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SME

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SME last won the day on August 8

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  1. SME

    Ricci's Skhorn Subwoofer & Files

    Thanks for this post! It is excellent and gets to the heart of matter. It's not necessarily that the sims are wrong but rather that they don't tell the whole story and miss important parts. I especially concur with the quoted passage and agree that smooth operation over a wide bandwidth is a good idea. Subs make plenty of sound above the XO point, and that sound can have a surprisingly big impact on overall quality. Problems there can make distortion sound a lot worse. For example, even fairly high harmonic distortion may go unnoticed by listeners if it is fairly uniform vs. frequency, but if certain notes have spikes in harmonic distortion due to horn resonances and whatnot, you'll get notes that tend to stand out, which is a lot more likely to be audible. More generally, I think people worry may too much about the sub range to the detriment of the higher bass frequencies. Bass isn't something that only happens below 80 Hz. Indeed, a lot of the sound of bass instruments including synths is in the low hundreds of Hz, and sound up to at least 4-500 Hz contributes to tactile sensation. Hot subs sound much better when the rise in response is spread out across the mid/upper-bass and low-mid range. Having augmented output capability up to something like 150 Hz rather than merely 80-100 Hz is probably nice to have in many situations.
  2. SME

    Porting the room - flat to 6hz

    Even if the filled window is behaving like an SBA dead wall, that is nowhere near sufficient to explain flat response to 6 Hz. I think the hypothesis described in the OP is reasonable. The mineral wool probably has the effect of making the port lossy, but I doubt it lowers the tune much if at all. I wonder if even more output at around 6 Hz would be seen if the mineral wool were removed?
  3. SME

    Ricci's Skhorn Subwoofer & Files

    That room looks small. Are you in an apartment? And you're planning on building a second horn? Wow!
  4. SME

