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SME

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Everything posted by SME

  1. I'm not the least bit surprised that the resulting mono tracks look very similar, even for cases in which the results sound totally different. In fact, I would expect this to occur especially when people are using your app to create the BEQs visually as opposed to doing it completely blindly and by-ear.
  2. I agree with you in part, but I think the differences can be greater than you'd think. A lot will depend on the particular mix and also the particular playback system and possibly some subjective preference. In TLJ, the failure to recover ULF from the surrounds is a sin of omission, which is relatively minor. Yes, it does mean that a spaceship might lose its weightiness as it pans from the front, overhead and to the rear, but at least the sound is not worse than what you started with. I picked the surrounds in my example because the difference is quite dramatic on paper and is one that we could all agree would be very audible with those discrete surround effects. However the front LRC channels are another story. Even though they roll-off at a similar point, their shapes are still quite different from LFE. So an EQ solution that is optimized to the mono sum average (which is dominated by LFE), could introduce new humps or bumps into the front LRC that weren't there before. Here's where we *can disagree* about what's audible and what's not. Though arguing from personal experience, even quite small bumps can be audibly degrading. Much depends on shape and bandwidth of the feature, in addition to the level, and also ... Audibility of differences will depend on the playback system. Systems with substantial bass problems may not reveal degrading resonances as readily. (That's not a virtue as such systems also fail to reveal a lot of content.) For example, a BEQ filter applied to front LRC that increases ULF while adding a slight bump around 55 Hz may have a pronounced boom around that area in general, but on a system with a severe boomy room mode at 45 Hz, the problem at 55 Hz may be hardly noticed. The BEQ might be an unqualified improvement on this flawed system, but on a system with very clean bass response, the 55 Hz bump may be much more obvious and degrading. If you had to choose between full ULF extension and balanced response between the deep bass, mid-bass, and upper bass, which would you choose? Personally, I'll take the balanced response over the ULF extension any day. IMO, the ULF is the least important frequency range. I believe the notion that "[global] BEQ that gives 80-90% of the improvement" is overly optimistic, but I am also inclined to judge the soundtrack for what it will sound like on a revealing system vs. an "average" one. So practically speaking, a global BEQ may be an improvement for most people who choose to use it, even if it does degrade other aspects of the bass somewhat. And I do understand that most people who have EQ capability at all can only use it on the sub output. I agree many filtered tracks can be improved to an extent with a global BEQ and that it's worth doing even if an independent channels BEQ would sound better. But I'm skeptical that a global BEQ will always be better than nothing at all. Focusing only on ULF, a BEQed track will always seem to be an improvement, but if one considers the sound as a whole, BEQ that introduces new bass resonances in some of the channels could end up sounding worse than nothing at all. Again, a lot is going to depend on the playback system. When doing these BEQs (whether global or channel-independent), it's very important to listen to the results on a system that is as accurate and revealing as possible. (This is probably my biggest gripe with the AVSForum thread where it appears BEQs are being developed using all eyes and no ears.)
  3. Again I disagree, and I think you're missing something key here. For any effect that is mixed exclusively to the surround channels and that contains sub bass , the difference between an independent-channels BEQ and all-channels BEQ is whether or not it's filtered at 60-80 Hz. The all-channels BEQ does boost the low end of the surround channels, but nowhere enough to keep it from rolling off rapidly below 60-80 Hz. I don't think we disagree over whether "filtered at 60-80 Hz" is subtle or not. To repeat myself, the only real open question is how much of the bass within the sounds mixed to the surround channels was copied to the LFE channel. To answer that question requires comparing the tracks side-by-side to see what the mixers did. To repeat what I wrote above, the total shelf gain recommended by @maxmercy for TLJ was +48 dB and not +60 dB. The +60 dB for GotG which also recovers meaningful content. Either way, any attenuation of say 60 dB by a filter should not be enough to force the relevant content into the quantization noise floor unless the tracks are getting down-sampled to less than 24-bits (i.e. 16-bits, -96 dBFS quantization noise floor) somewhere between where the attenuating filters were applied in production and where the BEQ filters are applied on playback. I believe DTS HD and Dolby TrueHD are always 24-bit. The rest of the production chain was almost certainly using at least 24-bit precision (probably 32-bit or 64-bit float in the DAWs). Realize that 24-bit has a quantization noise floor of -144 dBFS. It's actually quite impressive that I can get away with the +60 dB boost for GotG being that the signal inputs to my processor are analog (unbalanced actually). The ULF noise floor of my unbalanced analog connections must be in the neighborhood of -100 dBFS. This is incorrect because it is outdated, on two accounts. The situation changed with immersive formats. First, as I