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Everything posted by SME

  1. The biggest advantages for multiple subs arise when the different subs have overlapping frequency response. However, the full benefits can only be realized when the different subs have similar output capability. The advantages in this case are: increased output over one sub alone, and smoother in-room response. When the two subs are mismatched in output capability, then you can still achieve smoother in-room response but only by substantially wasting the capability of the stronger subs. That's because the smoother response requires the different subs to be run at approximately the same level, and the weaker sub will reach its limits at much lower level than the strong sub. There are some minor benefits to your usage of different subs for different ranges. For one, the different subs can be optimized to play their respective ranges. For example, the horn is very good above 40 Hz, and you can get away with building deep bass subs using drivers that don't do mid-bass well. You also will gain headroom and see less distortion when reproducing wide-band sub-bass signals. However, as with multiple subs handling an overlapping range of frequencies, the benefits are greatest when the output capability of the subs handling each range is approximately matched. Though how they should be matched depends a bit on the content you play and on room gain effects. For example a lot of music doesn't extend much below 40 Hz, so the horn may be able to handle that very well on its own and reach quite high levels. For home theater, however, a lot more bass extends into the 30s, 20s, or lower. It'll likely take a lot of output to keep up with the horn. Of course, that may not be a big deal if you don't ever push the levels that high. On another note, integrating subs of different types can be very tricky. You will want to do a lot of measurements and experiment a lot with different distance and possibly EQ settings to get the smoothest integrated response.
  2. Yeah, manual is best. IF you are going to EQ the LaScala by hand, that's still more reason to not use Audyssey because it will basically undo much of your work. And yes, there is quite a learning curve for doing it manually, but I'd argue it's well worth it. IMO, the performance of good equipment is held back considerably without good configuration. The speaker stand may be OK for sub measurement, provided it holds the mic well in an upright position. However, the tripod with boom arm is best because anything near the mic influences the measurement, especially with high frequencies. In the crossover region, you have two transducers in different locations producing sound. If the sound from each source arrives at different times, then there will be frequency-dependent phase shift between them. The phase shift will cause them to not combine coherently at all frequencies. You don't need to do a full-range sweep, but your sweep should cover at least 1 to 1.5 octaves above and below the crossover point. It's helpful to label each sweep with the distance and/or crossover point you used and view them all together to compare.
  3. I have experience with Audyssey, and while one of its key features is automatic setup of delays, my experience is that this function performs very poorly. When using Audyssey, I almost always achieved better results by manually adjusting the distances from what Audyssey gave. I also do not like how Audyssey re-shapes frequency response. It aims for a flat in-room target, which leads to a poor tonal balance including thin sound and weak bass. Setting distance for the sub optimally can be a bit tricky, but measurements are your friend. Do you have a good tripod for the measurement mic? I strongly recommend investing in one. Setup the tripod and mic at ear-level at the MLP and then run a series of sweeps, varying the sub delay/distance around 1 ms with each sweep. Look for the delay that gives you the smoothest frequency response around the crossover frequency that you choose. Once you've found the best distance, you can also try to incrementally adjust the crossover for an even smoother response. Note that the optimal delay may also vary depending on the crossover.
  4. Sound card and measurement mic upgrade

    Reversed polarity is pretty common with a variety of electronic devices and is easy to fix in measurements. My Motu 16A reverses polarity. I don't think it's a big deal, and I'm not aware of any empirical evidence that polarity inversion is audible at all, as long the polarity is consistent for all output transducers. The exception would be for crossovers that are designed to be inverted. Though this does highlight a more general problem when using systems with speakers of different types, even if they are from the same product line. The issue is not really timbre matching but rather phase matching. When the crossovers are not identical, there will be phase mismatch over at least some of the range, and this definitely is audible. I have discovered that multiple speaker integration in a system with biquad EQ capability can be improved by using all-pass filters.