    Ricci's Skhorn Subwoofer & Files

    A crazy idea I have is that there may be a psychoacoustic boosting effect around the area where the response is beginning to roll off. So something tuned at 40 Hz will tend to emphasize 40 Hz a bit because of its roll-off shape. Something tuned lower will tend to emphasize a bit lower.
  5. I haven't watched "Black Panther" yet, so no comment. On the subject of loudness vs. MV, I have noticed more soundtracks recently with the dialnorm offset set to something other than 0. A non-zero dialnorm offset causes the AVR/processor to attenuate the soundtrack uniformly when playing it back. The attenuation can be counteracted by turning the MV up, but most people just set the MV at their preferred level anyway. As for why dialnorm offset is used, the intent behind dialnorm was to ensure that different content with different dynamic ranges always played back at a similar loudness. The problem is that, dialnorm rarely gets set correctly, even though it's been around for many years. Most people have no idea when it is active. For example, almost all AC3 tracks on DVDs used the default value of "-4", which attenuates playback by 4 dB. OTOH, the DTS mixes that shipped on a few DVDs often specified "0", meaning they played 4 dB louder than the DD track. Later when BDs came out, most tracks were being authored with "0", so lots of people probably got fooled into thinking the BD tracks were "more dynamic" when instead they just played louder. Truth be told, there's little difference in sound between a DVD AC3 track and a DTS-HD track unless the latter is 7.1 instead of 5.1. Anyway, as I said, it appears that Disney is starting to actually set it to meaningful values. I think the idea is to move toward mastering home releases with consistent track-to-track loudness after dialnorm, where the loudness is measured in LUFS. As far as I know, there are no positive dialnorm offsets, so the most dynamic one can go is offset "0" which I believe is standardized to -31 LUFS. I think a typical home theater release would be at like -27 LUFS, which is about 3 or 4 dB more dynamic than a standard TV mix at -23 LUFS or -24 LUFS, depending on where you live. I believe movie trailers shown in the cinema are standardized to -21 LUFS. As I mentioned above, -16 to -20 LUFS is recommended for Internet content that's likely to be played on speakers on portable devices. IIRC, both "Thor:Ragnorok" and "Star Wars: The Last Jedi" DTS-HD tracks had dialnorm "-2". So if the mixers are setting it correctly, it means the tracks are up at -29 LUFS. That's pretty good. Some of the Atmos tracks may be even better.
  6. Well this story sure is a mixed bag: https://www.forbes.com/sites/johnarcher/2018/07/09/hey-disney-stop-getting-your-star-wars-and-marvel-soundtracks-wrong/ Oh how terrible! One has to actually adjust the master volume. No doubt Disney has put out some pretty lackluster mixes lately, particularly with regard to surround/Atmos usage. Though, excepting the odd one or two with major issues (Thor:Ragnorok Atmos mix / SW:TFA limiter) I don't know if I would call them "weirdly compressed" or complain about "cramped and muffled" mid-range. And bass extension? Sure most releases are filtered, but the same is true for other studios. And the same was true before this: Undynamic, really? True, Avengers AOU wasn't exactly an award winning mix, but it also wasn't full of nasty clipping like so many others before it. I and others thought it had plenty of dynamics after compensating for the lack of loudness using the Master Volume. Either way, it seems like the author's motivated to complain primarily because of the lack of loudness. Umm. Last I checked the bass on GOTG2 is just as filtered as the others, and while the track is slightly louder than the overall average for Disney (at the cost of some dynamic range of course), we're talking about maybe 2-3 dB difference? Not according to recent comments by @Infrasonic. I don't visit the cinema that often, but I would concur. In fact if a soundtrack, pretty much any movie soundtrack, sounds better in the cinema than at home, then I would argue that the home playback system is not suited for making these kinds of judgments. Ahah! They are mixing for TVs and phones. Except, not really. While I'm sure that Disney and others are mixing with an eye toward smaller devices, the low level of the content in the mixes has nothing to do with achieving that goal. The problem with limited output playback devices is that they have limited maximum gain, and their users find they can't turn up their master volume enough to hear the dialog clearly. So all else the same, such users are better off with mixes in which the content is *louder not quieter*. In fact standard loudness recommendations for streamed content of -16 to -20 LUFS are *much* louder than the typical loudness of cinema content, which comes in roughly between -27 to -31 LUFS but varies more because there are no hard standards. (Disney's UHD/BD/DVD releases probably come in very close to those numbers.) Also note that most streaming involves only 2 channels, so 7.1 soundtracks (including home Atmos) effectively have another ~11 dB headroom or ~14 dB if the LFE channel is counted. So the complaints about lack of dynamics and lack of bass are just silly. Indeed, even excluding the TV/laptop/phone viewers that don't own a "home theater" system, probably fewer than 0.1% of viewers have systems that are capable of playing these soundtracks with good sound quality (i.e., properly sloped in-room response curve) and at the reference level used in the studio (e.g. 79-82 dBC for a small room) without overloading. But never fear! Linked to the article is a Change.org petition: https://www.change.org/p/walt-disney-improve-audio-dynamic-eq-lfe-bass-levels-in-disney-home-video-releases Dear Disney: please fix the dynamic eq lfe bass levels. Or something. Umm, yeah, I think there was some bass there. And yeah, the cinema sub sound quality was above average overall FWIW. Compared to the cinema? No way! The cinema sounded like a faint murmur compared to the slam and roar experienced at my home. (OK, the BEQ probably helped some, but still!) And that was on the DTS-HD vs. Atmos track, which was apparently even more dynamic. Of course, I only have 5.1 instead of full Atmos, so maybe that's my problem. Anyway, I think Disney deserves some criticism for some of its recent mixes for sure, but it's mostly not technical stuff as far as I can tell. By my judgment, dynamics, tonal balance, and bass levels and extension are fine compared to the stuff that comes out of other studios. Rather, it just seems like overall production quality has diminished a bit, especially in the last year or two. The weird dynamics issue on Thor:Ragnorok (Atmos) was probably not intentional; it was just sloppy. It's less clear that this was the case for the -3 dBFS limiter on SW:TFA, but these are one off things. More notable is the lackluster sound design and unimaginative surround usage in a lot of recent films. Why, it's almost as if the crews are experiencing franchise fatigue or something. Though more likely it has to do with excessive cost cutting and tighter production schedules. Either way, it would help if Disney took criticism for the stuff that actually justifies it. Lack of loudness is not one of those things. We can only hope that Disney is *not* listening.
  7. SME