explained in my above posts, Atmos for cinemas (and probably other immersive formats) introduced support for bass management to be used for surround and overhead channels with dedicated "surround subwoofers", preferably located at the sides or rear of the room. The Dolby specs *require* every screen, surround, and overhead channel to extend to 40 Hz, using bass management as necessary to meet this goal. I don't know how many dub stages use 40 Hz vs. 30 Hz subs for surrounds, being that the front channels are still run without subs (usually extending to 40 Hz on their own). Second, cinema Atmos is not compatible with the home Atmos. This means that *all* Atmos BD and UHD releases are dedicated home mixes or masters. This work is done in small rooms that mimic home theaters, and I expect all of them use bass management for the overheads and surrounds (and probably fronts too). The Atmos home format (or rather the equipment that implements it) does not support separate surround subs, so bass managed bass all goes to the one SUB channel, which will almost certainly extend to 30 Hz or below. Also keep in mind that sound design is a separate step from the mixing. Most sound design is being done in small room studios where capabilities (for better or worse) are very different from the dub stage. Some sound designers might even work using sealed subs and small enough rooms to get significant ULF. But even if they can't hear the ULF on the track, it's not fair to say that it wasn't part of the sound design. Whether the ULF was part of a recorded or synthesized effect, it originated as part of the design process. The only question is whether the designers/mixers/directors are able to hear what they've done.
  4. I assume the after is with the @maxmercy BEQ applied? Either way, I strongly disagree that the fact that the surrounds have lower average levels means their shape is not audibly important. Average level depends on short-term level, duration, and frequency of effects. Most action happens up front and in the center, so it's no surprise that the surrounds have much lower average levels than the fronts. Yes. However, without additional information we can't tell how the LFE channel was used. There's no guarantee that content that's mixed to one or more front/surround channels gets sent to LFE too. Many strategies are possible, and you can't really tell what was done by looking at either of the PvA curves. *Of course* there are distinct effects in the surrounds! Any sound that gets mixed to the surrounds that has bass will be affected by the filters applied to that channel during production and the BEQ filters applied during playback. Unless most of that bass was copied to or sent to the LFE channel, an independent channels BEQ will recover ULF in the surrounds that an all-channels BEQ can't. The result won't be subtle for the discrete surround (and overhead) effects with sub bass, especially given the ~60 Hz filter! I do agree that the post BEQ surround channel averages look odd, particularly below 30 Hz. Why do all the surround channel curves converge to one curve below there? I would not expect that to happen. The shape of the curves and lack of finer details is also unusual. It looks like it could be garbage. Maybe insufficient precision somewhere? Are you applying the 1st order high pass at 10 Hz? And is it working correctly?
  5. How would you define "extremely large"? It's a bit hard to tell from your pictures, but it looks like the green curve for least one surround channel levels out at around -72 dB (average) below 20 Hz. To get this content to about the same level as the stuff at 100 Hz (-33 dB) requires up to +39 dB of shelf. Here's the total shelf gain for each channel for the @maxmercy BEQ correction: LFE: +20 dB LCR: +18 dB SURR: +48 dB (!) Note that these also use 1st order high pass filters (f0=3.0 Hz for LCR+LFE and f0=10.0 Hz for SURR) to remove DC noise in the track. I'm pretty sure this BEQ has it covered. Is +48 dB really an extreme boost? I don't think so. One of my other favorite BEQs, which I just watched the other day is for Guardians of the Galaxy: LFE: +40.5 dB LCR: +20 dB SURR: +60 dB (!!) As a point of note: my processor is connected to the upstream via analog, so I do have to worry about the effect of boost on the analog noise floor. With the GotG BEQ applied and no sound playing, I get periodic spurts of noise that are enough to register on my Motu 16A display ("-48 dBFS") and to light up the "signal present" lights on my amp. That's probably only ~1W actually going to the subs, so no worry there. However unfortunately, even at that extremely low level the ULF output is enough to cause one of my living room windows to make ticking noises. GRR!!! Ignoring that though, the above GotG BEQ delivers very excellent sound.
  6. By "levels", are you referring to relative level of each channel in the PvA data? If so, I disagree with your argument, at least in general. A lot depends on how the mixers created the LFE channel content. For TLJ, It could be that that the bass under 60 Hz in the surround channels just gets thrown out rather than being sent to LFE. If that's the case, then a global BEQ will likely have little to no effect on the surrounds because the ULF boost will be well below the 60 Hz roll-off. To figure this out, I guess one would have to compare the tracks, looking for some place where an effect with bass plays on the surrounds only and then checking if any of that missing bottom end found its way into LFE. I'm curious if @maxmercy has any insight here. How much evidence is there of mixers splitting content between surrounds and LFE using HPF/LPF pairs? I'd bet there are many different strategies used for managing LFE and that for a great many cases, a separate-channels BEQ will be much better.
  7. SME

