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Everything posted by SME

  1. Thanks Kyle! Actually, he jumped the gun a bit here and declared victory while the update was still running. NOW it's all done. Edit: Total count of posts updated: 169!
  2. Bulding the Room2 listening room

    A big caveat here is that the response of phone mics is a big unknown. I have a Google Pixel XL, and I recently installed an SPL meter / analyzer app. I figured it'd be helpful for a quick SPL measurement while playing music, without having to get up and go get my SPL meter. The app is pretty comprehensive and allows selection of Z-weighting for flat response, in the app at least. Unfortunately, the mic on my phone seems to have much diminished response in the sub range and also has poor dynamic range. IIRC, it taps out in the 80s dB, which is pretty much useless for measuring music that's loud enough to be fun. The sub response on the phone seems to be bad enough that it'd be useless for doing sub measurements. I wonder if they high pass it on purpose to improve speech clarity and reduce unwanted clipping? Oh well.
  3. Bulding the Room2 listening room

    I agree that there should be no advantage to SBA/DBA solutions, provided that "you can manage to somehow get rid of the cancellation reflections", or otherwise remove the effects of the room using another method. In both cases, one achieves very smooth (near-anechoic) bass frequency response (when viewed without using smoothing) across a wide region of space. I don't have an SBA/DBA, but I do have DSP-optimized filters to achieve the above near-anechoic ideal. This condition is maintained up to about 65-70 Hz, above which I have multiple issues preventing me from maintaining an completely smooth response. This is probably an issue in most other types of near-anechoic sub systems, including SBA/DBA also. To be honest, I wasn't expecting to have to go back and shape the broad response by ear the first time, and I was even more surprised when I had to shape the broad response *in a different way* after I re-did it with new crossovers. More recently, I did an EQ overhaul of my mains speakers above 160-200 Hz only, and I had to re-do the broad shape of the subs *again*. These weren't minor changes either. Whereas before, boosting below 40 Hz even 0.25 dB caused the bass to go to mud, with the latest config, I was able to boost the low end by several dB without loss of intelligibility. So at this point, I'm not convinced that any particular way is better than another. If anything, my work with mid and high frequencies has convinced me that some early reflections are better than none, provided that the speaker is well-behaved. The main reason so many pros are convinced that early reflections are bad is because they are experienced with using monitors with poor off-axis response, and the elimination of early reflections prevents that nasty off-axis sound from corrupting the better on-axis sound. With good speaker design, that's no longer an issue. It does lead me to wonder if maybe early reflections are good for bass too, to a point. It appears to be well established that modal resonances have negative perceptual consequences, but do early reflections have as much impact as we think they do? We practically hear through early reflections for mids and highs. We may also do so for bass, but it probably depends on how much reflected sound energy there is in total and how well it is distributed in time. At lower frequencies, there is definitely a propensity for the sound field to become quite structured within the room, even if discrete modes are not obvious. This is a substantial open problem that I plan to devote more time to in the future. I would strongly caution anyone not to read too deeply into frequency response data. Because we hear pitch (frequency) and level, it is easy to assume that a frequency response (FR) plot tells us how different pitches will be emphasized, relative to one another. However, this is far from the truth. An FR plot with smoothing is largely meaningless because the smoothing discards most of the information that's relevant to perception in the first place. An FR plot without smoothing and with phase data does contain that information, but it is a terrible visualization of that information. Time-frequency plots like waterfalls and spectrograms are kind of a step in the right direction, but it is hard to glean quantitative information from them and the information within them is still not weighted very close to how perception weighs the information. All of this makes sense if you imagine what it would take to analyze an IR to determine the true spectrum of a source within the room. One must deal with a variety of acoustic interference effects and possible obstructions in the path of the direct sound. It takes some very clever processing in order to accomplish this with the accuracy that our ears and brain do. As can be seen, I'm rather short on good advice here. I am less confident in what I know about bass reproduction than I ever have been, having tried a variety of strategies and having failed, in the sense of not achieving any consistency. And that's just in one room. And this is yet another reason why I am very skeptical of the relevance of particle velocity, independently from pressure. Why? Because there's so much we don't understand about how pressure response affects perception. Unless or until one constructs an experimental apparatus in which velocity response can be varied while keeping pressure response *exactly the same*, no one has proven anything with regard to the relevance of velocity response. In practice, this kind of test is extremely difficult to do. Almost anything that changes velocity response will change pressure response in some way. "Close" does not cut it here.
  4. Bulding the Room2 listening room

