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  1. Yesterday
  2. dgage

    Othorn - HT capable?

    I've stayed out of this conversation since I'm not a ported or horned guy and haven't heard a horn in an optimal setup yet. But for me, multiple smaller subs is preferred and recommended to get more even bass response in the room. Regarding how bass is perceived, if we compare an 8" midbass driver with a 12" midbass driver at the same volume, the impact of the 12" is noticeable over the 8". I'm not sure what the physics are but I liken it to a towel being snapped, maybe the 12" snaps quicker because it doesn't have to move as far as the 8" driver to reach the same volume. However, I haven't noticed a difference in impact between my 18" and 24" sealed subs so maybe the impact doesn't translate at (much) lower frequencies or I'm not doing it right.
  3. Last week
  4. Kvalsvoll

    Bulding the Room2 listening room

    Testing new configuration in Room2. V6030 compact horns placed in the middle of the room, on sides. Quite close to the lp, nearfield-horn?
  5. m_ms

    Othorn - HT capable?

    Thanks, @lilmike, for trying to keep me at a level of sanity I'm glad that you point this out, as I'm obviously not very well versed into the capabilities of these tapped horn bass beasts (I imagine the ones here mentioned are somewhat more impactful compared to direct radiating 15" subs), and moreover the Microwrecker build would save me some money (cheaper B&C driver and, it seems, a slightly less complex enclosure). How capable do I really need? Good question. To begin with I believe I have the mains to keep up with the Microwrecker: Simon Mears Audio Uccello. They're 105dB sensitive (actually measured 1dB or so higher), and without too much effort I'm sure they could turn out 120dB's - at the listening position. A 15" bass in a folded horn, 2" exit (B&C) midrange compression driver in a Tractrix horn (w/5" diaphragm) + 1" (B&C) tweeter comp. driver - that's part of a recipe comfortable at high SPL's. I take it they'll come up short next the Othorn when it really starts to stretch its legs well beyond 120dB's (I wouldn't go there anyway), but other than that it strikes me as a perfect pairing (just like with the Microwrecker). The Uccello's also, and importantly, have a rather large air radiation area, so in that regard as well I believe they're a good match with these tapped horns. Oh, and not least: they're all-horns and dynamic like few. I don't much care for most horn-hybrids, and I've heard a bunch.. Next, I guess my main goal is subscribing to the "bucket load of headroom" mantra, not least inspired by you guys, and how this translates sonically to more moderate and typical listening levels at 80-90dB's or so. I'm guessing at some point more headroom becomes irrelevant - like, when exceeding 30dB's - and it has been pointed out to me already that the Othorn may qualify in that regard in light of my specific needs and circumstances, like blowing those poor sparrows out of the sky with ship cannons. Occasionally I do go to somewhere between 100-105dB's at the listening position, and so if we strive towards no less than 20dB headroom the Microwrecker is not "shooting above the goal," as I see it - actually it just meets it (as if it weren't enough). However, if we look past max. SPL's the question could be posed, again, as to how the bass is actually perceived at more "normal" listening levels. How would a 21"-fitted tapped horn sound compared to a 15" ditto? Would one be able to register the increased size in driver diameter (and better power handling) via the Othorn as something that would create an even more effortless and impactful "feel"? I don't know. Maybe I'm asking the wrong questions. I do look forward to your upcoming builds! I'm tempted to halt my decision to see what's coming from your hands.. Thanks again for the feedback /Mikael
  6. lilmike

    Othorn - HT capable?

    A Microwrecker in a room of that size will blur your vision and modulate your voice when running at full throttle. I know, because that's where I have mine, and currently, I am only running the one sub, not even placed optimally. Still, I've measured peaks of 120 dB at the MLP. Be honest with yourself. How capable do you really need? Will you have mains that can keep up? I currently don't, not by a long shot, but I am working towards that. No matter which direction you go, there will always be a bigger, badder sub out there, always. There will be even bigger and badder ones right on the horizon. I am working on several...
  7. m_ms

    Othorn - HT capable?