  5. I can agree with a lot of this. I enjoyed the movie a lot, despite it being a narrative train-wreck. I really appreciate the story they were trying to tell, but they did a very poor telling of it. I think there was 30-40 minutes (or maybe even more) of footage closer to the beginning that should have hit the editing floor. In its place, they should have devoted a lot more time to developing the characters. Instead, the plot felt very forced with key character developments being implied "after the fact". The scripting was also sub-par with many lines and much of the comic relief being awkward and out-of-character. I'm really surprised now to see critics liking it so much. My opinions on movies usually side with the critics but not this time. Still, it was definitely worth watching, and I will watch it and enjoy it again, but I cannot call this a good movie for the reasons stated above. It is a shame because it could have been so much more. I hope this director does not come back to the franchise As for the sound, I agree that it seemed to have a lot in common with the previous movies. I didn't get the impression that 30 Hz was emphasized, but it's hard to say for sure without a good reference. The subs in the cinema I went to were decent and delivered balanced LFE with a fair bit of chest sensation and even some signs of extension into the 20s. Other than that, the tonal balance of the presentation was definitely mediocre. The speakers themselves seemed to be nice and smooth. I think they were fairly new model JBLs. The surrounds, which were visible, had nicely contoured waveguides matched to what looked like 10" woofers. The sound was quite thin overall, especially in the 250-500 Hz octave, and it seemed to emphasize 1.5-2kHz a bit too much while also sounding too rolled off at the top. That's completely consistent with what I'd expect in an X-curve calibrated cinema, and I reckon the re-recording mixers applied EQ quite sparingly on the dub-stage. Toward the end of the movie where things got a tad louder, the ~1.5-2k emphasis was slightly fatiguing. Oddly enough, the surrounds sounded significantly better, which made me glad that they got used a lot. I think they may have mixed the score a lot more into the surrounds this time than in TFA. The funny thing is that my wife and my sister and her husband could easily tell the inferior tonal balance as well. We all watched TFA on my system last Sunday. I'm real stoked to get this one home and hear it with some BEQ. I'm sure the sound will really come alive then. I have a feeling that it will have better bass than TFA did. I dunno if it will compete with Rogue One + BEQ though. That one was really awesome with the stupid 30 Hz hump fixed.
  6. The Bass EQ for Movies Thread

    I recently tightened down my latest system EQ config, including a complete overhaul of the surrounds that delivers stronger mid-bass and more bass overall. It's nice and punchy for music, without compromising deep bass, where it does show up. I did some testing with music mixed to mono and sent to the center and each surround to confirm that the mid-bass retained its punch on each channel. Over the last few days, I've been testing with movies. The opening bits of GOTG2+BEQ are even better than when I watched it before. The kick drum on the music tracks has life! Tonight I watched "Star Wars: TFA" again with BEQ. I tried with the full mid-bass boost in the BEQ, but backed the PEQ gains down to only +2 dB per channel and added about +0.75 dB @ and below 30 Hz . With the full +4, the mid-bass boost overpowered and killed the deep bass, but it obviously lacked punch at only +0. The extra +0.75 dB down low seemed to get things just right. There is a great balance of shaking effect and lots of chest thump. I can't guarantee these adjustments will do right for everyone else being that they are quite small. In any case, the movie was a fun ride. It was the first time my sister and her husband had heard my system since I got the new speakers and subs. They were smiling pretty big when it was over. Now we're all properly ruined before we go to see "The Last Jedi" at a cinema next weekend.
  7. It looks how I would expect with a Q around 0.7 or so. Every sealed sub essentially rolls off naturally at 12 dB/octave in the low frequency limit, without EQ (and of course, more voltage and power demanded by said EQ). Excursion always increases 12 dB/octave vs. frequency for the same SPL output. So once you go low enough, excursion basically remains the same without EQ. Of course, the roll-off will likely be much less your room. I gather that's a single driver? You were going to use 3 of them? If so, you are hopefully modeling the single driver in 1/3 the total volume. Now that you've done that, try modeling other drivers in a similar volume and similar voltage. (See my earlier posts for how to calculate the "wattage" that WinISD Pro expects from actual voltage, and visa versa.)
  8. The main thing that WinISD Pro and other simulation programs will tell you is the best-case max output output from the sub + amp system, at 1 meter ground plane. It's merely best-case because a driver that's pushed near its excursion limits will start to compress and deliver less output. That's not really easy to model, so it can help if the driver has actually been measured in a cabinet, like Josh Ricci does here. WinISD Pro can also help you compare different drivers and different cabinet sizes. As for how much Xmax is enough, well that ultimately depends on how much output you want and whether you'll give it enough amp power to use that excursion. And unfortunately, the ultimate output you get depends substantially on what your room is doing. There're just too many variables to give you a clear cut answer. If building a ported sub, a lower tune will allow for lower extension, of course, but at the cost of output above the tuning frequency. Lower tunes also generally require larger (and heavier) cabinets. So there are trade-offs. If you know you're going to have more than enough output and have no space or weight constraints, then definitely go lower. Some people build huge cabinets tuned as low as 12 Hz. (A few aim even lower than that.) The B&C21SW152 is a solid driver choice for a ported box. Better yet, lots of people have already built ported systems around it, so you can avoid a lot of design hassle by looking at work that others have done on here and in the AVSForum DIY section.