    Of course the boundaries of a real room are not perfectly rigid. They are lossy, especially at certain mechanical resonance frequencies. Your room is also open at the rear. However the side-walls, floor, and ceiling are rigid enough that the sound field between those dimensions is qualitatively similar to what I described. Pressure still peaks at the boundaries where velocity drops to almost zero, and there are likely dips at certain frequencies . My point is that the effect of multiple subs on your velocity measurements is entirely consistent with this perspective. I don't know what you mean by "intensity pressure". If you mean merely pressure, then we are in agreement that pressure is the most important characteristic for transmission of vibration from the air into the body of the listener. The proposition that the motion of clothing depends on velocity is interesting, but I suspect the situation is more complicated than that. It may have more to do with pressure gradients, which may coincide with areas of high velocity as it does for standing wave sound-fields, but high velocity and high pressure gradients don't always present together. It would be interesting to do some experiments with subs outdoors to see if clothing movement perception is affected by source distance, while pressure is kept constant. In the far-field of a monopole radiator, pressure and velocity both drop with 1/R, but the pressure gradient drops with 1/R^2. Thus, if what I suggest above is true, we'd expect less clothing motion at greater distances, even after compensating for SPL. To your point that frequency response and phase are very important, I totally agree, but I would say it's a lot more complicated than most people think. And of course, the sub range is only one part of the picture. The rest of the speaker response also impacts perception a lot, and 100-500 Hz is particularly important for tactile sensation. Unfortunately, this range is often harmed by speaker placement problems, but there may be ways to fix this with EQ. This is work in progress for me, but I can say with confidence that a perfectly flat or smooth in-room response is not optimal unless the room is completely dead. And if the room is completely dead, then you have another problem.
  5. Bulding the Room2 listening room

    So if the listening position is approximately half-way between the two side-walls and approximately half-way between the floor and ceiling, it makes sense that you see a lot of velocity along those dimensions with a single sub. The 1st order room mode contributes pressure peaks at each boundary and a pressure null in the middle. For velocity, the reverse is true with a velocity peak where each pressure null is. When you place sub(s) at opposing boundaries across a particular room dimension, they cancel the mode, leading to a much more even distribution of pressure and near elimination of the velocity vector in that dimension for those frequencies. More generally, where standing waves dominate response, the particle velocity at a point in space is usually proportional to the spatial-gradient of pressure, i.e., the rate of change of pressure with respect to spatial location. (This is more or less the same as saying that they are separated in phase by 90 degrees because [math alert] the derivative of a sine is a cosine and the derivative of a cosine is a negative sine, and so on.) The pressure gradient is actually highest in the nulls (it quickly rises in either direction), and velocity peaks here. In contrast, when approaching a rigid boundary, the pressure gradient in the direction normal (i.e. perpendicular) to the boundary tends toward zero as does the particle velocity. Here's a visualization, from here: Note that if you are concerned about sound intensity, it is essentially zero in both pressure peaks and nulls. In fact, a key feature of standing waves is that, on average, no net energy is transmitted through space at all, so sound intensity (in the RMS average sense) is essentially zero everywhere. Instead, the energy of the standing wave oscillates between potential (pressure) and kinetic (velocity) forms and sloshes back-and-forth between the pressure/velocity peaks and nulls in the process. Of course, the situation changes in the presence of a listener, which acts kind of like a membrane bass absorber. Sound intensity across the skin depends mostly on frequency (i.e. chest cavity resonance) and pressure because the impedance of the solid/liquid flesh is much higher than that of the air.
  6. Bulding the Room2 listening room

