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Everything posted by andy497

  1. Ages more, and my newborn turned into a semi-autonomous mini-human, and I can turn some attention back to audio. I swapped an aging laptop for a proper desktop HTPC, and many of my measurement problems (particularly with time-alignment not being repeatable) seem to have fallen away. Repeatability is very good, and predicted sweep summations using dozens of biquads across different channels match reality almost exactly. Quick q: Is there a way to export out of MSO into a minidsp compatible import format (i.e. xml)? If not, I may write it. It looks like it still will take a couple steps to export everything (I can only see a way to export biquads from MSO one channel at a time), but that would save a ton of time inputting when I'm trying out various configurations. p.s. I can't tell you how much I appreciate this incredible application. It's a huge contribution to DIY audio.
  2. Cool. Re: horsepower, yeah I found that was the easy part. I have an aging intel i7 for the HTPC, and I found that 64k taps x 7 channels took a few % of cpu. No big deal. The much bigger pain is the video delay, or lack thereof. I went down to 16k taps and am doing everything I can to speed up the audio chain and slow down the video one, but it's still never enough. Jriver, which I love/hate but mostly hate, handles movies fine, but I find I watch a lot more youtube in practice. Adding to that, shrinking buffers too much gives me popping and crackles, and I hate that even more than bad lipsync. First world problems...
  3. SME, are you DSPing with IIR or FIR? I was imagining lin. phase crossovers if you're doing complicated mains/subs blending, but at the low frequency end like that you're talking huge numbers of taps and possible major pre-ringing. I'm now very curious what you mean by each crossing mains to each sub differently. Can you elaborate?
  4. Back in college I got to spend a fair bit of time with an NMR (analytical chem version of an MRI, except the center diameter is super focused at maybe an inch or two across). I don't recall the field strength for sure, but I seem to remember 7 teslas, and it needed to be cryogenically cooled 24/7. It was strong enough that the CRT monitors in adjacent rooms had their displays color smeared and contorted permanently. We were required to empty our pockets next door and carefully inspect clothing for any ferrous bits, else we couldn't go near it, although I don't think anyone ever tested that. You had to climb the thing and get right near the meat of it to drop a sample in the top, and that was always a little scary. I thought my glasses were going to tear off my head or something, but there was no perceptible sensation. The cooling was actually hard work, so that might get tricky when retrofitting for voice coil duty. It was double vacuum-walled with an outer chamber containing liquid nitrogen and inner with liquid helium, and they're always slowly boiling off, so you need to keep replenishing or it will destroy itself. Liquid nitrogen is pretty common but helium was crazy expensive.
  5. There is so little spider left in that basket compared to the coil, I would think that would severely limit the travel. I wonder what guided them down these design decisions.
  6. I don't see why you can't correct that with FIR. Let's say you want to compensate down to 2 cycles at 20hz = 100 ms. At 44.1 khz sample rate, that's 8,000-ish taps. How many do you have available?
  7. Has any work been done on controlling the boundary layer in ports to delay separation? (e.g. targeted roughness, turbulators, etc.) I know you need to have a precise idea of the Re range, and any given solution will be very specific to that port geometry. Also, I imagine everything may be different in an acoustic resonator; not only is flow not fully developed, it's constantly changing direction. Still, it sounds like an interesting area of study.