  9. No, the X curve is not applied to the mix by default or in any kind of automated fashion. Instead, the X-curve imparts a tonal shift that affects what the re-recording mixers hear and influences the EQ they apply on the dub-stage. The mixers are likely to boost the highs and lows to compensate for what they hear. What you really should be saying is: room A acoustics =/= room B acoustics. Size is only one of many room variables, and in fact, listening distance and speaker dispersion pattern are probably at least as important. In some ways, this gets us lost among many details be especially important here. A crucial issue is to distinguish between the effect the above variables have on *perception* from the effect these variables have on the *metric* used in the calibration process. Ideally, the calibration process would rely on a metric that is 100% consistent with perception. Power-averaged response, which is the metric used for X-curve calibration, is not very consistent with perception at all. It is, however, strongly influenced by room acoustics. I'm assuming your response is with regard to the fact that music production doesn't rely on standards? Therein lies a real irony about the cinema standards. It is a case of "no standards" being better than "bad standards". The lack of standards in music forces engineers to adhere to established precedent, which serves as an informal standard. They listen to recordings they consider to be good references and mix and master to achieve approximate parity with those references. Dr. Toole calls this "The Circle of Confusion" for good reason, but in fact, I'd argue that the situation with cinema is worse. That's because, while on the one hand, the cinema standards fail to achieve consistency between different playback systems, the engineers trust in the accuracy of their "calibrated" systems and mostly disregard precedent when making mixing decisions. They simply mix to "what sounds best" to them and assume it will sound like that on other properly calibrated systems. Now to be fair, not all cinema engineers are mixing like I describe above. Through their experience, they have surely noticed that different dub stages sound different and have learned to compensate their mixing technique accordingly with the aim of achieving better results in a wider range of venues. Furthermore, the X-curve standard was actually a decent even if imperfect 1970s-era solution to a very real problem: high frequency absorption of screens is variable, and the best calibration tools that were available at that time relied on power-averaged response analysis of pink noise signals. It's just that today, we have much more capable measurement methods and a much better understanding of perceptual issues. Along those lines, I disagree that Dr. Toole's recommendation (see the second of the two above papers) for calibrating in-room magnitude-smoothed response to a standardized sloped target is the optimal solution, but I believe it'd be a big improvement over the X curve. His recommendation would effectively free up an extra 4-6 dB of headroom per screen channel in cinema soundtracks and would probably lead to a big improvement in the bass for cinema presentations overall. (I'll refrain from giving a detailed justification for this final point unless someone wants me to.) You're right. I didn't have to expand into great deal. I'm just a big geek, you know. And I'm actually quite excited because I think I've finally mostly unraveled a lot of things about film audio that were previously confusing to me and still are confusing to many others. I stand by my statements about the X-curve standard inhibiting headroom on cinema soundtracks, but in time, this is becoming a lot less relevant for those of us who mostly care about home theater, because home mixes are becoming more and more common and are improving in quality. I would not be surprised in the least if "Dunkirk" is a clear exception to this trend. It's probably a straight-up cinema mix and a very loud one at that. Which is still fine by me because I'll re-EQ it as needed when I get a-hold of it. The X-curve is still a big problem in cinema, and I think it's hurting the industry, even if they don't realize it or won't admit it. Dr. Toole has pointed out that many cinemas are hosting music and sports events and corporate video conferences, and stuff like that in order to bring in more revenue, but because they are calibrated on X-curve, all that other audio sounds like poo. That can't be good for their bottom line, and it's the kind of thing that customer satisfaction surveys aren't likely to reveal, being that the influence of audio quality is so unconscious. FWIW, you're like one of the least "asshole" kind of people on these parts, which is why it's kind of funny the way you responded to me. Often that kind of thing pisses me off, but I don't care at this point because I know you and because it doesn't matter that much anyway. Part of my confidence regarding the X-curve is that I can clearly hear it. I'm routinely identifying cinema mixes and re-EQing them to sound better. Ahh yes, so now you think I'm blabbering in Audiophilese? "I can hear the difference man! This will totally transform your audio experience for the better." OK fine, but consider that I really suck at understanding dialog in films. Like, my ears aren't golden at all but are tarnished, maybe even rust colored, right? So when I apply re-EQ and dialog that was shouty and muddy and almost impossible to follow suddenly becomes clear and intelligible, I take note. That's what I'm talking about here. If you'd like, name some titles, hopefully at least one of which is in my library. I'll put it in and try to identify if it's a cinema track that will benefit from re-EQ, I'll play around with it and then publish some PEQs to try to see if it cleans up for you. Is it worth a try? Otherwise, come visit me here in Denver and hear for yourself.
  10. Bulding the Room2 listening room

    I'll concede on this point. Though some domes are quite capable, and if they aren't crossed too low, you have to push things pretty high to reveal the difference at typical domestic listening distances. I agree they would hear differences, but not just in the soundstage. Really, any substantial differences in linear frequency response, on and off axis are likely to be heard.