    It's a very narrow room. Those are 12" woofers, so you can kind of eyeball the width as maybe 8-10 feet. I don't recall, but I think the room stats were discussed earlier in this thread. So with 4 subs, almost all the velocity is in the front-rear dimension? What do the absolute magnitudes look like in each case? I.e., the square root of the sum of the squares of velocity in each dimension?
  7. Hey! Who said there's anything wrong with chick flicks? ... assuming they have good bass, of course.
  8. You make a good point and are probably right that at least some home mixes are done with very minimal effort. We have no idea why the -3 dB limiter was put on the TFA track, and purposeful dynamics reduction is only one possibility of many. I do believe the TFA track was re-EQed, and I actually think it is very nice sounding as far as EQ balance is concerned. Of course, that doesn't necessarily mean that they devoted a lot of time to perfecting it. Maybe Skywalker Sound Studios offers a kind of standard re-EQ filter-set that they recommend to mixers as a starting point that generally works well for content mixed in their cinema track facilities. Seeing PvAs of cinema vs. home mixed versions of a track would certainly be insightful. Even better would be to hear from the mixers themselves, but unfortunately, they don't frequent forums much. And when they do, they often have to confront a lot of negative sentiment about their practices and about the quality of the mixes. I can't blame them for not wanting to visit when they face so much vitriol, but I also don't believe it's one-sided either. In some exchanges (not naming any names here) I have noticed a know-it-all attitude and an unwillingness to consider alternative viewpoints. I've seen insistence that home systems are inherently inferior to larger room systems or that small rooms can't adequately reproduce the lowest bass frequencies. (!) Then there's the stubborn insistence that mixing in the near-field somehow approximates a home environment better than a dub-stage, even though the data I've seen suggests more similarity between a home theater and dub-stage than between near-field and either environment, particularly if the dub-stage is calibrated to a more neutral target than the X-curve. By itself, the reliance on the X-curve standard is a serious embarrassment to the industry, a problem I don't blame on mixers because they have a job to do which is not calibrating the systems or developing the standards. Nevertheless, it is for precisely that reason that the insistence of "knowing what's best" for production of home content deserves serious criticism.
  9. That's not necessarily true, at least if the mixers are recent enough. From what I understand, "mixes" these days are actually implemented in meta-data rather than done literally. That's to say that a "mix" now-an-days consists of all the original tracks and objects along with instructions regarding how they should be combined. When an engineer hits "play" or instructs the system to export content for master, everything gets mixed in the DAW on-the-fly. If that's the case, then the filter(s) may just be meta-data regarding how to process a certain group of sounds or perhaps specific destination channels. Those filters could in principle be changed with just a few adjustments in the software. Of course, adjusting the filters is likely to have other knock-on effects such as a suddenly increasing headroom demand in parts that could lead to more aggressive activation of compressors, limiters, and/or more propensity for clipping. Of course, if it's a high headroom home mix with re-EQ, then there might be extra headroom for a more relaxed filter ... or not. If a mixer can adjust EQ levels for dialog separate from FX, then he/she might pull back the bass a bit in the dialog while leaving most of the sub effects hotter. Why not? Everyone likes more bass, right? Then again @maxmercy's BEQ work suggests there's already plenty of spare digital headroom in most tracks to not use filters. Incidentally, I have heard that there is some kind of unofficial limit for bass-managed output that the industry is supposed to adhere to, something like 120 dB total @ reference. (?) I think cinemas are told to spec their subs to be able to reach that level of output. However as evidenced by measurements here, not every soundtrack actually adheres to that limit.
  10. This thread is more about full-range content than bass, but it is content related, so I think it works best here. In the future, I may post this somewhere on AVSForum, but for now I want to keep it to a limited audience. As I've mentioned in the main LF Content thread, the X-curve calibration standard in cinema causes two major problems: Tonal balance that deviates substantially from neutral and from what is typical used (informally) for music production and what sounds good on a home system that is optimized for music. Inconsistent calibration between different dub-stages and cinemas. As I also noted, many UHDBR/BD/DVD releases these days have high quality home remixes that fix most of these tonal balance problems. This is true for most recent Disney releases these days (including, e.g. the new "Star Wars" and much recent Pixar and Marvel stuff). However, much legacy content as well as lesser quality home-remixes do not feature any re-EQ and retain the inverse-X-curve signature. The effect of X-curve calibration is to attenuate both high frequencies, via the -3 dB/octave slope in power response, and the low frequencies, which arises from forcing a flat power-averaged response even though virtually all speakers have a significant drop in directivity for low frequencies and what absorption is present in typical dub-stage / cinema rooms is also less effective at low frequencies. As a consequence of the altered tonal-balance, most mixes are likely altered to sound good in the dub-stage during the re-recording mix process in which highs and lows are boosted to compensate. The resulting