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SME

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Everything posted by SME

  1. My counter would be that the bandwidth is too large and too smooth to be a TH. The EQ might help with smoothing response, but what about overall bandwidth?
  2. Occasionally when watching movies with heavy ULF during the day time (often for critical evaluation), I notice motion in the corner of my eye coming from the flexing of the large, nearly floor-to-ceiling living room windows on my left wall. Yeah, the Hi Def Digest graph looks like it's probably "wrong", at least in terms of level normalization. Just comparing the peak levels between the two at 30 Hz, they are too far apart for lack of dialnorm compensation to be the only fault.
  3. Is that a sealed alignment or something else? It sure does drop like a rock below 20 Hz.
  4. The short answer is that different windows were likely used for each piece before averaging. There is no universal PvA. Different windows types and sizes change the emphasis of certain aspects of response.
  5. The design looks OK. The ports definitely exit close to the cabinet wall, but I think that will be OK as long as you don't go any closer. The wall will help lower the tune a bit more than the flared pipe alone would. However if that's your cut-sheet list, you are missing something very important: bracing. Bracing helps stiffen up the cabinet walls so that they don't flex as much which causes some bass to be lost and causes the enclosure to resonate at somewhat higher frequencies, coloring the sound. The bracing design is a little tricky being that you have to clear the driver as well as the vents while trying to avoid letting any panels have more than a 6-8" span unbraced. You might also want to build in some extra support for the vents, near the rear, being that they are quite long and presumably only attached at the front baffle. (You might also want to put foam or cloth between the tube and brace to avoid buzzing sounds if they vibrate against one another.) Ensuring no more than 6-8" of panel is unbraced is a good rough target, but often some compromise is necessary. Either way, some bracing is much better than none. You will also want to add a light amount pf absorptive material (polyfill / pillow stuffing is popular as are denim scraps) around the inside panels to try to reduce resonances in the air cavity, which can also transmit through the panels. After accounting for bracing, the cabinet volume will probably be a tad smaller, maybe 6.5 cuft (?) depending on design. That should still tune at around 20 Hz.
  6. Sorry I can't help there, but ... I love the playa pyro pachyderm! I did a couple burns, last time in '06. That year it felt like it was just a huge event, especially on Friday and Saturday nights when there were so many people everywhere who were so frelled up that I could have convinced myself I was in a raver zombie apocalypse. Also, I've been to places like Washington DC National Mall, NYC, and (gasp) Boston, and no other place has a greater proportion and variety of law enforcement agents than Burning Man. The event is a kind of a psychedelic fun house in the middle of an alien, post-apocalyptic police state planet. I understand the event is now roughly *twice as big* as it was then. It's amazing to me that it still functions, which is kind of a marvel of its own. Have fun with that car! Is there going to be a bar on top?
  7. I had the problem of spinning the bit too fast with *hand-held* routing in my speaker builds. I broke a couple 6 mm bits that way before I learned about RPMs and feed rates from the CNC forums. So I dropped my speed setting to "2" out of "6" (about 10k RPM I think) and sped up my manual feed rate. That helped a lot, but I still needed to make very shallow passes (like 3 mm or less) to avoid problems. That's what I did with my subs. Unfortunately the compression bit I used caused nasty tear-out because the tip of the bit is designed to be on the *other side* of the work-piece where it pushes chips up and towards the piece instead of away from it. So I think for my next project I will use a down-cut bit with very shallow passes and hopefully be able to keep it sharp long enough to use on more than one project. BB plywood just seems to be the worst enemy of cutters. Maybe the material I'm getting is just particularly nasty? It builds some damn good speakers and subs though.
  8. Hi @maxmercy, I'd like to request that you post the BEQs you did. You can post them in the private forum instead if you want. My wife has requested that we screen these films over her Xmas break, and I figure this would be a great time to evaluate your candidate BEQs. I can give honest feedback including sharing any tweaks I make myself.