    Hello, everyone -- Time for an update. I haven't yet initiated my proposed sub build, but could be about to soon. My old main speakers were sold about a month ago, and I have allocated some funds for the sub project. My initial plan was to build two of lilmike's Microwreckers, but I fear my 20 sq. meter listening room will be too crammed with a pair of those, so I'm considering only one sub. Being that the overall physical imprinting of the Othorn is about equal to the Microwrecker and that it houses a bigger driver for more "impact," I strongly consider the Othorn instead - even though it may sacrifice a few hertz at the bottom end. Now I know some here would rather have me looking into a non-horn solution, vented or otherwise, but I'm on a horn mission, so please bear with me in considering the following: The Othorn is build around the B&C 21SW152 (4 ohm version, I guess?), but at a downtick in price the sibling model 21DS115-4 has been mentioned as a viable alternative as well. I gather even with the 21DS115-4 I'll have loads of headroom, but is it worth saving some bucks to go with that driver instead of the 21SW152? Here in Europe the 21WS152 is only about €100 more expensive, so not a deal breaker as such, but I'm still considering the 21DS115-4. It's also 3dB's more sensitive, not that it might matter at all. If money isn't an issue I take it the recommendation is for the 21SW152. However, is the 21DS115-4's only drawback that it has a disadvantage at extreme outputs where its smaller voice coil will start heating up more rapidly, or is there a more general disadvantage, irrespective of SPL, that would have people favor the 21SW152? In principle price is not an issue for me; I just want the most capable solution. The being said I might as well save €100 if for all intends and purposes the 21DS115-4 will do just as well in my situation (and invite my girlfriend out for dinner for the saved bucks). I mean, I wouldn't use the Othorn in an outdoor rig at >75% of its performance envelope. Please chime in. /Mikael
  8. Kvalsvoll

    Bulding the Room2 listening room

    Same as the Marantz and Denon I tested, it is very likely they share the same processing. Note that it is the DIFFERENCE (17.5ms/20ft) that is interesting, as this is the number that defines how much delay is possible on the closer speaker to make it match the farther.
  9. 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).
  10. 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.
  11. Kvalsvoll

    Bulding the Room2 listening room

    Maximum speaker delay in processors/receivers - a critical property, which is usually not sufficiently described in the manual or product presentation. Anyone know the limits for different types, brands? I seem to remember this issue has been up before, but oooohhh.. using the rest of the day searching will not happen, and new information and new models may be available. The problem: Getting the timing correct is crucial for high performance sound quality. For systems with front main speakers and separate bass system that means to delay the main speakers so that they sum correctly with the bass system in the frequency response AND IN TIME/PHASE. On most processors this is done by setting a distance on the different speakers. Typically, you set the front to the measured physical distance, and end up adding several meters for the bass system (subwoofer) - THIS WILL DELAY THE FRONT SPEAKERS. If you know what you are doing, you set the delay using measurements, so that timing and phase gets as good as possible to achieve. If you read on-line guides and audiophile magazines and pay no attention to how things really work, you set the distance for the bass-system equal to the physical measured distance, and conclude that subwoofers always sound kind of sluggish and is best switched off for music. Since you are smart, you want to do it the way that actually works to get better sound, and end up seeing that the distance entered can be quite huge. And in some cases it may be possible to reach limitations of the processor in use. Obviously this is a no-go limitation for a processor, so if you have an installation that you know will require large delays, you want to choose a processor that satisfies this requirement. You want to see a specification for this number. But this number is not in the brocheur or manual, it is not in any "test" performed by on-line or paper magazines - because the don't even understand why this number is important - so the only way to know is if someone have found the data. My contribution: Denon/Marantz processors, AVR: Max distance difference 6m / equal max delay 18ms. Devialet amplifiers: 20ms. Hypex DLCP and my SA-700 amplifiers: 15ms. (Though not relevant on the sub amp, becuase it is the mains that need delay.) Onkyo processors, AVR: ??? Some readers now realize I need those numbers for the Onkyo.
  12. Those curves look vaguely familiar... Great results for all your hard work Brian.
  13. OK. Maybe I should finally start updating this thread after so long. Sorry guys, I was just really busy, and still am. Writing takes a long time, and I've always just put that on the back seat since more time writing = less time improving speakers. Here's a sneak peak of what's coming up. Since this is data-bass, here's a preliminary 1m ground plane output compression measurement of a single speaker without a limiter on. Once a proper limiter is done, I will be extending the sweep all the way to 115dB.
  14. Earlier
  15. 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.
  16. Kvalsvoll