    Bulding the Room2 listening room

    My Denon 3313CI AVR offers up to "60 feet" distance (a bit over 50 ms, in terms of delay) and a maximum difference of "20 feet" (~17.5 ms).
  8. I am a witness to the immense time and effort @lowerFE has been putting into his speakers. I'm one of the first people he talks to to share his successes and the various wacky things he discovers that don't work the way he expected. There's lots of the latter, and I must emphasize that this means he improving on a lot of "stuff that actually matters" and developing a very deep understanding of the process. Anyway, I'll let him elaborate on everything he's been doing when he gets around to it --- probably when he gets temporarily bored of making his stuff sound better. Most of our interactions followed his visit to me back in October, when I was able to hear the speakers in my living room. Neither of us would argue that they sounded as good as mine, but considering the size difference and the fact that fact that my system was "optimized" for the placements and room, his speakers very much held their own. Of particular note is how effortless and yet full and extended the bass sounded at moderate levels, which is quite a feat for a small speaker. So given what I know that he's done since October, I look forward to my next audition. From what I know he's done, he has made significant improved to practically every aspect of the design since my last listen.
  9. SME

    The Bass EQ for Movies Thread

    I watched this one today. The movie itself was kind of funky, but I appreciated it for what it was and would watch it again. Spectacular video and audio on this one. Center channel dialog track sounded a bit full in a lot of places, but otherwise, the sound was excellent. (At some later date, I might try to re-EQ the dialog to fix it up.) The Atmos surround effects were some of the best I've experience, despite my sitting outside the MLP on a 5.1 system. The sub bass effects sounded more distinctive and somehow imaginative than a lot of movies. I was often surprised by how laid back the mix was. It seems to fit the feel of the movie. I watched at "-4" and probably could have gone higher if my subs weren't already at their limits. (If only I had the floor space and the money, I could easily make use of 8 x UH-21" in here.) Anyway, thanks for the BEQ on this one. The restored ULF seemed to contribute a lot to the sound effects, and not just heft. In fact, most of the bass in the movie seemed quite tight and transient, except where it was obviously not supposed to be (some scenes with music). I've noticed before that the restored ULF can actually make the sound seem tighter and more precise.
  10. SME

    X-curve compensation re-EQ

    Yep, I am absolutely aware of these things. Almost no speaker sounds right with a ruler flat on-axis response. Of course, that doesn't mean that manufacturers are tweaking their speakers on the basis of actual power response measurements, which are difficult to do correctly. I'd bet that almost every speaker design gets tweaked at least once, based solely on subjective listening tests before being finalized. My guess is that most speakers start up with fairly flat on-axis responses as a baseline and then get tweaked from there There was a time when more speakers were designed specifically for flat power. As I understand it, there was a kind of rivalry between people who believed the flat on-axis was optimal and those who believed that flat power was best. JBL was a big proponent of the former. Allison was a big proponent of the latter. Allison did some remarkable work designing speakers for placement against walls and trying to optimize passive signal shaping to maintain nearly flat power under those conditions. I grew up listening to one of Allison''s designs, which I remember fondly. In the end, flat on-axis essentially won in listener preference experiments. However, flat power may be have lost in part because such speakers almost always had up-sloping first arrival responses, for the reasons you note above. Or maybe it's not about the up-sloping first arrival response but the *lack of slope in the power response*? Personally, I'm pretty certain first arrival is still very important. Otherwise, one would expect horns to sound rather dark compared to cone-and-dome speakers with the same on-axis FR, which is definitely not the case. In any case, I have no doubt in my mind at this point that power response counts for a lot.
  11. SME