mixes, in addition to translating unreliably between theaters, sound less than optimal when played back on a home system. The auditory symptoms are mixed. I find it easiest to hear the problems in the dialog. Sometimes only one of the excess highs or the excess lows is audible in the unaltered track because the boost dominates. For example, some cinema mixes, the dialog comes across very bright. In others, it comes across very boomy. Sometimes, the dialog seems relatively balance, in terms of high vs. low, but with the mid-range being relatively depressed, intelligibility often still suffers. Dialog is both much easier to understand and much more enjoyable to listen to when it's presented neutrally. Unfortunately, the required correction varies between track for both of the above reasons. Mixers don't necessarily attempt to defeat the X-curve alterations in any systematic way. Instead, they "turn various knobs" and listen until they are satisfied with the result. So the ideal filters to reverse their changes may vary between mixes. And because the X-curve calibration method isn't even consistent between dub-stages, EQ-adjustments that give good sound in one dub-stage may not work well in another. In fact, there's evidence that X-curve calibration doesn't even achieve consistency between the left and right vs. center screen channels vs. surround channels in the same dub-stage. The situation is a big stinking mess for sure. Nevertheless, even if the adverse effects of the X-curve standard on the mix cannot be perfectly reversed, it's possible with some rudimentary EQ to improve the sound quality of cinema mixes considerably. Now that I've finally achieved a stable, reliable audio reference in my own sound system, I've been giving attention to this problem. In this thread, I hope to document some of the candidate corrections that I've applied to improve the sound quality of various movies. I would encourage anyone with the required capabilities to give these a try and share feedback. To implement these requires the ability to apply various biquad EQ filters such as high and low shelves and Peaking EQs, ideally to the streams *before* bass-management. Though for my first pass, I'm applying the filters identically to all channels, so it should work fine to apply them after bass-management as well. One issue I imagine most people will have is that they have a limited number of free filter slots. The more filters used, the better quality correction that's possible. I will try to limit the filters to what's actually needed. Edit: I posted a candidate correction for "Wonder Woman". Sweet!
  11. Obviously, they remixed the track for the BD release. There are many good reasons to do remixes, but unfortunately the quality can vary a lot. As for filters, chances are that the mixers had no idea what they were cutting out. Often the systems used for these remixes aren't really up to the task. I've seen evidence that tiny near-field monitors and a single compact near-field sub, used in a huge room. Such a sub is likely to struggle to keep up even with the content above 30 Hz and at the lower monitoring levels commonly used. There are many possible reasons the filters get applied. First, many mixers assume the content below a certain point does not contribute to the mix and apply the filter as a matter of habit. Most cinema systems don't extend below 30 Hz either, so this shouldn't be a surprise. Second, mixers may be applying filters to protect their own equipment or prevent it from distorting. This is very unfortunate because I'd argue that the mixers ought to be applying such filters to their monitor output and not to the soundtrack itself. On the other hand, some mixers worry that if they can't hear what's going on "down there", then something they don't want to have on the track may slip through. I would argue that this happens anyway because of a combination of issues: poor quality monitors (which is most of them), poor quality listening space (near-field monitoring in a large room is a very poor listening environment for hearing soundtrack details), etc. Third, mixers may be applying filters in order to make the soundtrack louder. I'd argue that this is something that shouldn't happen for a home remix but probably does. While home remixes should be monitored at a lower level, consistent with changes in listening distance, room size, and other factors, they are often monitored at even lower levels than that. The assumption is that home listeners will listen even more quietly than a "room appropriate" reference level, so the mixers want to boost low level details to ensure they aren't lost. I don't really have a problem with that, but it goes very wrong when mixers start boosting the level of loud content too, in order to "increase impact" or "satisfy director's intent". The fact is, home listeners don't usually set their volume to a number but do so by ear based on loudness, so boosting loud content will only make home listeners turn down the content more, especially if they hear clipping and distortion that often gets introduced in such a process. I suspect that earlier home mixes including many BD re-releases suffered more from quality problems than more recent home mixes. Dedicated rooms designed to mimic home theaters and using better quality monitors (such as the new JBL M2/708 series) are becoming more common. However, filtering in general is still wide-spread and is more the norm than the exception, as can be seen by following this thread. We can only hope that, in time, more dedicated rooms are built and equipped with more capable sub systems. I can understand the concerns about equip a full size dub-stage with subs that extend into the single digits, but in a dedicated home theater mix room, this kind of setup should be much more practical. We'll see.
  12. Holy humped filter Batman! It looks like they used PEQs instead of shelves to attenuate the low-end, given the sharp rise below 3 Hz.
  13. X-curve compensation re-EQ

