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Everything posted by SME

  1. Maximum horn compression ratio?

    The BL^2/Re is the ticket. The Qes or Qts are influenced by Kms (suspension stiffness) and Mms (effective moving mass) as well. Higher values of either parameter increase Qes and Qts while affecting efficiency at either frequency extreme. Higher Mms reduces efficiency at the high end. Higher Kms reduces efficiency at the low end, although the K of the air spring in a small sealed box can overwhelm the effects of Kms in the final product. Comparing the IPAL-18 to the HST-18mk2, the HST has approximately double Mms but only about half the Kms of the IPAL-18. So the influence of these parameters on differences in Qes/Qts is kind of a wash. The IPAL-18 will offer much more upper frequency efficiency overall. The HST-18, while having quite a bit less BL^2/Re will still hold its own pretty well below the resonance frequency in an I.B. or large sealed box. As the box size is reduced, the HST-18 loses the benefit of its more compliant suspension and the efficiency advantage of the IPAL-18 will widen. However, note that the stronger motor of the IPAL-18 has a downside, at least in a sealed box. The stronger motor will tend to increase the back EMF around the system resonance, requiring higher voltage (but not power) to drive it. This problem is substantially mitigated by its very low resistance, provided that the amp is capable of operating stably under such conditions and capable of delivering the desired amount of power within the lower impedance parts of the response within the desired bandwidth. And of course, the IPAL -18 may require some signal shaping (EQ) to achieve the desired in-room response, but this may be true for either driver depending on the room. On the other hand, a ported box will tend to increase the sensitivity of a driver for frequencies near and a ways above the port tune, which can offset much of this effect. A horn will tend to increase sensitivity (and efficiency) even further, and the higher back EMF may even be more desirable for a flatter native response.
  2. Bulding the Room2 listening room