  8. Lots of excellent insights andyc56 and SME. I think I found my culprit. I pulled into REW two different "before" sweeps of the same sub, one a day apart of the other but with otherwise identical settings, to see if I could make one look like the other. A simple low shelf (what the internet speculates the THX BGC does) won't do it. At some point, I tried the REW merge A to B function, and it became obvious. When REW realigns the timing of one sweep, it becomes a mirror match of the other. That leads me to believe that BGC perhaps wasn't on and I just had a hiccup with the acoustic timing reference. My avr tries to be smart and retain all of the audyssey/speaker config/etc settings per input, but that can drive you nuts when you're testing and frequently changing inputs. I'm surprised that a timing shift like that would so dramatically change the computed frequency response. Further, my intuition would be that higher frequencies would be corrupted while the longer wavelengths would barely change at all, but it almost looks like the opposite - at least in the < 200 hz range. @SME: I've heard that minidsp had trouble with the low frequency filters, but I wasn't sure why. 56 bit fixed seemed like plenty. In fact, it's been a long time since my last comp. sci. class, but I thought 56-bit fixed offered better precision over a small range like -1 to 1 than float. Or is the quantization error happening somewhere in the multiply steps? Again, my understanding was as long as you have enough bits for the accumulator, fixed bit multiplies will have less error that float, especially if you're multiplying small and big numbers together. I suppose the devil is in the details. Upshot: I had time last night to take a new set of sweeps for just one position and let MSO do it's magic. The result of a few minutes of computation was pretty darn good. I plugged in the coefficients and measured +/- 2 dB or so over 10 to 90 hz where the subs drop off (I wasn't integrating with mains), and less than +/- 2 from predicted to actual (majority of difference 10 to 20 hz). I'm encouraged. Now I just need to ensure that I don't have timing problems when I take new baseline measurements. Once I move the mic to a new position, there's really no way to repeat a previous measurement to know for sure that it wasn't faulty. p.s. I looked at predicted/measured phase response for a sub with an all-pass, an LT, and a bunch of peqs applied, and it's really beautiful. Very good phase agreement from 10 to 100 hz.
  9. One week update: Well, I thought things would get easier after my initial 35 sweeps, but the work was just beginning. I have been having a terrible time getting the expected results on a given channel before/after eq. I thought this might be related to my reliance on the acoustic timing feature in REW, but that's working swimmingly. I can run individual sweeps of four subs, then sweep the combined subs, and the computed sum sweep in both REW and MSO precisely matches the actual. In contrast, when I sweep an individual channel, apply a simple eq and sweep again, the results are close but not quite right. Lots of eq points and lots of channels compounds the errors significantly. Obviously you can't do things like boost away a null, but I'm not trying to do that. I would expect simple eqs of a single channel in "minimum phase" regions to behave predictably, just as changing the master volume on the avr produces a nearly identical sweep at a different level. Which leads to... Finding no. 1: I discovered last night that THX "Boundary Gain Compensation" was enabled on my receiver. It's hidden in its own menu folder far away from things like audyssey and bass management. Grr. I thought this was a simple shelving filter, but before and after sweeps reveal some really strange effects on different sub channels. I turned that off and took yet more before sweeps, so I'm hopeful I'll see improvements when I'm able to next take some measurements. This means I'll probably want to remeasure and adjust the whole signal chain again as well. Finding no. 2: I'm growing skeptical of the "import REW biquads" function in my minidsp with 2x4 advanced plugin. First, it will accept 6 biquads for both the input and output channels, but there are only 5 peq slots. I'm guessing the 5th is dropped, but it's very hard to tell from the interface. I've been using five per bank just to eliminate that as a variable. Secondly, while biquad computations seem to match fairly well from the minidsp display graph to the MSO software, they aren't fully one to one. The frequencies of adjustments seem to line up, but the magnitude may be slightly different, especially on large adjustments, e.g. a deep cut that equates to -15 dB in MSO might register as roughly -12 dB in the minidsp graph. I'm thinking the minidsp plugin graph is pretty small and might have some display/rounding issues, but it has me wondering. Besides the disappointing and confusing results so far with before/after sweeps, I think the software is fantastic. If I give it a few peqs and shelf/LT/all-pass filters to work with, it's amazing what it can do. With a first-order all-pass to play with on each sub channel, it's able to align them such that the low-end extends down to a 8-9 hz -3dB point with almost no boost on any channel, and there is more headroom across the entire bandwidth. Setting things the old-fashioned way, my -3 point has been in the tweens with a 12 dB LT.