    I watched "9" tonight. My verdict? It sounds *excellent without EQ correction*. The dialog on the film sounds very natural and well-balanced with excellent mid-range clarity and just the right amount of fullness. It also goes without saying that the bass effects on this soundtrack are tremendous. The large hits are very wide bandwidth, combining both slam and weight. The "9" soundtrack is a reference for home theater audio systems in every respect.
  14. X-curve compensation re-EQ

    I want to make some other notes about recently watched movies. After finishing re-EQ of my front stage, I watched "Finding Nemo 3D" on BD (2010), IIRC released in 2012. Being that the original was mixed in Skywalker Sound Studios, I was curious if the re-release might have a quality re-EQed home mix. It would seem not. It's possible that they didn't bother with a remix at all, being that the major change was in the video (3D conversion). Or if they did remix it, they did so in a Disney studio instead of at Skywalker Sound. A -2 dB/octave HF slope and -2 dB bass shelf cleaned it up very nicely, as is typical for other cinema mixes out of Skywalker Sound like Wall-E. Apart from being tainted by cinema EQ, the mix is very high quality as with most Pixar stuff. It deserves mention that the dialog is noticeably hot on this mix. It makes me wonder if Disney did a "lazy" home mix and just punched the dialog up 3 dB or whatever. Tonight I watched "Zootopia". This is a very recent Disney release (2016) with audio post done in Skywalker Sound Studios, so I expected a re-EQed mix. However, the mix was obviously shelved in the 200-300 Hz range (the usual spot for that sort of thing). I could not judge the highs because there was obvious low-end masking, but perhaps this movie did not get a home mix, was not done in Skywalker Sound's dedicated room, or did not get re-EQed for some other reason. Who knows? As a curiosity though, the "Zootopia" BD contained several "Deleted Scenes", and the spectral balance of the audio in those scenes was totally different. It sounded much more natural and balanced, if even a bit thin (in the low frequencies). The more balanced rendition revealed a lot more mid-range detail and nuance in the actors and actresses voices. This was even more obvious at a few points in which the directors replayed snippets from the actual movie in order discuss the (deleted) scene that was about to be presented. As such, the difference in spectral balance probably did not arise because of work a mixer did on the audio for the special feature itself. My guess is that the audio for the deleted scenes was presented as it was recorded, albeit with whatever EQ was applied to the clean-up the signal from the mics. Clearly these scenes were cut during an earlier phase of development as some consisted only of hand-drawn stills while others were CGI rendered at lower quality. The implication here is that the dialog audio presented in those scenes was from an earlier phase of production, perhaps before it had ever seen an X-curve calibrated dub-stage, at which point, bass boost was applied.
  15. X-curve compensation re-EQ

    I plan to resume this work soon. It's taken me a long time to develop new EQ configs for my speakers after becoming aware of the "hidden resonances" I described. The effort has been very well worth it, and my sound quality is substantially improved from before, which was very good already. I finally finished updating the surrounds early today, but I want to take some time to watch a lot of movies and develop full confidence in this new configuration before attempting to do any critical work. From what I've heard so far, I'm not inclined to revise anything I've argued in this thread so far. With these new EQ configs, the broad tonal imbalances found on typical cinema tracks are somewhat less objectionable to me than they were before, particularly with regard to the low frequencies. However, the imbalances are just as apparent if not more so. On the other hand, fixing the hidden resonances will help immensely with getting my judgments right. The center had some significant issues in the 200-300 Hz, which I was aware of and did my best to work around. This range is crucial because it is where anechoic flat speakers tend to see a lot of gain from baffle step loss and/or reverb time increase and so is a region where alterations as part of the X-curve calibration procedure are likely to be more substantial. It is the range that I most often apply center shelving filters to deal with low frequency excess. I also had significant "hidden" resonances around another crucial area, between 2-3 kHz or so. In hindsight, this explains why "getting the knee right" seemed to be such an unforgiving exercise. Though, I imagine this region will still continue to be difficult simply because the X-curve "knee" resides at 2 kHz. As I previously noted, the soundtracks I've worked on sound quite bad with a sharp knee at 2 kHz. A more gradual transition from "flatish" to "sloped" seems to be required for the best sound, but getting the shape of the transition right is crucial and is different for every track. The area around 2 kHz has a big impact on speech and the upper-mid "punch" in many sounds. As an aside, I own a copy of Floyd Toole's most recent (3rd) edition of "Sound Reproduction: The Acoustics and Psychoacoustics of Loudspeakers and Rooms", which is a superb book on the subject, which is a superb reference for all things having to do with high quality audio reproduction. I can't recommend the book enough! Anyway, Toole dedicates an entire chapter to cinema sound and the X-curve with a lot of great info. He has a lot to say about the knee at 2 kHz as well: Anyway, there are many other details in that chapter that are worth discussing here at some point in the future. I'll just say that my arguments here are reasonably well supported by Toole's opinions as well. The X-curve standard degrades sound quality, both during the production process and during the reproduction in typical cinemas.
  16. The Bass EQ for Movies Thread