    Othorn - HT capable?

    Nice ideas here. I hope that XO processor works out for you. Definitely experiment with different settings. Measure and compare if you can and have the patience. Unless you're using real weak amps on the mid/high horns, you might seriously consider adjusting the passive circuit to attenuate things more. I don't know if that would be feasible in your case, but it could make a huge difference with regard to noise while also improving safety from faults (both for the drivers and your ears). Even if you listen real loud, I seriously doubt you need sensitivity that high. It may be fun to brag about, but practically speaking it's a nuisance with regard to noise and possible safety concerns in the event of a fault. My speakers have horns covering content above 850 Hz+ that are 108 dB sensitivity without passive electronics. I have a passive circuit to cut them by 6-10 dB over most their range and I've cut the output gains for them on my DSP an additional 6 dB, meaning that I probably hit like 50W peak before clipping in the DSP, most of which is sunk into the resistors of my passive circuit. My DSP interface has peak and average indicators for each channel so I can literally watch my headroom. I listen to plenty of stuff plenty loud, and it's rare for the peak level to exceed -10 dBFS. Only with a handful of cases do I see peaks up near the top. One example of some demanding treble (at least for my horns) is the hand-held phaser shoot-out in "Star Trek" (2009) when playing at a MV in which I probably hit peaks > 125 dB SPL (with help from the subs) in other scenes. Of course, with the kind of subs you're building, you may listen to stuff way louder than I do. But seriously, those kinds of levels are not going to be good for your hearing in the long-term. I can play a good dynamic multi-channel concert video at something like "+3 dB" vs. reference level (and a generous house curve on the bass), and it sounds wonderful without even a hint of strain! The sound is so clean, detailed, expansive, and powerful. And then after 5 minutes it stops and my ears will be slightly ringing and after a couple hours, my hearing will still be altered. That's hearing damage territory, and for me, that's *enough* headroom.
  8. I should be able to see that one soon and will give my feedback. I do hope it's not as bad as "Black Panther", which as I said sounded more compressed than analog TV to me. BP had a $200 million budget, and given that "sound is half the movie", there is no excuse for this. At this point, I don't really have a problem with studios pulling dynamics back a bit for home mixes. The mid-range emphasis that results when a cinema or dub-stage is calibrated to the X-curve target makes the sound smaller and whimpier. For a long time, I assumed that mixers compensated for the lack of bass and treble using EQ boost, but I was mostly wrong about that. I say "mostly" because I think some EQ does find its way in to a lot of tracks, but it's much less than would be needed to reverse the effect of the X-curve calibration. So even after EQ tweaks, the sound in the cinema is still lacking in bass and treble and seems small and whimpy. The mixers compensate another way: using the faders. The consequence is a mix with exaggerated macro-dynamics because the mixers are pushing up the levels of the big effects for more pop. There's a huge difference, however, between pulling dynamics back a bit for a more authentic "cinema-like" experience at home and crushing the mix to the dynamic range of a typical Internet pod cast. Also, I could be wrong but it sounds like BP just got shoved through an algorithm without any scene-by-scene consideration at all. I don't have a problem with automation being used as part of the process, but at the least there should be people going through the mix and making tweaks for different scenes. It took a lot of fader twiddling to get the cinema mix the way it was. It's ridiculous to assume that a simple algorithm can reverse all of that. The sad thing about all this is that most consumers don't complain. The industry thinks that means that it doesn't matter, but it does even if consumer don't complain. Most consumers don't notice the lack of dynamics. They don't notice the lack of emotional impact and lack of connection with the actors. They don't notice that the overall movie experience is less inspiring or that their motivation to purchase media in the future is diminished. And when the sales start to drop, will the industry recognize that quality matters even when the consumer isn't aware of it? Or will they just blame the loss of sales on pirates?
  9. FWIW, I liked TLJ as a film a lot more after the second watch. It's definitely flawed, but what SW film isn't? Yes! Because the channel architecture includes the LFE channel for extra bass headroom, any film mixer that wants to use LFE in the mix must effectively use some kind of bass management when creating it. Microphones and synthesizers don't tend to spit out separate LFE tracks, and I doubt the sound designers deliver content with separate LFE either. Even if they did, it wouldn't allow optimal budgeting of headroom in the mix, so basically it comes down to the mixer to figure out how to distribute the sub bass between the different channels of the mix. I'm sure many different strategies are employed including the wacky filtering schemes seen in TLJ and many other movies. I wouldn't be surprised if a lot of the bass that appears to have been filtered from the surrounds was simply re-routed to LFE instead. This is probably why almost every one of the BEQs developed by @maxmercy uses quite different filters on each channel, and why I believe this approach is usually necessary for good BEQ sound quality. Even then, it's probably not possible to fix everything. The bass management on the production side may be applying very different filters to sound that is redirected from each mains channel to LFE. For example with TLJ, it's possible that the 40-60 Hz part of the LFE channel contains a lot more bass that goes with sounds in the surround channels than screen channels, but below 40 Hz, it's a more even mix of screens and surrounds. So any adjustments made to LFE could have different effects on the surrounds vs. the mains. Lastly, we need to remind ourselves that bass management on playback systems has a lot of problems too. Probably very few systems out there have neutral sounding bass for sounds on all 7.1+ channels of a soundtrack because of sub crossover phase issues. The more savvy home theater people know to optimize sub delay for best sub crossover response, which makes a *huge* difference, but this is only possible on one channel (i.e. the center channel for movie optimization) or some weighted sum (i.e. left+right for music optimization). How many people here or anywhere have good response in the sub XO region on their surround and overhead channels? Yes, those channels do get used for bass, and they get used a lot more now because immersive formats for the cinema specify bass management for the surrounds. I personally have the capability to optimize bass management completely and separately for each mains channel and for LFE, which is itself optimized to blend best with simultaneous content in the center channel. IMO, this should be a minimum requirement for a "high performance" home theate, but it's not possible to do this with any standard home theater processors I know of (without spending 6 figures $ at least). I doubt very many "Atmos at home" production systems have that capability, which means they aren't hearing the bass right on their own soundtracks. (Sadly, I wouldn't be surprised if many of these don't even have the sub distance optimized for best center channel response.) At least Atmos in the cinema specifies that surround and overhead channels be bass managed to separate "surround subs" located closer to the back of the room, which probably helps a lot, but Atmos for home is still essentially a 7.1+ format.
  10. SME