    I listen to all sorts of stuff, and I'd be hard pressed to identify much that's really "demo-worthy". I'm rather more inclined to demo stuff that's familiar to other people, especially if they bring their own music, and some of the few tracks I have with more significant dynamic range. For that purpose, I like a lot of the stuff done by Chesky Records. I think I already mentioned "Clark Terry - Live at the Village Gate", which I find to be a very immersive and dynamic recording of a live jazz performance. Another album I have that I routinely go to is "Ana Caram - Amazonia". IMO, it's an all-around excellent latin jazz album, and the vocalist is superb. There are other Chesky recordings of Ana Caram out there too, but I have heard them. Another one, recommended by Bob Katz is "Pacquito D'Rivera - Portraits of Cuba". The brass on that one will really test the high frequency section of your system and your ears too. All of the above are, IMO, very dynamic recordings that I typically play in the low negative single digits or even at "0" and were likely mastered 10-12 dB louder than typical loudness war stuff. They also use relatively minimalist micing techniques and deliver a very nice sound stage, if you speakers are able to recreate it. I also have a fair number of decent classical recordings, but I need more. Unfortunately, most classical (as well as most big band) just doesn't have enough dynamic range to do it justice. I've noticed a lot of more recent recordings that have moderate dynamic range (i.e., comfortable playback at "-5") but have annoying clicking artifacts from the digital limiters in use. Some are much worse than others, but being familiar with the artifact, I find myself hearing it a lot more than I want to in things that otherwise sound fantastic like the "Star Wars: TFA" score (on the BD,; no idea how the CD sounds, but it's probably way more compressed). Reference Recordings is a label that offers some fairly dynamic stuff including some organ music with real extension, but I find a lot of their stuff to be unrealistically hyped at the top end (cuz, it sounds more hi-res that way?) and often distorted in the livelier parts due to limiters that just don't sound as natural as good-ole-fashioned analog saturation. Other than that, I like a wide variety of genres: classical / symphonic of various eras, jazz (especially latin jazz), folk, rock, blues, 80s (mostly wife's stuff which is a bit more alt than pop), etc. I also love a wide variety of world music, from celtic to reggae to raga to middle eastern. Genre-crossing fusion stuff is great. I like certain kinds of EDM a lot, but am real picky. IMO, there's a lot of garbage including most of the stuff that was popular back when I was young. I really like goa/psy-trance EDM though. Once I discovered it, I kind of stopped bothering with most of the other stuff, which just sounded slow and unimaginative in comparison (IMO). I did like some DnB stuff, but I didn't really pursue it as eagerly as I did goa/psy-trance. I kind of got more into the culture of the latter, even though the "scene" was practically non-existent here in the states until Infected Mushroom and Shpongle and whatnot started making the rounds. By then, I was too busy doing grown up stuff to really party more than once or twice a year, and those parties were tending toward more of a crazy dark-psy sound with even higher BPM (compared to older psy-trance which was quite fast at 145-180 BPM+) and sometimes chaotic, glitchy rhythms. A lot of that stuff was likely from artists that never managed to publish as widespread piracy pretty much killed any potential for commercial sales. More than anything else, I listen to electronic psy/ambient/dub stuff, mostly stuff that grew out of the goa/psy-trance EDM movement, which tends to be infused with more world music influence and is more focused on structured bass beats than pure free-form space stuff. A lot of the music I have I collected back when this stuff was unheard of here in the states, but of course it was only a matter of time before these sounds branched into other forms like dub-step, which went on to influence pop music of today. The vast majority of this electronic stuff is not especially dynamic, and overall sound quality is definitely variable. On the plus side, the mixes often contain a diversity of instrumentation, both acoustic and electronic; they tend to be bass heavy or bass driven, even though extension below 30 Hz or so is pretty rare; they make ample use of phase-based panning including outside the front stage; and they use *lots of reverb*. All of these are qualities benefit from a system capable of accurate reproduction. I just don't tend to play most of this music at especially high levels. Most was probably mastered at close to "-12" or "-14" like typical pop music, and most of my listening is fairly casual in which the volume is more likely at "-20" to "-30" or less. If I'm in the mood for a more involving experience, I'll probably push things up closer to the mastered level. There are always exceptions to these things. A few odd tracks have impressive extension and/or slightly lower loudness including a lot of stuff from "Infected Mushroom". The compilation album from Interchill Records "Gathering the Tribe" has 20 Hz extension and is a bit less loud than average, and I do sometimes crank things up into the minus single digits on that one. If you find yourself liking this kind of music, try to get a copy of the BD "Shpongle - Live at Red Rocks" as it's a real gem. (I know I've already mentioned that one.) The album releases I have by that group are rather loud and a bit rough sounding in the highs, but that BD is super smooth and has excellent dynamics and very convincing replica of the ambience of the venue, which I happen to live near and have been to on several occasions. Yes. I noticed the album had a characteristic sound similar to other selections by the artist you posted. Thankfully it's a lot less offensive than it was in previous iterations of my system EQ config. At "-6", it was definitely a lot louder than I would normally listen. For a more typical but "fully-involved" listening session, I would probably choose closer to "-12" or "-14", which I presume is where it was mastered. And for more casual listening, I'd use a similar level to most of the rest of the music I "commonly" listen to. At those levels, I doubt I'd have any problem with the sound, even though its slight edginess would still be noticeable, just as it is with some other albums I own and like. Indeed. I am more confident in the accuracy of my system now than ever before, but significant differences will always persist, no matter how close to perfect our systems may get. Do keep in mind that sound quality problems are a lot more forgiving when the level is lower. On my last config, I found anything about "-13" to be pushing it a bit, so being able to push up to "-6" while keeping things mostly comfortable says a lot about how even "small refinements" can make a big difference in a system's sound quality. I'm really surprised by how loud I can play a lot of stuff now without really experiencing discomfort. I thought the trumpet had fairly significant distortion, which was probably done on purpose. Though for me it just seemed way too overbearing. Maybe my ears were just tired from the high levels they'd endured before it. After "Piety", I played the song from "Yello" and then immediately followed that by "Metallica - One (live)" from "Through the Never" on YouTube. The last selection I played at a full "0", and it was not especially harsh even at that level. However, when the music stopped, I noticed temporary threshold shift (TTS) for a few minutes afterwards, which was not apparent from the previous selections. From what I understand, TTS is a warning sign of potential damage, which suggests that sound harshness or unpleasantness does not necessarily correlate with damage. It's possible for sound to be very clean and comfortable and yet be quite damaging if listened to for long periods of time. Thanks for posting all the selections. Most of that stuff I haven't heard. Unfortunately, I don't have access to Tidal, and Bandcamp playback recently broke for me for unknown reasons. When I did my replay, I had to rely on YouTube, for better or worse. Even with YouTube, things are dicey these days as Firefox recently stopped playing well with Linux audio. It's not the fault of Linux but simply programmers who don't try to understand how the system is designed. In the case of Firefox, it's fond of opening the audio device and keeping it open, preventing other apps from using it, so I have to divert it to a fake device until I want to actually play sound with it. I'll probably resurrect my Roku some time soon, which I had disabled because it hijacked and completely broke my HDMI chain, presumably due to buggy CEC support. At least it'll do YouTube, and if the phase of the moon is right, I can ChromeCast from my phone. Alas, I don't get why the young'ns rave about the convenience of digital streaming. For me, it's nothing but inconvenient. I'm much more likely to buy a DRM-free download or even an old-fashioned redbook CD than to try to stream. Mind you, I was streaming music all the time back in the late '90s and early '00s when the protocols were open and DRM hadn't infected everything. Now everything is locked-down and walled-off and the artists still aren't making any money, except from a few services that actually care like Bandcamp. Hence, I'm real sad that Bandcamp broke. I guess people on this site can relate to the fact that almost all streaming TV/video services only offer audio in stereo. How lame is that? They're offering us 4k (albeit at bitrates that deliver quality that's still inferior to BD) but only 2 channel audio? Get real.
  3. Bulding the Room2 listening room