  10. Thanks for all your effort andyc56! I've got about 16 hrs into MSO now and am having a bunch of fun. I'm out of town now, but I have three different i7 machines at home which will be cranking on solutions all weekend. Quick q if you have a sec: Is there a way to define global eq constraints? e.g. min/max cummulative boost. I'm limiting individual peqs, but it keeps stacking things and occasionally wants to push one channel up 25 db. For the same config I've gotten nearly as flat with no channel pushed more than 6 db, but i don't have a way to make it prefer smaller effective boosts/cuts.
  11. I can't believe I've never run across that software before or seen the thread on AVS. Very exciting! Getting a proper timing reference in REW has been a major pain in the past, but I'll keep at it. I was very excited about that recent feature since I do hdmi out with a usb mic. However, I've had nothing but problems with the asio half on the two laptops I have. Both are modern windows 10 machines but share the same behavior: I can get 3 or 4 measurements tops, at which point the next sweep has an awful crackling sound. I haven't been able to fix that with adjustments to the latency/buffer settings; the only way to correct it seems to be to reboot, and then I have to do another dance to get the hdmi device and minidsp usb mic detected correctly. I'll typically get no sound from the mic, or it will appear to work normally but the signal recorded by REW will sometimes be 18 db too high.
  12. That's really interesting about -3 dB in REW. That's typically where I've been measuring. This scope has a half-way decent FFT function, so I'll be interested to see if I can see that as THD in the output compared to -3.5 or -4, though I suspect you could do this more accurately in REW. Now I know to listen for it too.
  13. I'm kinda surprised that audyssey dynamic eq does not do any time-based dynamic compression. In fact, when listening to music with dynamic eq on, I could swear I've heard the pumping effect from the attack/decay triggers getting tripped regularly with the beat. Perhaps it was in the source material, but it seemed to go away when dynamic eq was switched off. That was an unusual occurrence however. Most of the time I like it.
  14. The 31.5 hz sweeps where max output continues for a cycle or more after input stops just boggles my mind. Crazy. And it's interesting that this type of "distortion" slips right by CEA-2010 testing. I guess you'd see that as ringing in a waterfall graph?
  15. I'm experimenting with not just matching/optimizing via a fixed distance delay, but using IIR all-pass filters to mate the group delay of different driver alignments. For reference, I have four sealed subs (4 cubic footish) and two ported (12 cubic footish, ~16 hz tune) in the theater and thought this might be interesting. I started in winisd and played with all-pass parameters on the sealed sim until the system group delay matched with the ported sim. It's possible to get very close with just guess/checking (though it would be nice to write an algorithm to solve for this, particularly for active crossover design of multi-way speakers). I then put those biquad coefficients into a minidsp and ran before/after REW sweeps of the subs separately and together in-room. Theory matched practice pretty closely. The trouble is that, while the phase plots in REW when measured at the seat look fairly close (and much tighter than without the all-pass), when I turn on all subs and run a sweep, the frequency response is markedly worse (bumpy with deep funky null). I'm wondering if part of the problem is much of the phase shaping happens say <30 hz, and there it's more about room modes than first-arrival. Alternately, I've got one sealed sub near-field right up against the seat with the rest dispersed around the room at roughly 10-15 feet distance, and that might be further confusing things. As it is, everything sums ok, but I've always thought it could be better.