    Ricci's Skhorn Subwoofer & Files

    While within their limits, amps are essentially voltage control devices, so the loads in parallel won't interact via an electrical path. There may still be mechanical interactions from shared air space or external proximity/boundary/acoustic loading effects but from an electrical standpoint, the two drivers don't "see" each other. In a series configuration however, it is possible for power to be transferred between the drivers unless they're exactly the same. Any electrical or mechanical differences between the two drivers may cause energy to flow between the two, which can complicate their behavior compared to systems with all drivers in parallel. It's also possible that this interaction could couple with another interaction (say acoustic or mechanical interaction due to proximity or shared air space) in a way that leads to a feedback loop that causes unstable behavior. In reality, the differences involved are probably too small for this to be a problem most of the time. However, I believe there are always opportunities for exceptions. Some manufacturers may be more consistent than others with regard to parameters that matter. Loading drivers into a tuned enclosure (especially one with higher pressures like a horn) could amplify certain problems. Running the drivers harder where non-linearity becomes a big factor is probably likely to accentuate such problems too. But this is all really just theoretical speculation. I have no idea if any of these effects are really strong enough to cause serious problems.
  11. Other movies watched recent: "Ghost in the Shell": I liked it a lot, and it had some cool mid-bass glitch effects and a very interesting score. The bass was OK otherwise, filtered in the 20s but not too aggressively. "Jumnaji": Worst bass I've heard in a while. It had some ULF, but it had a severe 30 Hz hump and almost no mid-bass. It was loud and boomy and had absolutely no punch or clarity and almost no tactile at all. For all the attention given to ULF, I think good mid-bass is more important. Without enough mid-bass in the track, the low stuff just sounds terrible. (Edit) "Ready: Player One" with BEQ: Bass movie of the year for me. Possibly my new favorite bass movie ever. Excellent full-bandwidth sound design. Very tactile throughout. You are *there*. I kind of miss the discussion here where we gave our subjective impressions of these tracks. Maybe all of that happens at AVSForum these days?
  12. So I watched "Black Panther" finally. I played the BD 7.1 at "-1" (!), and my Denon did not indicate any dialnorm compensation was applied. I have to agree that this soundtrack has serious deficiencies caused by excessive and/or mis-configured dynamics processing. It was bad. It seemed worse than the dynamics of old analog TV. Not only were macro-dynamics largely eliminated but the compressor attack seemed to act immediately and aggressively on transiently loud sounds such as gun shots. IMO, the consequences were wide and substantially degrading to the viewing experience. The experience was passive and uninvolving. The on-screen action and actors seemed trivial and insignificant. It wasn't just special effects that were harm but also the score, dialog, ambiance, etc. The scoring involving live acoustic instruments (vs. the pop / hip-hop style music) sounded completely unnatural and irritating to listen to because of the pumping effects. In one scene, dialog alternated between a few key characters and a large crowd, both of which were rendered equally loud but of course this made the crowd sound very whimpy, and the aggressive compression really muddied their voices together. It really broke the immersion. The acting seemed quite forced throughout the movie, but I could also tell that the dialog completely lacked the dynamics which may play a big part in conveying passion and emotion. The weird thing is that even if this aggressive compression was done on purpose, perhaps to optimize for hand-held devices in high noise environments, the level of the track was extremely low. I basically had to turn things up to cinema reference level to get sound that still wasn't that loud. That's a huge amount of gain as far as hand-helds go. On most if not all such devices, this track would be difficult to hear even with the volume at max. Oddly enough though, it had some decent bass in a few places, particularly in some of the quieter scenes where there wasn't as much mid/high frequency content to compete against. The tonal balance was actually quite nice, and I experienced a surprising amount of ULF with rather strong tactile effect. A lot of ULF just makes the windows rattle and maybe shakes the floor a bit, but this was felt quite strongly in the chest area. I believe the presence of higher harmonic frequencies with good balance all the way up essential for this sensation. The ULF itself makes the least contribution to the sensation but helps fill it out and make it more real and present. It's just too bad that the good bass was totally missing from a lot of transient effects. Anyway, I have a hard time believing that the problems with this soundtrack were intentional. It's certainly possible that people misjudged the quality of the track because they were too tired or something. Or maybe people were just rushed. Maybe mistakes were made that were to difficult to correct. Still, this is bad enough that I think it'd be reasonable to ask for a correction and recall. I didn't buy this film, and have no interest in doing so after what I experienced. Still, I hope this isn't an example of Disney's future. I thought some of the Pixar films like "Coco", "Inside Out", and "Finding Dory" were fine as far as dynamics were concerned. They might have been pulled back a bit compared to typical cinema tracks, but I kind of expect that with kids movies. And anyway they still were *way more dynamic* than "Black Panther". Edit: Just to clarify with my opinions regard to discussion in some of the recent posts: I thought "Black Panther" and "The Last Jedi" were kind of at opposite ends of sound quality and performance. TLJ had very nice dynamics, it just needs high master volume to get things nice and loud. BP is about the same average loudness as TLJ was (7.1 at least) but had practically no dynamics at all, for which there's no real cure. BP really looks like a QC error as does the Thor:Ragnorok Atmos track which apparent has severe compression but only for the first half. I hope it's just these two, but I guess I'll find out how Avengers 3 is.
  13. SME