    I revisited this album today, after my latest system configuration using FIR filters and offering significantly smoother highs and upper mids than before. This album is definitely a lot more listen-able (at high level) than it was before, but the qualities that made it sound harsh to me are still present. It has a distinct upper-mid/high push, and many if not most HF transients are clipped. The pumping with low frequencies seems even more noticeable now. All of these are among the usual tools (or side-effects of those tools) used by mastering engineers to achieve a louder sound while preserving a subjective sense of dynamics. I listened to most of it at "-6" ("-7" really, if I were calibrated precisely with pink noise to 85 dBC), and at that level, plenty of content still exceeded my comfort threshold. I can't imagine listening at "0" except for maybe single songs on a one-off basis. At "-6", some stuff was uncomfortable. The cymbal-like sound in "Fog", while relatively clean (for the album), was more upper mid than I wanted to deal with for the full length of the song, so I turned it down. Likewise, the trumpet in the last song is just plain harsh, even at modest volume. If you're listening to this at "0", then you must either have a tilted system response or a much higher tolerance for loudness than I do. In terms of overall loudness, I perceive this album to be no less loud than the majority of content these days. I don't know if it's meaningful to ask whether or not the heavy pumping with the bass was intentional. I am fairly certain however that the bass impact in this track would not have been possible without either the pumping or a substantial drop in loudness. The bass hits by themselves likely used every bit of headroom on the track. For comparison, I also pulled up "Yello - Stay" right afterwards. The vocal on that track is definitely heavily distorted, and while I thought it sounded "bad", it did not induce discomfort to my ear. I would say that the vocal distortion intentional or else it was a terrible recording (or both). The rest of the sound was quite smooth and, in some ways, a tad less loud than the Flashbulb album. The transients were not clipped as aggressively if they were clipped at all. So other than the vocal, I did not find it to be an especially harsh track. The issue (if there is one) is in the vocal recording or processing. By comparison, "Piety" has harshness in the mix/master. "Piety", of course, had no shortage of highly distorted samples as well, but these were almost certainly made to sound that way on purpose. With that said, it's OK if we disagree on these things. Every system sounds a bit different, and I know this probably better than most, having heard my own system with easily hundreds of different EQ configurations. What one hears is a consequence of the source material and the system linear (and to some extent non-linear) response combined. Flaws in the reproduction system can either enhance or suppress flaws in the source material. Though, I'd also add that flaws are more likely to enhance one another than suppress one another. Clipping is perhaps among the most egregious flaws because it introduces a broad spectrum of high frequency energy, which tends to emphasize every HF linear response flaw in the downstream system and often cascades to cause more clipping. It's something audio engineers should go out of their way to avoid because its' effects on the sound are so unpredictable from one system to the next and are typically worse on less capable systems. Anyway, I do like the powerful and extended bass on "Piety" and wish more content was done this way, albeit with more headroom and less pumping. Musically speaking, it's not quite a style I prefer, but I do listen to a lot of stuff that's similar. It's pretty rare to hear this kind of music with any extension beyond 30 Hz or so, and it really does make a big difference, even at more modest levels, IMO.
  4. The Room Here is a picture of the front stage in my living room as of August 2016: And here it is with the TV moved off the wall and the new center channel installed: Here are some pictures of the rear of the room as of September 2016: Regrettably, I did not take a picture of the front before I replaced my old cabinet with the racks or the front left/right speakers Hsus with my SEOS-15 TD12M prototypes. In case anyone is wondering, yes, this space is very acoustically challenging, but it has its upsides. Speaker / Room Calibration Through the years I have experimented with a variety of methods of speaker and room EQ. In my current approach, I mostly ignore the room, except for the subs, and I attempt to primarily correct the first arrival of sound. I aim to make the first arrival *mostly* flat. To analyze only the first arrival, I use the frequency dependent window feature REW with a crude 1/3rd octave resolution. This is roughly consistent with the bandwidths of the ear/brains filters. It's long enough to allow for delays at the crossovers but not so long as to allow inclusion of room reflections that the ear/brain could theoretically isolate. Note that this particular FDW length may not work well with other speakers or other rooms. YMMV. I say I aim for "mostly flat" because I use a bit of sub bass boost (boost of the bass in the first arrival, actually) and because I use configurable high frequency adjustments to optimize playback for different content. Because the HF adjustments are configurable, I leave them disabled (flat in the treble) in the measurements below, but these measurements *do* include the sub bass boost. A flat-ish first arrival in a listening room that's not acoustically dead tends to result in a frequency response that is sloped up toward the bottom end. Below is my frequency response for left, right, and center with 1/3rd octave-smoothing: Note that the responses look rather messy and exhibit about 8 dB difference between bass and treble. Note that with high frequency adjustments, the difference tends to come in closer to -10 dB @ 10 kHz vs. 20 Hz. Note also that the center channel response appears very hot in the mid/upper bass. This is because because of speaker/room interface issues that I currently have. The speaker is very close to the bare wall behind it, and this causes most of the sound of those frequencies to disperse diagonally and to the sides instead of forward. As such, the first arrival is relative lean. Now look at the first arrival responses, using FDW in REW with 1/3rd octave resolution: These responses appear rolled-off at the ends because these frequencies are somewhat delayed (ends of the bandwidth) and escape the influence of the filter. Apart from this, they are very smooth and flat, except for the aforementioned sub bass bump. The center is a bit messier in the mid/upper bass again. I could have made it ruler flat at the MLP here, but I opt instead to compromise a bit for off-center seats. Indeed, think this may sound more balanced, to the extent that the brain uses information from early reflections to ascertain the content of these lower frequencies. I'm not exactly sure at this point. The configurable high frequency adjustments are of three different kinds: distance compensation, upper mid-range tilt, and X curve. The measurements below depict the effects on the high frequency response of my center channel from using -0.75 dB of upper-mid tilt (blue line) and several degrees of distance compensation (red lines) on the high frequency response of my center channel: Distance compensation mostly affects frequencies above 8 kHz or so. Almost all recordings need this compensation, in part because high frequency sound naturally rolls off with distance. I chose to model this roll-off on actual physical data of distance roll-off, and it seems to be working out quite well. There is no standardized distance for monitoring music or theatrical recordings, and furthermore, a lot of music is monitored on near-field monitors that implement some kind of roll-off of their own. The consequence is that the optimal roll-off varies substantially by content. For mid range tilt, I currently use a Q 0.5 @ 2 kHz high shelf filter with variable gain. I'm not sure this is really optimal, but it works pretty well for me. Most stuff sounds good either flat or with a tiny bit of treble reduction. Rarely do I need more than -1 dB. I'm not sure why this is necessary other than perhaps precedence. Monitors in days past likely had just a bit of tilt, and the precedent stuck. A lot of monitors include a HF trim switch that engages a tilt like this. Of course, every design is slightly different, and many include additional > 8 kHz roll-off here as well. Lastly, I have X curve correction, which is used only rarely for movies that need it. Movies mixed on a dub-stage tend to use speakers behind an acoustically transparent screen, and this almost always introduces high frequency roll-off of some kind. Instead of compensating by making adjustments to the monitoring system, tended to boost treble in the mix to compensate. The X curve was intended to standardize the high frequency roll-off in the playback system so that mixes would translate, and it was based on the roll-off that was commonly observed at the time. (IMO, it largely failed because it measured pink noise energy response instead of first arrival SPL.) AIUI, dub stages largely ignore the X curve these days opting instead to simply accept the faults of the screen but also avoid hyping the high frequencies too much with the understanding that doing so may make the mix way too bright in some settings. In any case many movies, particularly stuff from the 90s and older, do benefit from some additional high frequency reduction, and the X curve shape is probably the most generically useful. The Speakers Click this image. It is an animation of the box assembly sequence: The cabinets are separate, and I'm building a left and right pair. The other bass box is visible