    OK, that makes a difference.
  17. The Bass EQ for Movies Thread

    Most of the BEQ configs posted here apply different filters to each channel *in the soundtrack*. That means that the filters must be applied before bass management (in which the bass for each main channel and everything from the LFE channel is summed and sent to the sub woofer). As such, using the PEQ built in to your subwoofer is not likely to yield the best results, and in many cases, it won't be clear which settings you should use. The benefit of the NanoAVR is that it is installed before your AVR in the device chain, and therefore, it is capable of applying processing to the channels in the soundtrack before bass management. Another point is that I don't know how big your room is, but these BEQ filters require a lot of capability below 20 Hz to take full advantage of. While the PB16 can certainly play below 20 Hz, its capabilities are quite limited compared to many of the systems people here use for that sort of thing. Of course, a lot depends on how your room behaves, but you might want to consider at least a second PB16 if not a more substantial DIY system if you want to get more out of BassEQ.
  18. Very Interesting. While nothing here constitutes definite proof, it does seem reasonable to me that the streaming version is a cinema track mix-down; whereas, the BD version is a re-EQed dedicated home mix. As I've argued before, cinema tracks need quite a bit more low frequency oomph for good impact in an X-curve calibrated cinema. That's because X-curve calibration undoes the natural in-room bass rise exhibited by an anechoic flat speaker due to boundary gain and reverb build-up. Of course a lot of people at home also have systems with less bass output, either because they calibrate to a flat curve (e.g. Audyssey) or because their speakers lack BSC or because they have boundary interference problems. Nevertheless, it appears that recent BD releases with home mixes done at Skywalker Sound Studios have re-EQ to better match systems that perform optimally for music playback. In terms of the graphs, the streaming version looks 5-7 dB hotter through much of the sub region. However, the gap may be much smaller after compensating for loudness differences in the mids and highs. In that case, it may be more accurate to say that the BD version is hotter than the streaming version in the 15-35 Hz region. Certainly the shift of balance toward deep bass could reduce the apparent level of mid-bass, even if the SPL is similar after compensating for loudness difference in the mids and highs. There's a good chance I'll buy the BD version of this film. I may be tempted to try out the streaming version to satisfy my curiosity. I could give my opinion as to whether the streaming version sounds like it is influenced by cinema EQ, for what that's worth. That UHD Atmos tracks often sound louder than BD DTS-MA is a curiosity. Almost all DTS-MA tracks have "0" dialnorm offset, and I don't believe any format supports positive offsets. It's possible that a lot of Atmos "home" tracks are just mixed hotter than the cinema versions, from which the DTS-MA may be derived from. Unfortunately, there are still no formal standards for home mixing and apparently no consistency between studios. For example, I believe (based purely on my subjective evaluation) that Skywalker Sound Studios applies re-EQ to home mixes, whereas most other studios don't. Skywalker Sound also appears to have a dedicated mix room and to use a calibration/mix level that's comparable (in terms of room size differences) to cinemas, i.e. 80-82 dBC @ 500-2kHz. Such mixes are likely to sound quieter, in addition to benefiting from more headroom and cleaner micro-dynamics than cinema mixes. OTOH, it appears that some studios monitor home mixes with calibration as low as (or maybe even lower than) 75 dBC and may still be monitoring near-field in a large room. Such tracks are likely to sound even hotter than cinema tracks and have more potential for clipping and other problems. Also under those conditions, the need for re-EQ is likely to be much less obvious for a number of reasons: (1) tonal imbalances are much less obvious and offensive at lower levels especially excess brightness; (2) lack of boundaries reduces low frequency boundary gain that boosts the bass of flat speakers / mid-field monitors in "small" rooms; and (3) per Floyd Toole, rooms with early reflections are more revealing of tonal balance flaws in a speaker, and I'd argue that this extends to soundtracks as well. From my knowledge, near-field monitoring in a large room is probably the worst environment to monitor a home mix in. Simply monitoring the mix on the dub-stage system, albeit with a Harman-like curve instead of the stupid X-curve, is likely to offer better translation than "near-field". Somehow I need to get the industry people over to my house to listen to and compare mixes.
  19. Now *that's* weird! It looks like the streaming version has hotter bass overall as well. Hotter bass and smoother bottom end roll-off. Does the disc by chance have a separate 5.1 track on it? Maybe the 5.1 track on the disc matches the streaming track. Maybe it is a mix-down from the cinema track vs. home mix for the 7.1. Apart from bass, did you notice any significant loudness differences between the 7.1 BD vs. 5.1 streaming tracks?
  20. Is the streamed version a multichannel track with LFE? Or is it only 2 channels? If it's 2 channels, it's possible they omitted the LFE channel on the mix down. Going by memory, I think the LFE channel often gets more 30 Hz hump than the screen channels. This looks like yet another track with BEQ potential, but that 8 Hz peak looks intimidating, kinda like the one in TIH that overloads my sub amp. Edit: On second thought, I don't see an obvious 30 Hz hump. Though, the BD version still has a lot more output there, relative to the higher frequencies. It may still be a difference of LFE vs. no LFE.
  21. Looks like it's roll-ed off a bit below 30 Hz. Could benefit from BEQ maybe? I know one thing I really wished for in the Interstellar track was those bottom notes on the organ, down to 16 Hz. This track looks like it has at least some life down there, but could there be more?
  22. Rockford Fosgate T3 19 discussion