    Ricci's Skhorn Subwoofer & Files

    Do you have any real world examples of this kind of positive feedback loop occurring, causing unexpected bottoming or failure due to uneven power dissipation? Or maybe just really bad sound? I'm skeptical, but I'm also paranoid. I'm also a bit paranoid about having multiple drivers share the same air space. I opted for a solid barrier between air spaces within my dual-opposed sealed subs. You might want to consider that too if these kinds of interactions between drivers are really a problem for you.
  14. SME

    Ricci's Skhorn Subwoofer & Files

    Are these DVC drivers? I believe manufacturing variance, possibly amplified by the loading of the cabinets, could play a role in unexpected differences in series vs. parallel performance. I don't believe such difference will only one voice coil. It is when there are multiples that you get a series connection between two VCs instead of between a VC and the amp, and that is where I can imagine some unusual electromechanical behavior to arise if manufacturing variances are significant. Just an idea.
  15. SME

    The Bass EQ for Movies Thread

    Apologies for taking so long to give a review. I *finally* watched this tonight! I just kept putting it off because I've been working so hard on new/improved treble optimization. I'm still not done and was thinking I'd save it for when I'm "done", but my wife got tired me of putting it off. I'm glad I listened to her. I think this is my new favorite soundtrack, and the BEQ takes it to 110%. I watched it (BD/Atmos soundtrack) at around "-5" on the MV. (I forgot to check to see if there was any dialnorm modification to that, so it might have been less). My system is configured with a pretty generous house curve (up to 10-12 dB "hot"). Pretty much all of the bass sounded full bandwidth, and the frequency balance was excellent. At no point did the low stuff overwhelm the mid-bass nor any of the rest of the spectrum. The sound effects were very cohesive from top to bottom, and the tactile sensations were detailed, articulate and at times brutal. There were multiple jaw-drop "jump out of my seat" moments where things just went BOOM spectacularly. The surround work in this mix was a big part of experience as well. What can I say? This this a superb demo piece: for my bass capability, my overall sound capability, for BEQ as a technique, for superb sound design, etc. It is state-of-the-art. A big thanks to @maxmercy for taking the time to do this!
  16. SME