  11. Here's a recent paper documenting B-chain cinema measurements and attempting to assess the side-effects of X-curve calibration: https://www.smpte.org/sites/default/files/SMPTE TC-25CSS-B CHAIN FREQUENCY AND TEMPORAL RESPONSE ANALYSIS OF THEATRES AND DUBBING STAGES 1 Oct 2014.pdf Dr. Toole was involved in the work above. Here is Dr. Toole's follow-up paper, which suggests a way forward for a more reliable, universal calibration method: http://www.aes.org/e-lib/browse.cfm?elib=17839 Unfortunately, both of these papers are significantly flawed, and I admit I've been slacking as far as communicating with Dr. Toole to explain the flaws. The major issue is that all the "frequency response" measurements rely on magnitude smoothing, which even at 1/48th octave resolution has side-effects that are unexpected to the people doing this work. As noted in my previous post, magnitude-smoothed response measurements are very different from power response or power-smoothed measurements (which the X-curve relies on), and I am of the opinion that neither consistently correlates with tonal balance perception. Hence, the notion that calibration of magnitude-smoothed frequency response in all cinemas and/or all homes to a single target curve (even if it's different for cinema vs. homes) will achieve translation is fundamentally flawed. I have been working on a solution to this problem using complex-smoothing, i.e. FDW, but it is still a work in progress and has considerable limitations. The method requires measurements at multiple locations, even when the goal is to optimize sound at a single seat. That's because reflected sound arrivals do influence perception. To achieve true single-seat optimization, I would have to develop more sophisticated time-frequency methods, and to get it right, I'd probably need to do proper psychoacoustic studies for which I don't have the resources to do right. I don't know. I haven't even taken the first steps down that path. After my various refinements and tweaks, I am getting superb results using my FDW method, but it is tedious and probably error prone for many situations. I still have to make some manual judgments and "fudges" to deal with interference issues involving both my center and surrounds. It'd be seriously awesome if someone would pay me to actually be solving these problems. That'll probably happen ... in my dreams, heheh.
  12. Do double-check with your friend, but I'm quite certain that calibration to the X-curve is still standard for cinema. And that is true despite better screen materials and compression driver tech. Yes, it absolutely does kill headroom, effectively, when compared to how things work for music and "home" mixes.. The 4-6 dB figure is a rough estimate on my part. The cinema basically calibrates for a flat power response, even though a flat speaker under similar circumstances is likely to develop an in-room response with much more tilt. This approach is flawed from the outset because research strongly supports flat direct sound response as preferred over flat power response. While music production does not rely on any standards, the established precedent for music is flat direct sound response, and I am convinced that targeting flat direct sound response leads to better "translation" as well. For treble, a 1 dB/octave tilt is likely to be typical for a flat direct sound speaker (that is, flat after compensating for screen effects), so the X curve is tending to attenuate the top an extra 2 dB/octave above 2 kHz and even more above 8-10 kHz. That's lost headroom because the content in the soundtrack is boosted to compensate. However, there is attenuation at the bottom too, and it is has more severe consequences on headroom. Research suggests that even in fairly dead cinemas, direct-sound-flat speakers exhibit a substantial power response rise, starting below 500 Hz or so and rising more rapidly below 200-300 Hz. This probably occurs in part because of the relatively poor bass absorption of typical cinema room treatments, but perhaps more important is the considerable drop in directivity exhibited by almost all cinema speakers (and consumer speakers too) in that region. Unfortunately, the amount of attenuation during calibration likely varies a lot more than at the top. In fact, evidence suggests that front left/right vs. center vs. surround speakers in the same venue often sound different after calibration. As a consequence, cinema mixes translate poorly even to other cinemas. Cinema sound will never be "great" as long as this broken standard remains in wide use. So because of the attenuation performed during calibration, the low frequencies are boosted on the track also, and unfortunately the amount, center frequenc(ies), and shape of the boost(s) are inconsistent from track to track. Based on the measurement data I've seen as well as what "sounds right" to me when I do my re-EQing, I'd guess that low-shelf boosts of 3-5 dB, centered between 200-300 Hz, are probably common. So that's 3-5 dB lost headroom for bass where the majority of the energy of a soundtrack lives. Additional boosts centered at lower frequencies may also be introduced, but I believe those are less common and less likely to impact the quality of the track anyway. The stuff in the 200-300 Hz range has a big impact on the quality of the dialog, so this adjustment is almost always made, in conjunction with the adjustments made to the highs. Given what I describe, I would argue that 4-6 dB is a totally reasonable figure for lost headroom. When that same track is remixed "for the home", it is remixed on a system that lacks attenuation on the high end and is less likely to be attenuated on the low end either. The system is likely to use either quality flat-direct-sound monitors as-is (with bass response problems being address primarily through acoustic treatment instead) or calibrated to a curve more similar to Harman's recommendations, i.e. with a significant slope. The level calibration is done using pink noise band-limited to 500-2kHz, allowing the rest of the power response to either "fall where it may" or adhere to a non-flat target. (Actually, it's usually magnitude-smoothed response not power response that's fit to a target. Sadly, few people in this business realize that the difference between these two is significant.) Argh, see what you made me do? I wish I could explain it more easily. The gist is that the X-curve standard is flawed and leads to somewhat consistent attenuation of the highs and very inconsistent attenuation of the lows during calibration, and the compensation performed during the re-recording mix costs headroom.
