursion-limiter.html http://www.diyaudio.com/forums/pc-based/289698-idea-linkwitz-transform-ladspa-plugin-lookahead-boost-control.html
  25. If it's the only source then your life is much simpler as you no longer need an audio interface with inputs and you don't need a dsp engine that can sit there spinning on those inputs. If you're using windows then EqualiserAPO is one option, jriver is another. The former sits in the windows audio system and does not use ASIO (or WASAPI exclusive), the latter is more flexible as it has both a loopback interface for rendering system sounds and is a player with a highly capable dsp engine as well. jriver is probably the simplest option as well as one of the most feature rich from a DSP point of view. If you prefer linux then jriver is still an option, though lacks the loopback interface, brutefir, for a convolution option, and some sort of alsa/ecasound combination. An audio interface can cost as much as you want to spend, from a few hundred up into the stratosphere . My preference is to use the same interface for measurement as for playback so I like devices that have at least 1 mic pre and at least a few inputs. To that end, I've had a focusrite saffire pro 24 and an rme ff800 in my system though I currently use a motu 1248. A motu ultralite mk4 is probably one of the most compact units you can get with sufficient output channels, has solid drivers and should be a nice, clean interface. On the other hand, if you go for something that is purely a DAC then a cheap option is something like the minidsp u-dac8. Ultimately the right choice for you depends on how much you believe in the impact of electronics on audible SQ, certainly lots of options out there anyway.
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