    Ricci's Skhorn Subwoofer & Files

    I've commented before that I believe punch depends substantially on the linear response of the system in the upper bass as well as the sub. I'd go even further and say that mid and high frequencies can matter a lot too. Even fairly tiny resonances in the treble can perceptually overwhelm the bass, causing several dB reduction in subjective output. Why this happens is a good question for scientists to answer some day. In any case, punch depends *crucially* on linear response, and a major fault *anywhere in the spectrum* can be harmful. Note also that native sub responses are mostly minimum phase, which means that measurements like group delay and waterfall are not really distinct from frequency magnitude response. What you'll typically see is higher group delay and waterfall decay in and around regions where frequency response slope is changing a lot, such as around the vent tune or sealed box resonance but also around higher frequency resonances that appear in the driver, port and/or horn. As such, I believe it's helpful to focus on (native / anechoic) frequency magnitude response shape when talking about linear response in general. Of course linear response is not the whole story with subs, especially when they are driven hard. In general, non-linear response is very complicated and strongly dependent on the signal. THD and power compression measurements are the easiest ways of characterizing non-linear response, but unfortunately they are not very useful. What we mostly care about is response to music which mostly involves transients and complex multi-tone signals. THD and power compression only offer small hints as to what might happen there, and there are many mechanisms of non-linearity that neither THD nor power compression adequately reveal. These include mechanisms that cause inter-modulation distortion (IMD) and noise, which are probably far more important perceptually speaking. In terms of direct negative impact on sound and tactile quality, THD is probably least important. Indeed, THD (both odd and even harmonic) is frequently added to music *on purpose* during mastering to give it a *subjectively improved* (and louder) sound. This is done especially to create high frequency content an alternative to doing EQ boost, because the EQ boost often brings out more noise and more HF harshness caused by recording deficiencies. The spectral balance of the synthesized harmonics can be very smooth, which will sound very good when reproduced on accurate speakers. (It can sound harsh or bright on inferior speakers though). I'd guess that more than half of music masters and probably most movie soundtracks get this treatment. THD added to sub bass can potentially increase tactile sensation also. In terms of equal SPL, the body has a lot more tactile sensitivity in the ~100-500 Hz range. The reason why we feel subs as much as we do is because of the comparatively higher SPL. I think the bolded part along with IMD, which I presume is often caused substantially by non-linear inductance effects, are key when it comes to distortion. Almost all music involves transients (wide-bandwidth) or multi-tone signals, and the mechanisms that influence IMD but not necessarily THD will impact reproduction of both of these types of signals profoundly. Of course, IMD depends a lot on driver characteristics too but the cabinet reshapes not only the linear response but also distortion components of the driver. A cabinet that naturally suppresses higher order overtones (harmonic or otherwise) and noise is likely to help the sub "get out of the way" of the rest of the system in the upper frequencies that are crucial to punch. Probably any sound that is produced that is not harmonically related to the signal, including noise and IMD is very degrading to sound quality, especially when it reaches into the higher frequencies where the ear is much more sensitive. One experiment I think would be interesting would be to try to strategically adding some absorption into one or more passages to try to further smooth and suppress the output of mid / high frequencies. Obviously there's a trade off as far as losing "CEA output" and possibly adding some power compression, but there may be a "sweet spot" which gives an even cleaner sound and perhaps a tad more "usable output" if one's trying to avoid any hint of audible strain. Are you saying that the systems with a sharper, more defined knee give better punch? Or that they are better in general? I'm a bit confused because sharper knees are typically associated with more internal resonance, less "control", less damping, and/or less motor strength, which seems a bit counter-intuitive. In terms of tactile sensation, the subjective emphasis of frequencies around that knee could help or hurt depending on your baseline. It's really complicated and I don't really have a formula for determining how a particular linear response will feel. I am wondering @radulescu_paul_mircea , have you tried EQing the lower part of the response of one cabinet to look like another? I'm pretty sure doing so will change the sound quite a bit, but I'm curious as to how much it closes the gap in terms of "kick" or whatever. Another thing to consider is that a sub with more extension may substantially activate room modes that were largely untouched before, and so supressing just the new room mode(s) with EQ could help clean up the sound and get the "kick" back.
  17. SME

    The Bass EQ for Movies Thread

    Zero headroom? That's not good because it implies the processing is using integers rather than fixed or floating point. This means that in addition to the potential for internal clipping, precision problems are likely to arise for low frequency filters, particularly below 20 Hz. How bad it is depends on whether it's processing with 16-bit or 32-bit integers. I'd guess that 32-bit integers might do OK except for the very low frequencies but careful testing is advised. With 16-bit integers, the results are likely to be complete garbage for ULF filters. I would not advise using ffmpeg for either analysis or processing unless you can figure out a way to force it to use floating point instead of integer processing. Even then, I'd want to know whether it applies any kind of dithering when converting from float back to integer. This is probably a pretty minor concern if the output format uses at least 24-bits but for 16-bit output, dithering is crucial to avoid raising the quantization noise floor too much, degrading sound quality particularly in quiet passages. While applying some gain before and after is a feasible work-around, you effectively give up signal-to-noise ratio in the process. A reduction of -10 dB is basically equivalent to giving up 1.66 bits of information per sample. At least reducing gain beforehand makes clip detection possible. Without upstream gain reduction, it's impossible to tell whether the clipping occurred during the processing or was already present in the original track. This fact is irrelevant to processing with ffmpeg because the clipping occurs within ffmpeg itself. This is a potential issue with any DSP; however, the vast majority of competently designed DSPs do processing using fixed or floating point rather than integers and have substantial internal headroom available.
  18. SME