  16. I'm working on my signal chain and thought I could add a few data points here. I don't fancy sweeps, as I was using an oscilloscope to look for clipping/distorted waveforms. However, that saves me having to worry about my loopback measurements and a questionable soundcard. Here's the signal generator from REW out via hdmi into an Onkyo TX-NR809 receiver. screenshot of 5 hz: The yellow boxes at bottom are displaying period, frequency, Vpp, and Vrms. With AVR at maximum possible non-clipped output, I saw 22.5 Vpp (!) at the sub out. In the screenshot you can see that 5 hz was 20.2 Vpp, and also see the min values from 2.5 hz where it dipped to 15.3 Vpp. I make that 20 * log10(22.5 / 20.2) = roughly -1 dB at 5hz and 20 * log10(22.5 / 15.3) = roughly -3 dB at 2.5 hz. Interestingly, I could push the AVR crazy hot without clipping the output. The above measurements were taken with the master volume pegged at +18 and sub out at -2.5. Adding another half dB chops the waveform unmistakably. Because I assumed it would clip much sooner (my Onkyo TX 535 does), I had the sub out previously at -15 with lots of amp gain, since I regularly go above 0 master volume with music. It looks like I can get a bunch of SNR back. Unless I'm missing something and the AVR is not routing all the input signal to the sub preout. My AVR was reading multipcm in with REW running, but I couldn't tell exactly what channel(s) the signal generator widget in REW was outputting to. I'll next try to see if I can grab an actual full sweep into the scope from REW where I know I'm outputting on exclusively the sub out channel. That leads me to my next concern, which is that I don't clip my minidsp inputs. I can adjust my AVR down a bit to not exceed the balanced minidsp 4 Vrms max in, but I don't know if/what effect bass management has. More testing required... Also, I measured my Onkyo TX 535 upstairs and saw similar rolloff results --I think it was 1.5 dB down at 3 hz. Its preout was even hotter - up to 27.4 Vpp or 9.6 Vrms before clipping.
  17. I use https://shapeways.coma fair bit for prints. The quality is very high, but it's still FDM, so things usually have a slightly rough finish. They do a variety of materials including fiber reinforced (extra cost and requires a human to verify your model) and a bunch of metals (speakers still too inexpensive? Try printing the enclosures out of platinum!) It gets pricey fast however, and the maximum dimensions are typically small (e. g. < 30cm), but that's easy enough to check. All of their material pricing is on the site as well, without having to upload a model. All that said, I don't know why you'd want to print over using wood. Maybe to make complex curves/shapes?
  18. What's the passive crossover stuff for? I assumed it was all active. Is one of the woofers low-passed to avoid off-axis cancellation or something?
  19. Aw nuts. Yes I suppose bridged would be a problem. It's too bad differential probes are so expensive. Do those flukes have any kind of data out feature? I know my cheapy tekpower does. It's actually pretty decent. I think it captures 10 samples per second if you pick low precision: www.amazon.com/TP9605BT-Cellphone-Connection-Interfaced-Multimeter/dp/B00SGKR9FA
  20. It's part of the design of the scope to have ground terminals of all channels wired together and to the main earth ground. It would be unsafe not to. In the above image, you can see that the black side of probe 2 is on the other side of the load from the black of probe 1, which is your problem. There is huge potential between those two when they are on either side of the measured load. As to why this happens, the resistance on a single probe from + to - (where the scope measures it) should be around 1 megaohm, so it can measure high voltages without things smoking, but the grounds between two probes on separate channels are probably < 1 ohm (you can continuity from the outside of two BNC barrels on the scope with your fluke to confirm). In the image below, I reversed where you're clipping red and black on probe 1. I think that should keep your scope happy with the grounds always at the same potential (but I would carefully test starting at low output; I am not an EE), and you'll be able to capture live real-time data the way the good lord intended. You can also click Invert on any channel in the rigol to turn things back right-side up. While you're doing that, it would be neat to see THD at various levels from the amp. I know you can hit MATH -> FFT to view the full harmonics of a signal in db scale below the main graph, but I'm not sure if it spits out THD measurements. Probably. In any case, this setup would be worlds neater and easier than copying down fluke readings with pencil and paper.
  21. I imagine a 12K should be able to break-in just about anything you want.
  22. Re: break-in, I don't know how you are going to give them the necessary excursion * time without needing to re-drywall your house or have your neighbors move away. What you need to do is park those things near Xmax in free-air for a few days. So... have you tried wiring each cab with one driver out of phase, just for the purpose of breaking in? You could run all four cabs at once with alternate drivers canceling, and you'd have a whole lot of air whooshing and hopefully little net sound output. The cabs might do some dancing, and you'd want to first find the new shifted impedance max in that arrangement to avoid overcooking them (as they are coupled and not free-air), but besides that it sounds cake. Also you'd want to shoot some video of the whole arrangement huffing at war volume and post that here.
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