    X-curve compensation re-EQ

    Let's first assume that I have access to some future version of the tools I've created to do this sort of thing. The tools would probably take the place of REW for measurement but could export WAV files that one could import into REW. These tools use measurement data to obtain accurate estimates of first arrival and in-room power response, from which they use to suggest optimal correction curves. From there, the problem would be to generate filters that could be implemented on more modest DSP hardware than I'm using now. Some of the more recent MiniDSP products look pretty capable as far as offering FIRs along with a greater number of biquads than earlier models. With mathematical optimization for the biquads, it may be possible to implement some pretty good corrections on the more capable MiniDSPs. A complication is the fact that so many different MiniDSP hardware configurations are possible, each with somewhat different constraints. Of course, my tools are not currently available for wider use, and I don't know when they will be or how affordable they'll be. So what would I do without those tools? That's a tough call. An approach similar to Lygndorf's for estimating power response may work OK, but there are many details to be mindful of. Randomizing the measurement locations in 3D space is probably critical, and I'd opt for more than 6 measurement locations if possible. The different measurements should probably be level-matched before averaging. Then there is the task of smoothing power response while keeping first arrival approximately flat, and it's important to prioritize fixing low Q features over high Q features of the same magnitude. Contrary to common belief, low Q features are more audible than high Q features. To get this right requires smoothing, and unfortunately, REW lacks power smoothing, which I believe is the best kind to use for this. Most of the other stuff can probably be done in REW in some way or another. What to do without a good power estimate? At that point, falling back on existing advice is best. Start with speakers that have an excellent native power (and on-axis) response. Place them far from walls or install them flush-mounted, if possible. If not, some absorption on the part(s) of the wall(s) closest to them may or may not help. EQ can be experimented with but it's hard to know where and how much to apply. Modal resonances may be a relatively easy problem to treat with EQ. The peaks tend to stand out in the in-room measurements, and they can be particularly annoying to listen to. Be sure to confirm they are actually modal resonances by looking at several measurements throughout the room. Be willing to experiment to get the right EQ filter. In my "optimized" system the modal resonances are only partially suppressed in most of the in-room measurements. Unfortunately, none of this helps with lower Q stuff, including broad bass shape, that's critical to getting the best sound. One can try to optimize broad shape by ear, which is what I did up until I applied my latest tools, which finally gave me a superior result than I could get by ear. Sorry that I can't give a better answer than this right now.
  12. SME

    X-curve compensation re-EQ

    I would say the patent gives a lot of details, even if it does not give a lot of specifics. It's way more informative than the marketing literature is, at least. The gist of the method(s) described by the patent is as follows: Fundamentally, EQ is optimized so that a measurement or averaged cluster of measurements at/around the listening position are made flat. However, the measurement/average is potentially smoothed and filtered before the correction is computed, and the correction is subject to constraints. No additional information is given about the smoothing. That's too bad because the particulars of the smoothing potentially has a big impact on the quality of the result. They do describe performing measurements at 1/12th octave resolution, which is pretty "meh" as far as these things go. That's the good news. The bad news is that the measurements rely on pure tone sine waves, so the results will have a lot of uncertainty when used to estimate the shape of the whole spectrum. Before the correction, the response is pre-conditioned using filters. The filters are derived from idealized models of normal and expected in-room frequency response and/or power response. They essentially take the place of a non-flat EQ target curve by reshaping the listening position response (or average) before calculation of the EQ parameters. The models discussed in the patent include: a high frequency directivity roll-off (expected more in power response than at listening position), low frequency room gain, and low frequency high pass / roll-off. The parameters for the pre-conditioning filters may be calculated using the measurement itself (HF and LF roll-off models) or may be based on foreknowledge of certain characteristics (bass room gain model). The end result is very similar to developing and applying a target curve that is customized to the room and speakers and possibly to listener preference. The constraints are derived by taking additional measurements around the room and averaging them. This average is intended to be a power response estimate. (Based on my experience, the quality of this estimate is probably very crude.) The power average is also pre-conditioned with filters, just like the listening position measurement (or average) was. The pre-conditioned power average is then inverted, and the inverse is used to develop upper and lower bounds on the EQ. The idea here is to prevent the EQ from doing anything too extreme (in either direction) to the power response. === There are many likely benefits to their approach. The pre-conditioning filters effectively customize the target curve for the room and speaker. I'm a big critic of EQing to a one-size-fits-all target curve, so this is a big plus to me. Their models are probably over-simplified, but I would guess that they make a big improvement over one-size-fits-all. My guess is that they stumbled upon this approach when trying to figure out how to re-EQ speakers close to walls. The use of a power response estimate to develop EQ constraints is also a big plus. This is a solution for the "hidden resonances" problem I described above. However, I don't believe their's is a very good solution for a few reasons. The worst part is that their measurements rely on pure sine wave test tones. In anechoic measurements, pure tones tend to be OK because there are no acoustic effects (except involving the speaker itself). The impulse responses aren't very long except for high Q low frequency resonances, which are fairly rare in competent designed speakers. In-room, it's a totally different story. Acoustic effects will effectively contribute a lot of uncertainty to the measured values. It's because a measurement at a narrow frequency is a poor statistical guess about what FR is doing in a fuzzy region around that frequency, which is the kind of information that's needed to do the correction. Their tech note indicated that pink noise was rejected as a test signal because it didn't "reach deep enough into the impulse response". I'm pretty sure that's wrong. The pure tones certainly engage the late parts of the impulse response but they don't reveal them to the measurement system any better than the pink noise does. The most likely reason they chose pure tones over pink noise was because the signal-to-noise ratio was better. Unfortunately for them, the "insight-to-signal" ratio using pure tones is not so good. Moving to REW-style log sine sweeps and applying appropriate smoothing would probably lead to big improvement in the performance of their system. I would argue that they are also worrying too much about the listening position measurements. Practically everybody assumes that in-room frequency response will tell them something about what they'll hear in that position, but the reality is that things are a lot more complicated than that. The information is there, but the primitive analytical techniques in widespread use don't do an especially good job of recovering it. Power response is a major key to the puzzle, but it's not obvious how to determine speaker power response using in-room measurements. As I said, the approach taken in this example is quite crude, and would still be crude even if the measurement methods were improved. To follow-up on my comments about room vs. speaker correction: The best "room correction", by a long shot, is the space between your ears. Listeners are extraordinarily well-adapted to listening in environments with early reflections, and experimental evidence suggests that listeners *hear better* in the presence of early reflections. As such, the goal is not to correct the room but to correct the speaker for the room and let the brain do its thing. The big problem is that no one has a particularly good perceptual model to apply to in-room IR/FR measurements to assess what listeners will actually hear. Almost every product relies on EQing in-room response to some kind of simplified target, and each such product adds its own twist to make it unique to make it not actually suck. The use of a power response estimate to constrain the EQ is very good idea, but the methods described in the patent for doing so are rather obvious and primitive, IMO. Still, I don't doubt that it sounds pretty good, especially compared to other room EQ systems. It's possible that they have improved their analytical methods some beyond the patent. However, the tech note suggests that they are still using the primitive pure tone measurements, and I believe that holds it back considerably.
  13. SME