    The Bass EQ for Movies Thread

    If it were me, I would probably study the source code more carefully to try to figure out what it will do, but a quick test would be to try a pair of filters that cancel each other, for example: Low Shelf gain=+X; Low Shelf gain=-X at a common frequency, where the X is some relatively big number that will cause clipping depending on how much internal headroom the processor has. After running a track through the pair of filters, the input and output should be very nearly identical unless clipping or severe precision loss occurred. If the internal processing is floating point, then internal headroom and precision should be very high.
  19. I don't have confirmation of this, but I believe that many Disney and Skywalker Sound mixes in general have more mid-range and less bass than typical cinema content. I believe this is actually a good thing, even though it may reduce apparent loudness and/or dynamics compared to most other mixes. These mixes are likely to sound better on the vast majority of audio system out there including systems optimized for music playback. However, the response of different home systems varies a lot, and the variance is not necessarily any better for "full blown home theaters" vs. TVs and hand-held devices. For example, a lot of auto-EQ calibrate to targets that are bass deficient, IMO. If the Disney/Skywalker Sound movies sound whimpy, even after adjusting the master volume up, it may be in part because the playback system is calibrated to a bass deficient target. Cinemas are also bass deficient because the lower part of the X-curve is flat, whereas a natural in-room response from an anechoic flat speaker typically rises toward the bass in the bottom. Cinemas are also treble deficient because of the steep -3 dB/octave roll-off in the upper part X-curve vs. between 0 and 1 dB/octave with an anechoic flat speaker. I personally find most of the Disney/Skywalker soundtracks to be quite satisfying on my system, which has a substantial "house curve", consistent with a flat anechoic response. I do wish the tracks were more consistent though. Even though most of the Disney/Skywalker stuff has better balance between bass, mids, and treble than typical cinema tracks, the balance within each region is often weird. A lot of this probably reflects the limits of existing calibration technology, which is an area I'd like to see improvement. While most rooms exhibit fairly similar in-room response characteristics when using anechoic flat speakers, the differences are enough that calibrating to the same in-room target in different rooms won't lead to consistent sound. In the long run, it would be good to see the X-curve standard for cinema and dub stages go away to be replaced by a truly accurate calibration standard. This would likely eliminate one of the biggest differences between cinemas and home systems. Better yet, with all the production done on accurate systems, much less EQ would be applied to the mixes overall leading to a much better result overall on *all* systems *including cinemas*, where sound is often the worst. Indeed experiments suggest that cinemas sound better with anechoic flat speakers than with the X-curve calibration, *even when playing cinema content* mixed in X-curve calibrated dub stages. Yeah, really!
  20. SME

    The Bass EQ for Movies Thread

    Another issue that may be a lot more challenging to address is proper support for the lower-end MiniDSP devices, the 2x4 and 2x4 balanced, which use 56-bit fixed point processing instead of floating point. This causes errors due to precision loss below 20-30 Hz, which become worse with decreasing frequency. These errors can be substantial even when using the MiniDSP in the mode that is optimized for low frequencies, without which any kind of ULF EQ is pretty much useless. I got bit by this problem pretty hard back when I used a MiniDSP 2x4 for in-room sub EQ. The errors are not small. Even with a floating point implementation, the precision of the floats may matter. (I haven't thoroughly tested it.) Processing the audio in 32-bit float format should be good enough, but the biquad coefficients and temporary variables may need to be double precision for sufficient ULF accuracy. My implementation does this. I don't know if the floating point MiniDSP units represent the biquad coefficients and temp variables as 64-bit or not. Likewise, I don't know how much precision the fixed point MiniDSP units use to represent the biquad coefficients. These are details that may need to ascertained by reverse engineering in order make tools like beqdesigner as accurate as possible. What about DSP built into amps? It's the same story there too, and each device may behave differently. (This is probably part of the reason many amps don't allow filters below 20 Hz.) These issues don't just affect beqdesigner but affect implementation of the BEQs posted in this thread too. It's just that the independent channel BEQs are designed to be implemented upstream of bass-management, which limits the devices that can be used to apply them to devices that probably have at least 32-bit float precision. Along these lines, it would probably be good to investigate ffmpeg to understand how it will process biquads. I took a look just now. It appears that the internal implementation supports 16-bit and 32-bit integers along with 32-bit and 64-bit float with coefficients always being represented using 64-bit float. The integer formats don't offer any headroom, and because each biquad is processed separately, clipping will occur immediately if any one biquad pushes the signal above full-scale, even if this excursion would have been canceled out by a later biquad. As such, it is crucial that ffmpeg be used in such a way that it uses floating processing internally, and of course, the result will only be accurate for the highest precision floating point DSPs.
  21. SME

    The Bass EQ for Movies Thread

    So AIUI, if one is doing a multichannel BEQ there's currently no way to generate a graph showing the bass-managed result. Doing so requires re-analyzing the track, just like calculating headroom limitations. One can of course make some assumptions and calculate an estimate of the bass-managed result, but it won't match the real thing which requires a longer run. I didn't see anything suitable for streamed processing with biquads in scipy.signal but I may have not looked closely enough. If you're already requiring ffmpeg for extraction, then it makes sense to try it for processing. Hopefully it does the job well.
  22. SME

    The Bass EQ for Movies Thread

    What you describe could be done by import/export of raw PvA data, which should be quite easy to implement. We'd also need a place on-line to host exported data so that people could enjoy the benefits you describe. At the same time, processing is still necessary in order to support key features including headroom/clipping analysis and independent channel BEQ. It does appear that ffmpeg supports generic biquads, so that is an option. However because processing is needed for "internal use" in addition to the use case of processing .mkvs, using ffmpeg may be a bit clumsy. I don't know the program well enough to say for sure. What I do know is that one probably wants the processing to support streaming, and it should be as fast as possible including using multiple CPU cores. I would suggest a simple multi-process architecture using UNIX sockets, but this thing has to run on Windows. As such, code to support the Windowsy methods for inter-process communication would have to be written as well. If there is interest, I'm willing to contribute my C/Python code to do stream processing of biquads along with a simple makefile for builds on UNIX. For Windows, someone else will have to step in, and whoever builds the Windows binary release will have to have a Windows C compiler tool-chain. If asked, I would write code for the multi-process streaming processor using UNIX sockets as well, but I don't think it would work on Windows. Edit: Actually, I think I could make a version that'll run multi-core just using the standard Python multi-threading support. It should work fine cross-platform.
  23. SME