  13. That's not important. With EQ, you can reshape the frequency response to be practically whatever you want. However, this has no effect on the ultimate capabilities of the sub system. If you need (for example) a 30 dB boost at 10 Hz to make it flat there, you most certainly can do it with EQ. However, as soon as you play anything remotely loud at 10 Hz, the amps are going to clip or the sub is going to bottom. The point of the simulation is to see the max capability of the sub, with or without EQ. The EQ feature in the simulation is likely to mislead you as to the sub's capabilities. Sorry, I wrote a typo in my post. I was trying to say that 130 dB SPL down to 30 Hz is not unreasonable. Likewise for 120 dB or higher @ 15-20 Hz. Those numbers may appear to be extreme, but they aren't far off when reproducing many movie soundtracks at a spirited / reference level. For a ported sub that's intended to extend low, high Xmax remains important, but motor force becomes more important than for sealed in a large volume. Investing in pricier drivers with a bigger motor is usually worth it.
  14. Bulding the Room2 listening room

    Yeah, and I'd bet that the majority of listeners, whether audiophile or not, could not reliably discriminate horn vs. dome based systems after listening in a room blind-folded. I'd love to be able to do this kind of experiment to see the looks of surprise if nothing else.
  15. Well that settles it. When it's time to watch Dunkirk, I'm only going to play with the surround channels turned on. I have to say I hate clipping a lot less now that I am listening to a more tonally balanced/neutral presentation. That's not something Audyssey delivered at all, and it's also not something one gets without tweaks when listening to a cinema soundtrack vs. a quality home remix. In both cases, the highs are overemphasized, which makes clipping truly dreadful. Cinema tracks are mixed in an environment with the X-curve applied, and most of the time, those tracks get brightened to counteract this roll-off. The X-curve also suppresses low frequencies and bass, so that part of the track often gets boosted too. Between that, and the fact that pink noise energy accumulates in large, cinema size spaces and inflates the calibration measurements vs. loudness means that the practical headroom available on a cinema track is nowhere near as much as one would think. The X-curve effects alone probably kill 4-6 dB of headroom. Taking into account room size effects, and for all practical purposes, a cinema track has only a little more headroom, per channel, than typical music releases. On the plus side, cinema mixes do have more channels plus LFE to work with. Now that I have a grasp of the real problems, I'd like to see the X-curve standard abolished some day. We'll see if that ever happens though.
  16. Newbie with 26 questions

    If you are capable and willing to invest the time to research and build, DIY is a great idea. Definitely check out the DIY section on AVSForum as lots of people document their efforts there. Some useful info to have is what kind of space you are working with, what your listening preferences are, and what your budget is. Being new to this, you probably have little idea of what a capable system can do. I recommend aiming well above the capability you think you'll need, for two reasons: First of all, systems often sound "too loud" simply because they are being pushed too close to their limits. It's usually a good idea to have a bit more capability than you actually use so you avoid even approaching those limits, where things can begin to sound harsh. Second, once you experience the sound from a clean system with the volume turned up more, you may find you like turning it up higher than you thought you would. When choosing speakers for playing movies, a common guideline is to choose something that can play at cinema reference level with ease. Even though you probably won't play that high in practice (in part because cinema reference is actually a lot louder in a home than in a theater), the extra headroom will still be beneficial if you decide to use some kind of EQ (including room EQ) or for supporting boosted bass levels in general. Most speakers run out of gas in the bass before the mids and highs, and it's the bass that tends to be the most demanding part of the soundtrack. For example, I typically listen around 5 dB below reference, but the bass response of my speakers and subs is boosted more than enough to make up the difference. Anyway, good luck on AVSForum, and feel free to ask more questions here. You'll probably get more responses though if they have to do with subs.