    X-curve compensation re-EQ

    Thanks for the reference. Since I had not heard of roomperfect, I decided to visit their web site to try to learn more about their product and the marketing language. The main web page was sparse, but I found a bit more info on this article: http://lyngdorf.com/news-what-is-room-correction/ To their credit, they seem to dedicate effort to improving in-room speaker power response. However, as best as I can tell from their description, their methods are nowhere near sufficient to obtain an unbiased measurement of in-room power. There is another, potentially larger problem: Here they seem to be implying that one of their goals is to preserve the (presumably desirable) unique sound signature of the customer's speakers. This tails in with the marketing of the product as a "room correction" product rather than a "speaker correction" or "EQ optimization" product. So how exactly do they achieve this goal? How do they distinguish between the speaker's "specific and desirable product performance" and the room's degrading influence and only correct for the latter? They don't explain how, and I very much doubt they have any way of making this distinction in the first place. I believe the industry of "room correction" products has a kind dirty secret. To the extent they work at all, it's mostly about speaker correction not room correction. I can offer two possible reasons why the technology is marketed this way. First is simply ignorance. Room effects cause the vast majority of variation within in-room frequency response measurements, leading naive engineers to erroneously conclude that the room is overwhelmingly to blame for poor sound quality. Second, "room correction" probably sells much better than "speaker correction". Most audiophiles don't want to be told that their speakers (possibly costing 5 or 6 figures) are flawed. It's much easier to point to the huge variations in-room response measurements (+/-20 dB !!!) and sell people on the idea that *their room that is holding back their speakers*. I realize I have emphasized room-dependent problems in my discussion above, but my focus is more general. IMO, the best description for my approach is "in-room speaker EQ optimization". It is intended to take the place of two activities which are typically performed separately: voicing and crossover design (usually performed using anechoic measurements) and so-called "room correction". I personally could care less about preserving the "unique sound signature" of particular speakers or other "audiophile" products. I just want the best sound possible from my system.
  14. SME