  9. I'm sure that rings like crazy! And it's got only "3 Stars" dynamics to boot. I haven't seen that movie yet though. I think a lot of people have floor or wall resonances around 25 Hz, and so a track like that is likely to shake things like crazy, especially on vented subs with a similar tune. I've been mostly slacking a bit on movie watching lately, but I did watch "MI4:Ghost Protocol" tonight (for the first time) and thought the bass and overall sound design was very solid. I wish more movies had bass like that.
  10. My SP2-12000 is still on 120V, on the same 20A circuit as all the rest of my gear including: a 55" LCD TV, 2 Emotiva XPA-5 gen2s, a Motu 16A, a Denon 3313CI AVR, two (fairly low power) PCs, a BD player or PS3, and a couple routers. Loud movie bass passages with ULF lasting several seconds can be tickling the clip lights, but I've still never tripped the breaker or even heard any sign of distortion suggesting voltage sag. The mains run from the panel is only about 15 feet, which might make a difference. Also, my subs are crazy efficient at 30 Hz and up, so the amp doesn't have to pull much current unless there's a lot of ULF.
  11. Shall I take it as a positive indication that you aren't selling any of the other stuff?
  12. To add to this: Hornresp reports acoustic power output as a normalized value such that if radiation is omni-directional (throughout the solid radiation angle chosen for the simulation, e.g. 2-pi) it matches direct sound SPL that would be measured. However when there are directivity effects, it's normal SPL to be quite a bit higher than what Hornresp shows, simply because there may be a lot more sound radiating along the ground than say vertically above the sub. Another potential cause of higher output (albeit over a fairly narrow region) that I don't see mentioned is port resonance. The Hornresp simulation doesn't seem to show that, but the impedance and measurement data suggest that could be going on here also.
  13. Just to double check some things. The 7 cuft should be the *internal* volume left over after subtracting stuff like the internal bracing and *especially* the port. Also the driver takes up some space, though probably not more than 0.25 cuft or so. The port dividers reduce the cross-sectional area of the port slightly, so you are right that this will drop the tuning a bit more than without them. It shouldn't be hard to calculate how much area you lose ... For 2 dividers of 3/4" thickness in a 3" tall slot port you lose (2 * 3/4 * 3) = 4.5 in^2. It's that simple. There is one more important consideration worth mentioning. Vents have a "pipe resonance" (like an organ pipe) at a certain frequency depending on its length. Your 40" pipe is likely to resonate around 150 Hz. That's a bit lower than we'd like but is "do-able", especially if you are crossing over at 80 Hz and not higher. Certainly we wouldn't want a vent that's any longer. As you can surmise when trying to make a low-tuned cabinet as small as possible, there is an inherent trade-off between getting vent area large enough to not overload at high levels while keeping the length short enough to fit into the cabinet and to avoid making the pipe resonance too low. This is a trade-off that vented system designers have to routinely make, and it gets worse when either tuning lower or trying to use a driver with more output. Your final design looks pretty good as far as balancing trade-offs. For the length, area, and volume you specified, Hornresp indicates a tune of around 22.5 Hz. A 3rd order high pass filter at 18 Hz appears to control excursion well and gives you a response that's roughly -6 dB at 20 Hz and -10 dB at 18 Hz. Depending on what your room is doing, you may get useful output quite a bit lower than that. If you plug a vent and use the same HPF, you lower the tune to 18.5 Hz, which gets you -6 dB at 17.5 Hz and -10 dB at 15 Hz. Again, your room may help things get a bit lower. In both cases, vent velocities are very reasonable, so you're not likely to see much compression. While plugging two vents will technically get you a 13 Hz tune, you don't really gain any usable output or extension this way, so I wouldn't recommend using it this way. Anyway, good luck with the build! We look forward to hearing of your progress.