    Let us know if they work! Only -28 though? If calibrated to cinema reference, that's like background music.
  23. iPAL enclosure details

    Only in so far as you lose a slight amount of volume by having a solid panel between the two air spaces.
  24. Holophony

    When I first heard about it, I thought wave-field synthesis (WFS) sounded like a really cool concept. I'm a lot more skeptical now. Part of the problem is that a system of immense complexity (and expense!) appears to be required to achieve a high quality realization of WFS. Second, it's not clear that its really solving the right problem. In its ideal realization, WFS can synthesize a complete, spatially consistent (or spatially-dependent, if so desired) sound field within a listening space. This is basically the Holy Grail of audio. If an entire sound field can be reproduced in the space perfectly, then the reproduction is absolutely true and correct. In reality though, WFS cannot be realized ideally with any practical configuration of existing components. As a consequence, there will be errors in the reproduction. On paper those errors may be fairly minimal, especially compared to the gross distortions to the sound field induced by the effect of acoustic boundaries in a "normal" system involving speakers playing in a room. However, it turns out that people are very well adapted to listening to sources reproduced in rooms with complicated acoustic effects; whereas, they may not be that well adapted to listening to the errors that arise from WFS. From some reading, it would seem these errors have been minimized enough for the strengths of the technology to be fully appreciated. That is encouraging, and I'd certainly like to hear a setup some day. I'm sure it has its benefits and its applications. Though I can't help but wonder how much better the tech could be if wasn't so obsessively focused on creating a perfect / anechoic sound-field replica and instead took advantage of the acoustics of the space its in to achieve a smoother, even if less "correct" sound. Edit: I forgot to add that I'm in the early planning stages of trying to build my own arrays consisting of many independently-controlled elements, to be used as surround and/or Atmos speakers that provide far more even seat-to-seat coverage than conventional speakers could achieve. I'm not sure if the approach I plan will look like WFS or not, but I have rather different objectives in mind, so who knows?
  25. Sundown ZV4 18D2 - sealed enclosure

    Yeah, it should work OK. It may not absorb as well as some other options like rockwool or fiber glass, but those are a lot more messy to work with. You'll want to use staples or some other method to keep it in place. Another option which is more pricey is to use acoustic foam wedges with spray adhesive, like I did. This is kind of a misleading conclusion. Dense stuffing will indeed alter the relative response shape to have a lower Q, as though the enclosure has more volume. However, the response shape changes because of a loss of output near the resonance and because of an increase in output below it, which is what happens with more volume. So even though too much stuffing *looks like* an increase in volume , all you're really doing is throwing away output. You still want enough stuffing to absorb the rear wave from the woofer and control standing waves inside the enclosure though. Yep. What's important is total cross-sectional area, so basically: number_of_cables * pi * (D/4)^2, where D is the diameter and can be looked up in AWG charts via Google.