    X-curve compensation re-EQ

    Speaker tuning with radiation pattern, on-axis response, power response and then add in room acoustics - which will be more or less an unknown parameter for a speaker designer. This is difficult and complex, and has huge impact on perceived sound - in contrast to amplifiers, dac's, all the nonsense products. I also believe that this field has not yet been fully discovered, there is still more to learn and find out. Experiments focusing on how perception of sound relates to the technical parameters are key factors for improvement. Horn speakers sound different from trad-hifi partly due to fundamental differences in radiation pattern. It is impossible to make them sound equal, because tonal balance depends on what sound is being reproduced. If you tune for flat and equal steady-state, the transients will sound different because decay profiles are different.
  17. 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.
  18. 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.
  19. Kvalsvoll

    X-curve compensation re-EQ

    @SME, you are aware that loudspeaker manufacturers tune their speakers ("voicing") according to power response and estimated typical room acoustic properties, and have done so for decades? If you look at on-axis measurements of typical hifi-speakers, they often have a response that deviates considerably from flat - look up some of the well regarded speakers on stereophile. This is because the radiation pattern changes with frequency, and by adjusting the response the "sound" can be changed into more balanced and neutral, but the measured charts seemingly show a "defect" speaker. Since most speakers have a pattern omni at low f and then narrowing towards higher f, they will radiate more power at lower frequencies. This causes some problems if you design a speaker with radically different pattern - a speaker with flat fr and flat power will sound different from these typical once, and since music is mastered for use with the typical speaker and room, it is necessary to alter the reposne of the flat speaker into something that gives a perceived tonal balance more similar to those typical ones.
  20. maxmercy

    X-curve compensation re-EQ

    So what would you do if you had what most people on forums have available: limited budgets, limited placement options, some room treatment, REW measurement capability and limited parametric/IIR correction capability? Would you mainly focus on minimum phase problems in the LF and largely ignore MF/HF, or how would you go about optimizing a system with limitations like the above, given what you know now? I ask not only because that's the way my system is set up, but many others with miniDSP or other DSP components who have had less than stellar results with the on-board AVR 'room correction' products. JSS
  21. 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.
  22. 3ll3d00d

    X-curve compensation re-EQ

    Yes that is part of the audiophile style marketing they use. You will also find not many measurements are available to verify what it is actually doing and how well it performs. FWIW I think https://patents.google.com/patent/US8094826?oq=8094826 is their patent, it seems pretty generic to me though. To be fair I have listened to their high end kit and it is good so there some substance there, it isn't just audiophile blather.
  23. Kvalsvoll

    Bulding the Room2 listening room

    Compact Horn, basically a rear loaded horn with resonance chamber for extended usable range.
  24. Ricci

    Bulding the Room2 listening room

    What type of alignment are those? TH?
  25. MemX

    Ricci's Skhorn Subwoofer & Files

    Would one or more of the Bossobass amps be suitable if the Clone amps are not up to it? http://bossobass.com/Bossobass.com/Home.html There is a description of the A14k on the Raptor page: http://bossobass.com/Bossobass.com/Raptor Systems.html but the Amplification page itself is giving me a 404... http://bossobass.com/Bossobass.com/Amplification.html I've not seen @Bossobass Dave for ages on here, though - is he still going? Is he still selling kit??
  26. 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.
  27. 3ll3d00d

    X-curve compensation re-EQ

    fwiw this is basically the way roomperfect is marketed (albeit with a healthy dollop of audiophile voodoo mixed into the language used)
  28. 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.
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