  14. I get an "Internal Server Error 500" when trying to load the link you posted. The link may only work for you. That design should work decently, though depending on the type of ports used (like flare characteristics if any), the length you gave might tune a bit lower than 20 Hz. Also keep in mind that the vent area limits how much air can be moved which limits how much output you can get at the bottom before the vent overloads. And it's home theater content that's more likely to push the sub to its limits. One thing I'd consider changing is to add a 3rd 4" vent, increasing the lengths and cabinet depth accordingly. I'm not sure you can do this with the Precision Ports I posted a link to, earlier in the post, and if you can, it'd probably be a (maybe ugly) asymmetrical arrangement. Another option is to do a slot vent, which can be built right into the cabinet. It should be braced every ~8" or so, which can also be used to the vent into individual chambers that can be plugged. See for example the Skhorn and Skram designs posted here. Do you think you'd be able to fit a second one of these subs somewhere? Subs tend to sound better with at least two in different parts of the room.
  15. Oh, I see. I guess I've always interpreted "power compression" to refer to *any* kind of compression that occurs in the speaker. Maybe that's not really the correct usage of the term. What I meant was output compression in general.
  16. Ideally dialnorm assures similar dialog loudness at the same master volume, regardless of the dynamics or crest factor of the particular mix. The way it's supposed to work is that the final soundtrack is measured for loudness (which takes into account spectral balance factors, to an extent) using a standardized method, for example LKFS. Then the dial-norm offset is set based on where the loudness falls vs. a reference value, which I believe is -31 LKFS. So soundtracks with -31 LKFS, -27 LKFS, and -24 LKFS, should respectively have dialnorm offsets of 0 dB, -4 dB, and -7 dB, and they should all sound about as loud when played at the same master volume, even though the latter example of -24 LKFS is probably a lot less dynamic than the first. Of course all this assumes consistency between different titles in the loudness measurement method and setting of the metadata on the soundtracks, which still doesn't happen. In the old days of DVDs, the DD tracks on them very often had a "-4 dB" offset, and I believe this was because that was the default value. (Some titles still came with other values.) For BD, a lot of tracks are DTS-HD, and those encoders probably default to a "0" offset. The Dolby TrueHD tracks are more likely to use a non-zero offset, but I believe this is less consistent than it was for DVD DD tracks. So the consequence of inconsistent use of the dialnorm offset parameter actually has the opposite of the intended consequence.
  17. Air flow is very non-linear versus velocity, and losses increase very quickly at high velocities. A rough rule of thumb is that you compress 1 dB for every 10 m/s (also depending somewhat on the flow area and overall shape of the passage) , but there is also a saturation point, maybe in the 30-50 m/s range (again depending on details), where output stops increasing altogether. Often by the time you hear chuffing, you're already at that point.
  18. The problem with too much vent velocity is not just chuffing but also power compression, and substantial compression can set in well before the chuffing becomes audible.
  19. It's like the age-old Information Technology help desk joke (based on a true story), "have you tried turning it off and on again"? Except for movie soundtracks it's: "have you tried adjusting the volume control to the loudness you want?" An entire Loudness War has been fought over --- catering to the whims of the volume-control-challenged masses. Of course it doesn''t help people when the soundtracks aren't the slightest bit consistent in their setting of the dialnorm metadata. So it's like the worst of both worlds.
  20. Hi @peniku8. That's not really necessary unless you really want to do it yourself. I expect that DC and ULF noise can be introduced in an analog tape device under the right circumstances but that it's often prevented or minimized by the presence of blocking capacitors somewhere between the tape output and digital input. But wait, capacitors work essentially like a filter, one which may not cut the noise as much as a desired BEQ boosts it. OTOH, running a whole 7.1+ channel mix through a tape machine with only 2 channels might not work well. At the least it'd be tedious (de-mux, do 4 runs, time-align, and re-mux) but any wow and flutter are really going to screw up the time alignment between channel pairs. Perhaps an even better explanation is that it's simply a digital plugin in the mastering chain that's contributing the noise. Maybe it's a tape machine simulation plug-in. And here again, it might include a built-in filter to reduce the DC noise but which cuts the noise less than the desired BEQ boosts it. You're right that we'll probably never know for certain, but it's clear to me now that this noise could have easily been introduced (probably unintentionally) into the "mastering" chain. If that were the case, then the Atmos mix was probably not derived from the DTS mix but is a fork of an earlier version of that mix. Perhaps they are independent home mixes derived from the same original cinema mix?