in the background, surrounded by a bunch of clamps. On the bottom is an AE TD12M-4A in a 1.5ish cuft box with an aggressively flared 4 inch port. The cabinet is 1/2" B/BB plywood with double thick front baffle and pairs of shelf style braces running horizontally and vertically (4 braces total). The design is somewhat noteworthy for being under 9" deep. On the top is an SEOS-15 horn with a Denovo DNA-360 mounted on it. The horn cabinet is 1/2" B/BB and is not braced. It is slightly wider and deeper than the bass cabinet, partly because I started them before I started the bass cabinets and changed my mind and opted to make the bass cabinets smaller. These are currently up and running with fully active crossovers and EQ provided by a prototype version of my custom DSP software, running on a Linux PC with a Motu A16 interface connected via USB. Each mains driver is powered by a channel from an Emotiva XPA-5. The amp is rated for 200W/300W into 8/4 ohm with all channels driven, but I expect to "effectively" get 500-700W peaks for the 60-120 Hz range due to the resonance above the tuning frequency, the very low DC resistance of 3.2 ohm, and the amp's beefy power supply that'll be hardly tapped to run the horns. So far, I am very pleased with the performance of these speakers and active crossover. Here is the a recent measurement of the response (no gate, 1/6th octave smoothing) of each driver playing separately and in unison along with the subs. (Note that I'm tweaking the response all the time. I'll try to keep this image current, but it probably won't work out that way.) This measurement was taken with a -30 dBFS RMS sine sweep and a calibrated mic. The pink noise calibration signal (-30 dBFS RMS pink noise band-limited to 500-2000 Hz) reads at about 73.75 dB. I zoomed in on the data a lot, which makes the response look ugly, but in reality I'm +/-4 dB or so with 1/6th octave smoothing everywhere except for the big dip at 165 Hz. Even though my room has some treatments, I still have some significant acoustical problems involving the floor and ceiling, which accounts for the unevenness. I could have EQed much of that out, but for the time being, I'm concentrating on EQing the speaker response and avoiding acoustics, and I have purposely slanted the response through the mid-range to obtain a flatter power response. The crossover filters, 2nd order HPF for the horns and 4th order LPF for the woofers, are centered at 950 Hz. Note that the woofer and horn aren't completely in-phase here. This is an experiment, in which I have purposely mis-aligned the drivers in time to try to steer the lobe above or below the MLP, putting the MLP closer to the edge of the lobe. Then, the response must be boosted to avoid a dip, and this increases power response too. That's a good thing because directivity typically goes through a maximum where two drivers are crossing over. The idea here is to avoid that directivity peak to achieve both smooth direct response and smooth power response through the crossover region. My measurements suggests that the strategy works, but it's not clear how this affects the sound at off-axis locations that may be at a slightly different vertical angle. The cross to the subs is at 110 Hz for now because that works best with the filters on the subs as they are now. In the long run, they will be blended with the subs over a wide frequency range to optimize multi-seat response. This means, I am also planning to build subs that can player higher than the traditional 80-100 Hz as well. This is the first part of a larger project to upgrade all the speakers and DSP in my system to fully DIY-built solutions. I have started building a center channel of identical design, and I have drivers for new subs and surrounds drivers on order as well. Edit: Added details about speaker/room EQ configuration and in-room response.
  5. Thank you for your kind words. This process has been enormously challenging and time consuming, but results (so far) are immensely rewarding. I never imagined sound of this level of quality was possible, in this space or otherwise. I remember when I first moved into this house in December 2012 and set my system in the living room for the first time. The sound was so disappointing. My original plan was to try to finish part of the basement for a dedicated room, as soon as I could afford it. However, the basement options were full of ugly compromises. In one area, I would have 14 feet of width but would have to deal with ceiling obstructions with only 6 feet overhead clearance. The other part offered a full 7 feet of headroom but only 10 feet of available width. I had to come to terms with the fact that the basement options would be suboptimal regardless and decided to try to make the best of the living room space instead. Today, I could claim to have a set up that approaches "world class" performance, all while leaving the living room largely functional, albeit with lots of weird looking panels and diffusers. Thankfully, my wife has been very accommodating. Her skepticism toward acoustic treatments melted away once she heard the difference. She also happens to be quite the bass lover, lucky me! ... Some day I need to update the first posts on this thread to describe my "current" configuration. Right before @lowerFE's visit, I migrated my speaker DSP configs to use FIR filters almost entirely. I also modified the crossover to 850 Hz LR8 (acoustic). The FIR filters are much cleaner and more precise than the mess of biquads I was having to maintain. The horn/woofer crossover is also linear phase, which I opted for not so much for sound quality improvement but to eliminate group delay that confounded my tonal balance calibration method using short FDWs. The result provided a significant improvement, albeit not as dramatic as some changes in the past. Still, it was worthwhile enough for me to demo with the FIR filters, despite the fact that the bass still needed work. So @lowerFE was able to hear the speakers sounding as good as they ever have, but the bass was not as good as I think it could have been. In fact, I ended up making substantial changes on Saturday night, between his visits. On Saturday, the main/sub XO was linear phase, and I ended up redoing everything to minimum phase XOs and less aggressive shaping to reduce pre-ringing. That was kind of a hard lesson for me, which is that pre-ringing really does bad things to bass transient response and tactile sensation. The problem was most obvious to me when listening to the Danley fireworks. I could actually perceive the pressurization before the bang happened. Even with those changes, some pre-ringing persisted and is present in my current config. I don't know how perceptually important that is though. Since @lowerFE left I've EQed down the 70-100 Hz range a bit, as it was subjectively too strong, but the bigger change was to move my bass boost from being centered at 70 Hz to being centered at 155 Hz instead. I decided to try to better mimic the floor gain from a "typical floor standing speaker". I had tried bass shelves at higher frequencies like that before, but it seemed to work a lot better this time. The extra mid bass really brought more punch and overall loudness to the table. Now I'm trying to figure out how to reduce pre-ringing further while maintaining smooth frequency response, keeping excess group delay in check, ensuring coherent summing across multiple channels, and doing all of this at every seat location. It's a remarkably complicated problem, and while I have powerful DSP to attack it with, it's not at all obvious how best to apply this capability. I also have a problem of a rattling window pane (at around 60 Hz, unfortunately), so I am trying to reduce the bass build-up in that corner to keep it from rattling as much. I intend to eventually try to optimize using an automated algorithm, but automation is useless without a precisely defined objective. And in the long run, I expect I won't be able to get the results I want with the equipment I have. I still intend to replace the MBMs I have. The open question is *where* I'm going to put the new MBMs. I can put some of them behind the sofa like the old ones. I can also put some of them on top of the subs, between the subs and left/right mains. (The "pseudo-line" approach.) And I can put some up on the shelve above the TV, adjacent to the center channel. The locations behind the sofa are starting to fall out of favor with me because it's difficult to avoid pre-ringing problems. In fact, I can't really avoid pre-ringing in the dining room and kitchen areas when using behind-the-sofa MBMs without using multiple switchable DSP configs, which I'd like to avoid. So I'm curious if I can get away with MBMs on the front stage only. I think the approach has potential, given how the center channel measures. That is something I will investigate in due time. Some time, I might start a thread about bass phase response / group delay. It seems to be a substantially neglected issue with regard to system optimization and may have a strong bearing on tactile response performance. While it seems counter-intuitive that minimum phase crossovers may (often) be superior for mains/sub crossovers, minimum phase systems actually appear to have the properties we want most. We want as much energy as possible to arrive at the start of the impulse. Too much positive excess group delay, and energy does not arrive until too late to contribute to perceived impact. (Post-temporal masking effect.) But any pre-ringing has the effect of shifting the perceptual reference point, the "start of the impulse", to a place where there's very little energy at all. (Pre-temporal masking is very weak.) So what achieves these goals? For a particular magnitude response, the minimum phase response maximizes the amount of energy in the initial impulse. I suspect that this is what's needed for the best tactile "kick".
  6. Replacement AVR / processor