    The Bass EQ for Movies Thread

    Forgive me if this doesn't make sense as I haven't looked at the code yet, but ... Isn't signal processing already done? How else does one preview the effect of applying filters? Otherwise, it's pretty easy. The main issue is that pure Python code for biquad processing will be very slow without help from an external library or module. I have about 40 lines of C code (including ".h" header) which does the job and can be loaded in a Python program using cffi. The "hard" part is the housekeeping required to ensure the C module gets built or distributed as needed for different platforms. Technically, this only works if the EQ is the same on all channels. Even then, there's no way to know for certain whether the EQs will cause a signal that's worse than the usual worst-case scenario (i.e. full-scale on one channel or all channels simultaneously), which may cause many systems to clip downstream. Also note that distributing graphs for each channel separately does not allow for different EQ on different channels because of phase effects. Edit: I see that the frequency response of the biquad is calculated using scipy.signal.freqz() which evaluates the transfer function of a biquad filter but is not suitable for actually applying the filter to a stream. ... With that in mind, the beqdesigner probably does not handle filters on separate channels correctly because, related to my note above, the only way to compute the combined PvA is by actually applying filters to each stream, summing the outputs, and applying the PvA analysis to the result. That's because the phases won't necessarily match between channels at any given time. This is actually a big problem because to do it "right", the entire soundtrack essentially has to be reprocessed each time an adjustment is made to the EQ for a channel. I think a re-analysis must also be performed to compute peak levels on each channel and for the combination to check for clipping potential. I believe these re-analyses are among the most time consuming part of @maxmercy's current process.
  24. SME

    The Bass EQ for Movies Thread

    So I see that BEQ has suddenly taken off on AVSForum, which appears to have partly inspired @3ll3d00d's designer software. These are very positive developments. However, I'm a bit disappointed to see that the BEQs posted to AVSForum are intended to be applied to all channels, and there appears to be little if any post-BEQ QC performed. Instead, most of these seem to involve EQing the PvA to flat and calling it enough. I even see people arguing that the quality of a BEQ should be judged "objectively" by how smooth or flat the resulting PvA is. Ugh! I fear many of these BEQs may be doing more harm than good to the track. As we know very well here, a PvA is not a reliable predictor of perceived tonal balance on a track. It's certainly informative, but is nowhere near definitive. There's really no way to know how something sounds without listening to it on a good "reference" system and making a subjective judgment. Undoubtedly, this is complicated by the facts that personal preferences vary and that it is not yet known how to calibrate different bass systems to sound exactly the same, but I don't know of a better way to deal with the problem. It's probably perfectly OK if there are multiple BEQs out there. Different people will have different insight and of course will hear different things. FWIW, I have a pretty aggressive house curve on my system which arises from my novel calibration approach based on the concept of apparent power. Curiously however, my approach leads me to a curve that tops out around 20 Hz and is somewhat diminished (by a few dB) below there. Furthermore, I've noticed that soundtracks I like also frequently have a bit less ULF than 20-40 Hz bass. They don't have a steep shelf or HPF but often they don't push levels below 20-40 Hz that much. I think @maxmercy gets this right by looking at each channel and trying to ascertain what filter/filters were used rather than just making the PvA look pretty and listening to the final result. I believe it makes all the difference. All the same, it might not matter much for most people, especially those using Crowsons. I doubt very many bass systems out there are particularly balanced, including those with very high output capability. If one's sound already leans heavily in certain directions (such as ULF over mid-bass or vibration over acoustic) then the nuances of better quality BEQ quality may be mostly missed. The ability to send a lot more content to the Crowsons may be "good enough" for most people. P.S. I expect to evaluate @maxmercy's "Ready Player One" BEQ hopefully this weekend. It looks like one I will enjoy. :)
  25. I just watched "Blade Runner: 2049" tonight. The movie dialog seemed a bit louder (2 dB?) than average, so early on, I turned it down to "-8" instead of the typical "-5 to -6". No adjustments were made for the music vs. effects. This one definitely could have used a bit more mid-range, like many cinema tracks. The bass was just off the charts LOUD, like sustained square waves from 16 Hz up. The chest throbbing on the low notes was intense. Unfortunately, the house noises were also rather severe. I will say that while I appreciate the score's tribute to the original, the original Vangelis score was *way* better. This movie was also quite good, but not as good as the original. Indeed, even though the surround and Atmos effects were nice, I think the soundtrack of the original (albeit "Final Cut" remix) was better and more immersive.
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