  21. My "Like" is for this part of your post. Incompetent is definitely the wrong word and wrong concept, really. Cinema mixers are generally very competent because if they aren't, they don't get new contracts. But competence really refers to the ability to fulfill a particular job function, which isn't really saying that they know everything to know about the craft. In fact, there is probably a lot about this craft that is as yet unknown and a lot that is "known" that's actually just wrong. The issue I have with some of the recent Marvel releases is dynamics crushed to the point that it's less dynamic than an old analog TV program and doing so in a way that breaks a lot of the artistic integrity of the presentation. It's one thing if a particular scene isn't as BIG as it would have been without dynamics reduction, but another entirely when the dialog in a heated argument gets quieter just because more than one person is talking at once. Or using peak limiting in a way that snuffs transient sounds out of existence. For the most part though, I think mixers (and the other sound people) do a decent job considering the ridiculous time pressure they endure and the complexity of these soundtracks. And this is especially true given the spectral balance problems resulting form X-curve calibration. It's just tragic that this quirk affects the creation of the art in the ways it does. If they had neutral systems and the cinemas were neutral too, I think it would have a huge positive impact on the quality of the overall art. It'd definitely bring us closer to the performers, for better or worse. Exactly. There's no way the DTS and Atmos tracks were each created from scratch. One is likely derivative of the other or at least derived from a similar original. So the fact that one has this noise and the other doens't is puzzling. I know you can apply de-noising to a complete mix, but I doubt anyone does that unless they are trying to restore and re-master old content or something. Maybe that's it, and I just need to think more flexibly. I think @maxmercy sees ULF noise (or really "DC noise", under 3-5 Hz) quite a bit in movie soundtracks, more often in older tracks. However keep in mind that this noise isn't really a problem except when doing BEQ because the BEQ boosts the noise along with the desired content. At least some of the time, the "noise" may simply be due to tracks with the DC noise not being filtered any more steeply than the tracks with desired ULF content. Curiously though, some soundtracks have this noise all over and others don't have it at all. It's totally hit-or-miss. ==================== As a separate response to this overall discussion, I would caution people not to read too much into the PvA data when trying to understand why different soundtracks sound different. Certainly differences in spectral balance are possible, and seemingly minor spectral balance differences (like 1 dB) can actually have a major impact on the perceived sound. Perception is very relative in terms of what's happening at different frequencies, and you can't really see what is happening, spectrally, with the individual effects by looking at the PvA. Keep in mind too that dynamics processing might be quite different between tracks, which likely explains why the PvAs are not exactly consistent in the fine details. Yes, both tracks (and the cinema track too, if it's different) are likely seeing plenty of dynamics processing, which can affect how the sound is perceived. Also, literal dynamics is only one parameter that affects perceived dynamics. Spectral balance affects (e.g. momentary shifts in broad spectral balance and "saturation" effects) can give very different impressions of dynamics even with the SPL pegged to the same number. And the consequences of these differences may all be expressed different on different systems. So the situation is way more complicated than can be depicted with a PvA or even spectrograms, though sometimes these tools reveal interesting things. They are useful tools, but they can only explain so much.