    @Kvalsvoll, I was talking about 2 channel systems with full-range speakers without subwoofers. They don't generally need delay because the low frequency drivers are co-located with the rest of the drivers. This includes speakers using either passive crossovers to the LF section or active crossovers (with built-in amplification). Some speaker / amp combos can extend as low as most subs go. Either way, the integration is relatively trivial. It could be said that using subwoofers, in separate room locations, solves one problem but creates another. It (partly) solves the problem of poor in-room response in the sub frequencies at the locations that are otherwise ideal for the speaker. It creates the new problem of integrating the speaker and sub response, which typically affects a region of frequencies that is crucial for reproduction of bass in music, around 60-120 Hz. I'd argue that, for the vast majority of music, accurate reproduction of those frequencies is far more important than the extra octave or two of extension that people using subs chase after. Nevertheless, I'd bet that the vast majority of systems using subs have serious frequency response problems in that range because this integration is not at all trivial. I'm not saying that using separate subs is inferior to using standalone speakers. A system that uses separate subs will absolutely out-perform a system using standalone speakers, *if* they they are configured optimally. Rather, achieving the optimal configuration for a system with subs is not at all trivial. It's hard enough to do with a single sub, and enormously more complicated with multiples. Most consumers don't have the knowledge or equipment to achieve even half-decent results, and even the more advanced consumers struggle to get "good" results. Count me among them. I have more DSP capability than just about anyone on these forums along with subwoofers-only response that looks almost "picture perfect" , yet I'm still trying to find the best strategy for integrating my subs + MBMs with my mains.
  7. Replacement AVR / processor