  17. Bulding the Room2 listening room

    Nice. It makes me recall the conversation I had last time I visited my doctor, who had heard from my wife that I was doing a lot of work in audio. Our conversation went kind of like this: Dr: "So what kind of turntable do you recommend?" Me: "I don't really listen to vinyl and haven't researched the options, so I don't really have an opinion." Dr: "Really? You only listen to digital? I have a friend who built a vinyl only system. I like it's sound. It's so sweet." Me: "There are things to like about the sound of vinyl, but it's possible to capture most of that 'sound' as part of a digital recording, too, if you prefer. The other good thing about vinyl is that music masters are often better for vinyl than for CD, not because of it's technical superiority but because the medium is harder to abuse for loudness." Dr: "Hmm. So then what kind of DAC do you prefer to use?" Me: "I have no strong opinion either way on DACs. Most DACs are made very competently and are unlikely to significantly contribute to sound quality." Dr: "Really? Hmm. Well, what kind of amps do you recommend? I use Parasound amps." Me: "Again, most electronics are perfectly fine for audio, provided that they are competently made. Most receivers of common name-brands like Sony, Yamaha, Denon, etc. are perfectly acceptable, as long as you have enough power for your speakers and listening level preferences. Your Parasound amps should work fine too." Dr: "Oh." Me: "I take a systems engineering approach to audio reproduction. That means that I am primarily motivated to improve on the weakest aspects of the overall system. The weakest aspect of any playback system, by far, is the speakers. The second weakest aspect is the listening room design. Electronic signal processing can be used to improve on both of these aspects, to some degree, so is worth considering too. Everything else is of minuscule importance." Unfortunately, the conversation didn't go much further from that point. That's a shame, because we'd finally gotten to the stuff I think is interesting and worth talking about. He never told me what kind of speakers he owned but did concede that his room was probably far from optimal. Next time I visit him, I will ask him if he was surprised by my responses and if he's given the conversation any further thought. I also hope to eventually persuade him to visit for a demo. I don't have any idea what he'll think. Maybe I should insist on blind-folding him before he sees the system. @Kvalsvoll, seeing as you have transitioned from a vendor of "hifi" wares to stuff that actually sounds good, do you have any older customers that come in and are shocked by what you've done? You've chucked a whole lot of fancy electronic equipment, and you've replaced your "hifi" speakers with ... horns! I imagine that comes as a shock to a lot of people. When I've gone to the local Rocky Mountain Audio Festival, I've been in rooms with CD horn speakers and seen people walk in, notice the speakers, and then make a disgusted face and walk out without even bothering to listen to them. Yet, I'll go into a large room with mediocre sounding but exquisitely finished speakers connected to huge racks full of electronics via cables the size of fire hoses and see numerous people standing agape. It's totally absurd. Like, there's a certain wealth level beyond which the majority of people lose any semblance of judgment.
  18. Bass system integration

    Hey. I'm glad someone started this thread, even though I haven't really had time to post. I generally agree with what's posted above as far as what to do as a starting point. However, I personally have found the sub/main crossover (XO) issue to be ugly at best. The trouble is that the LR4 XO, which is typically used is largely theoretical for subs/mains but not practically realizable. In a speaker, an LR4 XO is typically implemented between drivers that are closely co-located and typically vertically-oriented. For drivers that are direct radiating and installed flush with the baffle, the sound will typically be time-aligned and the XO will integrate ideally for listeners located anywhere perpendicular to the vertical axis of the baffle. Also much of the off-axis sound that is heard, that which arrives from the sides, should also integrate ideally. The ideal LR4 has each driver -6 dB from its pass-band level at the XO point. Thus, the levels sum to 0 dB under ideal circumstances. Under less-than-ideal circumstances, such as when the drivers are not time-aligned (a common problem with horn speaker design) or the listener off-axis vertically, circumstances are less-than-ideal, and the levels sum to less-than 0 dB. For some XO types other than LR4, it's possible for the levels to sum to greater-than 0 dB, but this is often undesired as this peak in the off-axis spectrum will tend to be audible. === Unfortunately when it comes to subs, the different drivers involved in the XO are frequently neither co-located nor vertically oriented. Furthermore, the wavelengths in the XO area are long enough for nearby boundaries to substantially interfere with the direct sound from each driver. The consequence is that the LR4 XO for mains and subs is rarely ideal. Using the "sub distance" setting, it is possible to make the LR4 XO ideal, but only for a single input channel (i.e. front center) for a narrow band of frequencies (rather than, necessarily the full XO range), and only at one seat location. With more effort, other compromises are possible taking into account more than one channel and/or more than one seat, but it's almost never possible to achieve ideal results across the XO frequency range for multiple channels and multiple seats. In fact, to do so practically requires co-location of the speaker and its own sub(s), which completely defeats the main advantage of bass management in the first place. (I.e., subs in locations other than the speakers often perform better in-room well below the XO frequency.) As above, where the XO circumstances are less-than-ideal, the response sums to less-than 0 dB, so without additional EQ, sub systems