    X-curve compensation re-EQ

    I do not follow... How can one correct something that is inherently designed into a speaker (power response), if it does not depend on speaker location? Or are you talking about correcting power response depending on the speaker's location in the room? Is this why incredible amounts of headroom are needed? Allow me to clarify my statement above: The speaker's *total acoustic power output* response does not depend on the location of the listener. It does, however, depend on the location of the speaker. With that said, there's no reason not to correct power response issues that occur in the native (anechoic) response of the speaker. In fact when analyzing in-room measurements, it's not really practical to distinguish issues that depend on the speaker location vs. those that don't. Very few speakers have ideal native (anechoic) power response either. Even if the mid/woofer drivers measure very cleanly when on an I.B. and the cabinet doesn't have any panel resonances, the cabinet shape still contributes variations. In practice, placement and treatment options are almost always limited and serve as only partial solutions. Often these options also involve compromises. For example, most speakers sound their best in a room when aimed a certain direction, but this makes a flush-mounted installation difficult, especially with overheads and surrounds. Also, any absorption removes valuable reflected sound energy as a side effect. (This assumes small rooms where decay time reduction is rarely necessary, except at the modal resonances.) Even in the most ideal circumstances, there is likely to be benefit from DSP if it's done well. That leads to an interesting question: To what extent is it possible to work-around acoustical problems using DSP? The answer obviously depends on the capabilities of the DSP and quality of the algorithms used to compute the filters. So what if one uses the best possible DSP capabilities and algorithms? Well, no one can really answer that question because the best possible algorithms probably haven't been invented yet, and the best available algorithms may not be very good. Conventional wisdom says that DSP cannot fix most acoustical problems and can only optimize sound at one location or perhaps compromise for handful of locations. This reasoning is entirely valid from the view that the goal of DSP correction is to fix "problems" in the in-room response measurement. The name of the company that produces the Dirac Live software refers directly to the goal of most room EQ system: to achieve an in-room impulse response measurement that looks more like Dirac Delta, which corresponds to a perfectly flat frequency response. But what if all this conventional wisdom is wrong? Empirical evidence suggests that anechoic chambers make terrible listening rooms, yet they are the closest to achieving an ideal Dirac Delta. OTOH, 99.9% of the listening we do in real life is in rooms with significant reflections. Maybe the reflections don't harm sound quality at all. Maybe the real "problem" with reflections is that they confound our ability to measure and correct the speaker itself. Or that they help reveal the less-than-ideal power response of the many speakers that otherwise look great when measured only on-axis. Anyway, my recent experience suggests that DSP (done well) *can mostly work around* the kinds of acoustical problems that affect subjective sound quality and not just for a single listener location. The filters do usually require extra headroom to implement. Therefore, changing speaker placement or installing absorptive treatments may be beneficial in conjunction with DSP to reduce the amount of boost required. Some boosts will still likely be required to overcome limitations of the speaker itself. Edit: The text editor widget glitched and wouldn't let me add another paragraph to my response! One caveat to add to all of the above is that a human listener can only ever *estimate* the power response of the source from the available information. For the most part, these estimates can be remarkably accurate, perhaps excepting particularly pathological rooms or listener locations very close to untreated boundaries. However, the accuracy of the estimates does deteriorate for bass in small rooms where the sound field becomes highly structured. As such, the conventional wisdom that one can "only optimize sound at one location or perhaps compromise for a handful of locations" does apply to bass to an extent, particularly below roughly 150 Hz depending on room size. However, this is not nearly as bad as one would expect by looking at in-room FR measurements. The subjective variance is still much less than what is observed directly in the measurements. Several strategies may be used to reduce subjective variation of bass. One example we're all familiar with is to place subs in multiple room locations. I suspect this may be beneficial even if the placements are not optimized, e.g., to cancel modes. Of course, independent filters on multiple subs has the potential to do even better. When I did this before, I was optimizing in-room frequency response, which didn't sound nearly as good as I'd hoped. I expect that optimizing for a superior objective will deliver much better subjective results.
  15. SME

    X-curve compensation re-EQ

    I can see this, by correcting response at one location, you create problems and ringing at others. Yes, but this is true *not just at other locations* but also at the location where the single measurement was taken. The ear and brain use information from reflections to hear through most acoustic effects that are particular to a single location. How can 'precise' correction of a reflection be 'corrected' for many locations? The peaks and dips will occur at different freqs depending on location from the speaker.. A *reflection* cannot be corrected for multiple locations using DSP, and often one should probably not try to "correct" reflections because they actually facilitate the hearing process. (Note: some reflections may still be degrading for some frequencies.) To oversimplify just a little: What we wish to correct is the effect of one or more *boundaries* (among the many other things) on the speaker's *total acoustic power output* response, which does not depend on room location. Yes. Your anecdote was puzzling to me for a while, but not anymore. It makes perfect sense now. Of course that doesn't mean that EQ (even high "Q") can't improve sound. Rather, the problem is that we misinterpret smoothed FR graphs, which don't really show us what listeners will hear.
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