    Yes, I did generalize for a tapped horn, and now that I think about it, I might not be correct even for that case. Several 10s of milliseconds sounds very high for "room acoustics" effects. A full cycle at 60 Hz is 17 ms. If you are delaying more than that (in addition to distance and "internal" effects), then you are probably adding unnecessary group delay, which likely impacts transient response sound quality and slam. FWIW, I've been studying this problem quite intently lately, trying to improve integration between my speakers and subs. Unlike most people, I have practically unlimited DSP resources to throw at the problem, where the only real practical limit is latency. I would say that the processing capabilities built into AVRs and most processors are woefully inadequate for achieving an optimal outcome. The"THX "LR4 sub/sat crossover" is largely fantasy that rarely occurs in real world conditions. The best that most people can do is a brute force evaluation of different XO frequencies and sub delays, where typically response is only optimized on one channel and at one seat. Yet even this effort requires more sophistication than most users are capable of. (Readers of DataBass and some of those who read AVSForum are obvious exceptions.) No wonder a lot of people prefer bass from 2 channel full-range speakers vs. subs. While the in-room "placement" of the LF drivers in such speakers is non-optimal, the XO is (ideally) optimal for that placement. I've noticed that good anechoic-flat full-range speakers, when pulled far enough from walls, can deliver impressive slam; whereas many sub systems including many with big horns or many large drivers struggle in this respect. My recent experience suggests that phase (or rather group delay) effects are more important than most people realize. And it's not what people think. I.e., a ported sub isn't necessarily sloppier than a sealed sub, though that obviously depends on the competence of design. Such effects are largely minimum phase. (A good thing.) Rather, it is the excess group delay, which arises from crossovers and distance differences that appears to be important. Pre-ringing in particular seems to really kill tactile slam, and it should be noted that FIR filters are not the only way to introduce pre-ringing into a system. Pre-ringing can arise merely from placements and/or delay settings. Any situation in which sound from a sub may reach the listener before sound from a speaker potentially involves pre-ringing. Rooms with rear subs are likely to exhibit pre-ringing for rows behind the one used for calibration. What's not at all clear is where the perceptual thresholds lie for hearing and feeling of pre-ringing effects. Anyway, I still have a lot of work to do here, and at some point, I may try to do some more formal testing of excess group delay effects, including pre-ringing, as this information would be very useful for optimizing sub systems for multi-listener environments.
  8. Replacement AVR / processor