almost always exhibit a substantial power response dip or hole in the XO region, which happens to be a critical region for musical bass and for a lot of punch, impact, and chest slam feeling. Considering that lack of punchyness or chest slam is a very common complaint for home systems, I suspect that sub/main XO problems are often detrimental to this aspect of the sound. The only solutions are to fill it in with EQ, possibly causing an annoying resonance at certain seats and/or room locations,, or to abandon the LR4 XO. However, no other XO is obviously better, and the best solutions will likely have to be custom tailored for the room, equipment, and seats. My own room is kind of a worst-case example of this problem. The mains and front subs are both located such that the side-wall reflections contribute substantial destructive interference in the XO region frequencies at the seats as well as much of the rest of the room. The subs also suffer destructive interference from the ceiling. The MBMs I have are behind the sofa, so reflections affect things less for listeners at the seats, but the ceiling and side-walls still have a negative impact on them, due to their locations. Anything resembling an ideal LR4 involving the 70-100 Hz range on this system is a complete fantasy. These problems inspired me to develop a DSP system that allows me to specify filters for each sub unit *and* for each input channel in the XO. So for example, I might want to utilize my left MBM a lot more than my right MBM at 90 Hz when crossing the front left channel and visa-versa with the front right channel. The XO can be extensively optimized for each channel and fewer compromises. Achieving good results for multiple listeners is still very challenging with this approach. I am able to simulate the raw / unprocessed response at the seats, but the optimization is tedious, difficult, and time consuming. My approach thus far has been to manually iterate until I give up. I expect that better tools, supporting FDW and other analysis methods, will help me to get better results. However, I also plan to add more mid bass sources and am still trying to figure out which locations will be most useful. At least for now, I'm doing a lot better than I could before. Previously, I could make separate filters for each sub but they were shared among the XOs for all the channels. This helped me to reduce multi-listener response variations in the LFE channel, but the benefit largely fell apart in the XO region and above for mains channels. I did have EQ for the mains channels downstream from bass management, so what I did was run the subs a bit hot through the crossover and adjust the mains as best as I could for a smooth XO response. The bass through the XO was pretty good in the MLP,, albeit a bit hot, but it was uneven everywhere else. Anyway, this is still a work in progress. I finally finished an overhaul of the surround filters using the methods I used for the front speakers. This included a new bass optimization as well. All I've heard so far are some of the Atmos demos and other trailers. Wow! There's a lot of bass in the surrounds, and the fact that previous configs neglected to fill the XO hole made this quite apparent. I look forward to trying some movie scenes tomorrow and am hoping my surrounds have enough headroom. I was often pushing them to near clipping before this new config, so we'll see. I may need to figure out how to implement a DSP limiter. :/
  19. As far as PvAs, I typically give more attention to the peak than the average plot. Still, I agree that PvA shape does not line up with bass sound quality 1-to-1. For example, lots of movies peak at 30 Hz, but not all have 30 Hz boom.
  20. Newbie with 26 questions

    Hmm. Lots of questions here. For the time being, you could try some basic troubleshooting for the "sub". Play some bass heavy music and put your ear right up to the sub. Does it make any sound then? Subs typically don't just handle the "LFE" channel, especially on a soundbar system. If the sub doesn't make sound with just music, then try to figure out why it isn't working. If it does make sound, then it's likely that there is a configuration problem on the PC that is preventing the soundbar from receiving a full 5.1/7.1/Atmos signal including the LFE channel. Do you hear sound from the soundbar when the Atmos test file tests the "surround" or "overhead" channels? With that said, a soundbar micro size "sub" is not going to deliver much bass. Note I put "sub" in quotes because the sub included in that soundbar hardly qualifies as such. Some people around here, myself included, use mid-range drivers that are larger than what's in that "sub". Furthermore, chances are that it's not just "sub bass" frequencies (e.g., below 100 Hz) that are being diverted to that unit because the soundbar uses drivers that are very tiny. I don't recommend trying to use the soundbar with a separate external sub. Even if it can be made to work, the soundbar is probably specifically designed to be used with its included "sub" and may need the unit to handle frequencies that are higher than a true sub in a higher performing system would typically handle. As such, you'll likely leave a lot of performance on the table doing this. For a more capable system with serious sub capability, plan on obtaining at a minimum: an A/V receiver (AVR) and front left/right speakers. Decent speakers will probably do more than the soundbar with its sub can do. Once the speakers are squared away, then go for the sub(s). (Multiples are usually better.) A center and/or one or more surround speakers is also highly recommended, but you can live without them when you're just getting started.. You may or may not need amps for the subs and/or speakers, depending on what kind of subs you buy or build. Most commercial subs and some speakers have the amp built-in. Do you consider yourself to be DIY friendly? If you are willing to build your own speakers and subs, you can achieve much more performance for the dollar, albeit at the "cost" of your own labor. I'd argue that the speaker selection is probably the most