  22. My wild guess is that the "noise" was contributed by a mastering plugin, maybe a subharmonic synth which was different or configured differently for the DTS vs. Atmos mix. Let me turn your question around. If the noise was actually deleted from the Atmos track rather than merely filtered, how and why was this done? One possible answer to the "how" is a de-noising plugin, but my understanding is that such plugins are not fire-and-forget and the results need to be actively monitored and settings tweaked to get the best results, which these engineers probably could not do for the lowest frequencies. Maybe they just ran it blindly? In which case, I'd expect more destruction of non-noise content in the de-noised version. But why? Why would they apply a de-noising plugin just for stuff at the very very bottom of the spectrum? If one is worry about that noise "causing problems" in playback devices, a simple high pass filter will do just fine to prevent that, and in fact both tracks are HPFed already. I guess one possibility is that de-noising was applied to the entire track, and the removal of extreme LF noise happened as a consequence. Still, even with carefully hand-tuned settings, this de-noising is likely to collaterally damage some of the original content, and I don't see how it would help the soundtrack in any way, except maybe in some of the dialog recordings which can and do come with unwanted hiss sometimes. Hmm, maybe the de-noising *is* selectively applied to dialog samples with hiss, and maybe the ULF noise is part of *those* samples. (I really doubt that though being that almost all dialog probably gets high-passed at 60-120 Hz or so.) I know that on the video side, de-noising is pretty standard for home release (and probably cinema too) where it's used to try to scrub out film grain, and in this application, it also tends to degrade the actual content and can contribute new artifacts also. So maybe the Atmos track has been de-noised, in which case it's possible that the DTS version retains more content and has fewer artifacts, making it the "better track" if one is not bothered by the noise. Either way, the fact that such noise appears on one track and not the other poses interesting questions. Assessment of sound quality via purely objective means is probably very difficult if not impossible. That's a big reason why humans are involved in the mix process, and also why it's essential for the humans doing the work to listen with a monitoring system that's as neutral and accurate as possible. It may be possible for a machine learning algorithm, trained using listener preferences, to provide an assessment of sound quality. However, I would not expect this to work consistently well for a number of reasons. It might also be possible to write an algorithm specifically to detect loss of information due to lossy encoding at reduced bit-rate, but this is a very specific case. And furthermore, quantifying how much information was lost does not tell us the impact of that loss on sound quality. The subjective sound quality impact of these losses could be estimated using the same psychoacoustic models used for the lossy encoding itself, but this is still just educated guessing and for what purpose? Most media can be obtained in a lossless format or at least a lossy format with a high enough bit-rate that the subjective impact is going to be very subtle. I did recently see evidence that some, maybe all Dolby AC3 (i.e. "Dolby Digital") encoded tracks are low-pass filtered at 20 kHz, and for a lot of listeners and systems, this could have as much or more impact on the sound quality as the lossy encoding. So I guess LPFing at the top is one thing that can be objectively assessed.
  23. Glad your wife didn't veto your purchase after what she had to "suffer" through, lol. Keep in mind that a subwoofer in a car get a lot louder than it does in a house where there's a lot more space to fill. If building a vented sub, the tuning frequency is very important because the sub won't play much content below the tuning frequency. So at 29 Hz, you'll be missing out on most of the content below. If you want to *just* get to 20 Hz, maybe aim for a tune of like 22 Hz. Also, I think a 4" diameter port is a bit small. You need enough area to avoid chuffing and compression at high output. How much you need depends on what the overall design looks like, but for a 15" woofer like that, I think you'll want at least 2 x 4" pipes. Do you want to tune lower? The trade-off is that you need to make the vent longer or the box larger, keeping the vent area the same. Working from the numbers you posted, a pair of 4" x 20" pipes ought to get you a tune around 22.5 Hz. Actually, it'll be lower for that particular cabinet if the pipe exit is near the wall because the wall effectively extends it. Alternatively, you can make the cabinet larger, and this will also help boost the output around the tuning frequency. You may want to look at simulations though to see what you're ultimately likely to end up with. If you do decide to use one or more 4" pipes, this product is real hard to beat if it works for your cabinet design. The flares help a lot with improving performance at high output. They are also very easy to install. If you decide to use them, I recommend using the formula included in the manual to calculate the length because it takes into account the flares properly. And also keep in mind again that if the exit is near the wall, it'll tune lower than expected. (Don't let them exit less than ~3-4" from the back to you don't constrict the flow there too much.) I think the XLS 1002 amp is a good choice as it includes DSP which you need for a vented design to apply a high pass filter to protect the woofer from frequencies it can't play.
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