    I do, but I don't count. IIRC, my channel trims are +3 for the mains, which allows me to run at up to +5 MV without clipping for those odd tracks that need higher than theatrical reference level playback. This minimizes between it and my DSP. I still have the sub out set to -12, even though I technically could do -7 while retaining the same headroom, and that's because my bass management is done downstream in my DSP so the "sub out" only has to pass LFE. If I need more gain in the system, I can boost directly in my DSP, which can easily clip any of my amps. That what I usually do for those occasions in which I want to "boost the sub". That's pretty rare for me as well. I demoed some dub step at ref level with a +/- 6 dB tilt for LowerFE when he was visiting. It was funny because at first he was surprised to see my cones moving with music that was "not so loud". Then he loaded up the SPL meter in REW on his laptop, which showed continuous output in the one-teens with occasional peaks clipping the UMIK (> 120 dB). I gotcha. I feel like I've asked this question before. (head smack) So the delay is 1/4 length at tune? So 12.5 ms for a 20 Hz horn? I can see where that'd be a problem.
  9. Replacement AVR / processor

    Output of 13 VRMS is very nice. Can you set the Marantz trims to -12? IIRC, these units barely pass WCS with bass management using -12 trim and MV "0". However, all bets are off if you have Audyssey enabled. It's unfortunate that they did not incorporate enough digital headroom for Audyssey processing on top of everything else, but it's not something I use, except for Dynamic EQ in "Audyssey Bypass L/R" mode. The 20 ms delay limitation is a bit weird. On my Denon 3313CI, I am able to set the "distances" as high as 60 feet (~52.8 ms), and I have confirmed that it does indeed delay the video output appropriately. This is very useful to me to accommodate delay in my downstream audio DSP. However, the difference in distance between any two speakers cannot exceed 20 feet (~17.6 ms). So it would seem that the AVR can buffer up to a few frames of video but cannot buffer more than 20 ms of audio. Priorities, right? Anyway, as long as all the speakers are far away, the Denon/Marantz should still work for large rooms. It just has problems when some speakers are very close vs. very far, or when some outputs are processed with much higher latency than others. BTW, I've never heard of horn subs needing a lot delay to integrate properly. Can anyone comment on that point?
  10. Replacement AVR / processor

    I've read a lot about the Emotiva being glitchy and prone to occasional pops and other issues. Maybe they have made some improvements, but I would avoid them with the knowledge I have now. I also recall that their pre-outs don't provide much voltage headroom. I vote for the Denon Pro, or if you change your mind and want more features, go with the Yamaha or one of the Marantz pre/pros. Edit: infrasonic, do you know how many volts the Yamaha can put out before clipping?
  11. Replacement AVR / processor

    I've been very happy with my Denon AVRs (2112CI followed by 3313CI, IIRC), apart from paying for lots of features I don't need. Things I do care about are low noise floor (check, even with unbalanced); predictable relatively bug-free operation; no foul sounds or turn-on/turn-off thumps, even during power outages; minimal ULF roll-off; low operating temperature, etc.. The consumer AVRs can put out 4 Vrms, though that spec is not published anywhere I'm aware of. Presumably the pro version should do at least that much. (The published level specs appear to be for the inputs rather than pre-outs.) I do wish they let me turn off the amps in this model (3313CI), which would help more with heat. The ability to turn off the amps requires a more expensive model. But all-in-all, I believe the Denons are some of the "least bad" among AVRs and are generally competently implemented, which is what counts the most. I hear way too much about AVRs from other vendors that have all sorts of weird problem including unexpected thumps and other untoward emissions. Those kind of things bother me a lot, especially using a high gain system where a bad signal output could be extremely jarring. If you pick up that receiver, let me know how it works for you. I may consider it in the future. Though, I may just opt for one with the extra features. The unbalanced outputs on my existing Denon are very clean, and the noise floor on those outputs is much lower than the noise floor of my Emotiva amps, despite using balanced connections for those.
  12. Which 18" for 3way SPL AND extension

    Oh yeah. I totally forgot that John can do a "PB" variety. Email him and ask him where the dust cap resonance lies. I doubt any other vendor can/will tell you that. I think that may be the winner for your app. The extremely low and linear inductance is hard to beat when you are trying to cover 15-350 Hz.
  13. Which 18" for 3way SPL AND extension

    For the low extension, you may want to pass on the AE TD18H. I don't know how bad it will be, but it will probably leak a bit around the phase plug. The IPAL does look quite peaky. The TSAD and BMS look to be the best choices. How high do you want to be able to cross? My UH-21s are quite smooth up top, and don't break-up until 650 Hz. I believe. But they do roll-off quite a bit above 100 Hz. The shorting system in the TSAD may be superior to the BMS, but I don't know for certain.
  14. JTR Speakers Captivator 212Pro Discussion

    Cone flexure occurs at lower drive levels too, but it should remain linear at low drive levels. Many materials with elastic properties (metals, polymers, composites) exhibit some mild non-linearity at higher stresses, but this should manifest as a gradual increase in distortion with drive level. Where the stress-strain relationship changes abruptly, the material is usually permanently damaged. This is could involve elastic deformation (i.e., bending of metal, which does not return to its original shape afterwards) or fracture. You say that the driver is not hitting suspension limits, but if it's being pushed "close to Xmax", maybe it actually is, even if you don't expect that it is based on simulation data. Let me suggest a mechanism for why this could occur even if one believes the driver is being operated within its limits. More typically, driver excursion is only high at low frequencies. At frequencies well below resonance, the motor force primarily acts in the direction of the driver motion and is primarily opposed by the suspension (including box air spring, in a sealed box). When pushed to the extremes, the drop off in motor force tends to limit the excursion vs. power applied. In contrast, at high frequencies the motor force acts opposite to the direction of driver motion. It acts in conjunction with the suspension forces to slow the driver down and reverse its motion. As excursion increases and BL drops, this part of the opposing force also drops, and depending on the suspension properties, the driver is likely to exhibit *more* excursion than expected for the power input. This does not mean that the output increases because output depends on acceleration, not excursion. Physically speaking, acceleration describes both rate of increase in velocity (positive) and rate of decrease in velocity (negative). Sound comes from oscillation (positive and negative) of the cone acceleration. So paradoxically, increased compression and distortion will be observed at the same time that excursion increases. At some point, the coil can literally fly out of the gap, and it's up to the suspension to restrain its motion, or else. It's likely at that point that the sound may abruptly change from mildly or moderately distorted to very heavily distorted. As I discussed in previous posts, if driven hard at resonance, it may exhibit very little distortion right up to the point that it suddenly bottoms (soft or hard). This occurs because at resonance, motor force has only minimal effect on the driver motion at its extremes, so any BL distortion may be inaudible even if the coil is flying out of the gap. This is especially true for high Qtc alignments.
  15. JTR Speakers Captivator 212Pro Discussion