important choice you make for a system. Most "good" subs offer suitable sound quality within their capability, and if you want more bass than your subs are capable of, you can just add more of them. The same is not true for speakers which must be replaced if you want better quality or more output. I consider the bass range to be everything below 250 Hz, even though subs are typically only used below 100 Hz. In my experience, speakers contribute a lot to both the bass sound and tactile feel. Weak speakers can really hold back a capable sub system. Sadly, I don't think most consumer speakers are really any good. The vast majority these days are just too small and inefficient. By inefficient, I mean that they don't produce much SPL even with a lot of amp power. Often their spectrum is tilted to the top to make them sound "detailed" (but not accurate) and louder than they should, at the expense of good bass impact. Very few speakers have a truly neutral spectrum, which IMO offers the best in terms of both detail and impact. Even the vast majority of "high end" speakers are far from neutral. As such, it's important to realize that many speakers that cost only a few hundred dollars can sound just as good or better than speakers costing thousands. Many people like using pro-style speakers for movies, even speakers designed for specifically cinema. In a home where you are sitting a lot closer, these will likely deliver plenty of powerful sound without any need for external amps. You can also find a lot of DIY designs that look and perform much like (if not better than) cinema speakers. One last point. If your speaker shopping leads you to choose something that is either inefficient (needs lots of power to play as loud as you want) or is self-powered (particularly common among some pro-style speakers), you'll need an AVR with pre-amp outputs. These AVRs are a lot more expensive than the lower end models, unfortunately, but it's better to buy what you need the first time. For that matter, if you are serious about using an external amplification (either separate amps or built-in to the speakers), you may consider shopping for an A/V processor instead of a receiver. A processor has only pre-outs and no amps built-in, so it can't power speakers directly, but it uses less power and runs cooler as well as being more likely to offer high quality XLR outputs, for example. Anyway, this should give you some things to think about. Note that this forum is pretty strongly focused on subs and doesn't get a lot of traffic. You can get a lot more input if you post on a place like AVSForum. It's a pretty big place, and I think it has a dedicated newbie forum. However, if you are seriously interested in going the DIY route, I recommend putting your first post in their DIY forum instead of the regular newbie forum. There're lots of people there that'd be happy to assist, err indoctrinate you into this hobby and can walk you through designing a very capable system including subs that can deliver everything you crave and more. It's not about "rattling your trunk" or deafening yourself, either. Quality reproduction of sound takes a lot of horsepower, so to speak, but if done well, the results are astounding and incomparable to what you're probably used to at a typical cinema or home system. Good luck!
  21. New CEA 2010 in room Measurements.

    I guess I missed that. Anyway, he probably did the CEA in-room just to show how CEA test results can vary dramatically outdoor vs. indoor. Every room is different, and there's no way to really standardize in-room testing. As such, it's mainly there to show the large differences that are possible.
  22. New CEA 2010 in room Measurements.

    I can't find what you are referring to. Do you have a link?
  23. New CEA 2010 in room Measurements.

    I don't know of any such thing. If Josh is doing this now, it's the first I've heard of it.
  24. I think the "Series Resistance" parameter in the dialog you show is for the resistance of the wire going to the drivers. Just set it to zero. The EQ option is there to show you EQ will effect the response shape, and this shape change. What it shows in the SPL output view, however, is not the output you'll see for the power input you specified. The EQ you specify is applied to the curve in the SPL output without consideration of the additional power and voltage requirements to achieve the output. Remember, SPL at any frequency is always limited by excursion, voltage, or power, whichever runs out first. For evaluating the driver and cabinet, I recommend you don't use the EQ feature to avoid confusion. As for what SPL to aim for, that depends on how you listen and on the in-room response, which varies between rooms and different placements of sub and listener. It also depends on the nature of the content. Movies at or near reference level can demand a lot of output. Designing for 130 dB SPL down to 30 Hz is not reasonable with 120 dB or even higher down to 15-20 Hz. Output in the mid one-teens appears to be sufficient below there for most films, but there are exceptions. In most cases, the answer to "how much SPL" is: "the more the better". However, with the drivers installed in the riser, the actual SPL at the seats may not be as important as the amount of vibration felt at the seats, so you might be able to get away with turning them down a bit more. You discussed building ported subs for the corners with some bigger beefier drivers, and those seem like the way to go for the higher SPL stuff.
  25. Several different sets TS parameters are sufficient to model a woofer. Of the parameters that are typically specified, many are redundant and can be calculated from some of the others. As such, it's kind of hard to succinctly describe which parameters are absolutely required. For example, one commonly used set consist of: Fs, Vas, Qes, Qms, Re, and Sd. Frequently, Mms is substituted for Fs. BL may be substituted for Qes. Rms may be substituted for Qms, and so on. Only five parameters is definitely not enough, although if they don't give Re in the TS list, you might be able to find that info elsewhere.