    I kind of doubt it's cone flexure. That's what cone break-up is all about, which doesn't set in until much higher frequencies. As a guess, you could be encountering strong inductance or actual flux-modulation effects. Strong IM distortion may sound a bit like mechanical noise if it's bad enough. Being IM distortion, I guess it might not show up as THD, depending on the analysis being used.
  16. Should I go ported?

    Overkill? What's that? Seriously, I thought my 4 x 21" subs would be overkill for 16 Hz+, even as I worried about getting enough output below there. It turns out my floor is pretty much inert under 10 Hz, and not very exciting in the 13-18 Hz range either. So I'm planning to add Crowsons some day. All the same, the content above 16 Hz is very clean and sounds better louder, I think. All the while, the placement locations gives poor response for 80-120 Hz, so I'm quite glad I have all the extra headroom. As far as the Crowson's and my couch are concerned, I'm thinking of cutting some plywood sheets to set the sofa on and putting the Crowsons under those. I think that'll raise the sofa by 2" or so, which isn't bad. I don't know whether I want to do 2 or 4 Crowsons. Only 2 may be plenty for me. I don't need things to get too crazy. I just want to fill in where the floor isn't active, including the single digits.
  17. Should I go ported?

    Another thing you can do is run a sine sweep with the mic right next to or slightly inside the port. The resonance is very easy to see that way, as is the tune usually. I believe the 0 degree phase should line up with the impedance minimum. If it doesn't it may be a sign of slight measurement error. It's probably not a big deal for you though. 16.5 vs 17.5 Hz isn't really a big difference. I bet that thing is going to kick some butt. Are you going to run it with another SpeakerPower amp?
  18. Should I go ported?

    The tune can be found precisely by looking for the nearby zero phase crossing. I'm guessing the stuff at 110 Hz or whatever is the port resonance if it comes close to the modeled. If the port opens nears the rear of the cabinet, its length may be effectively extended by it, which would lower both the tune and resonance.
  19. Ricci's Skhorn Subwoofer & Files

    An 18" behind your couch? That's a joke compared to what two Skhorn's will do. IIRC, @Beastaudio is running *one* Skhorn these with together with 8 x 18s to handle the ULF. If you are using these for home theater, you could probably just plug a port or two for more extension. If you make them easy enough to remove, you could even switch back and forth depending on content.
  20. Nice! I gather that is without the +7 dB compensation? (Actually, it may only +5 dB, since Katz regards cinema calibration to be 83 dB @ -20 dBFS pink noise).
  21. Psychoacoustics and hearing

    Can you post a link to the article? I could not find it on Google, and the above link does not work for me either.
  22. Bulding the Room2 listening room

    I like the compositions in the album quite a lot. The bass in most of the songs is of very inventive design, unfiltered, and tremendous. With that said, I think it would be better with a lot more headroom (like +10 dB or more, ideally), not unlike most music releases. It's necessary in order for the tracks to share well with the intense bass that's present. Pumping was apparent throughout and at times extreme, and while at times it may have been intentional, I didn't feel that it always worked musically. Apart from that, I noticed a grainy harshness to the highs that I've noticed with other stuff from the artist that you've recommended. The nature of the sound is rough and edgy and lacks the silkiness and air that a good master has. I'm only guessing here, but it sounds like there could be peakiness somewhere in the 5-12k range and poor extension beyond that range. I've noticed some mixes like to put peaks in places in the treble to "bring out detail", and I guess that tactic might work on speakers that are otherwise recessed. It's also possible that the edginess arises due to clipping or some limiting strategy. I'd have to download the tracks to inspect for clipping. Other than that, it's possible that the problem exists or is worsened by digital lossy compression. Usually Bandcamp streams are reasonably HQ, and I rarely notice blatant artifacts. But who knows. I'm just trying to cover all bases. Anyway, is it just me that hears this harshness? Honestly, I don't find playback beyond that which I use for standard "loudness war" stuff to be comfortable to the ears. That's still enough for some powerful bass though.
  23. That would be the Katz recording, here (registration required): https://www.digido.com/portfolio-item/we-have-lift-off-now-in-surround/ Make sure you get the 4.0 version without the music. The recording at the public viewing area peaks in the 120s. Of course, that's like 3 miles away from the actual launch site.
  24. Bulding the Room2 listening room

    I used to do that sort of thing a long time ago when I used a Linux PC for DVD playback. The player had surround virtualization built-in. The effect worked a lot better with headphones, and there were times in which I actually played the movie with the stereo on (so I could feel some bass) *and* the headphones on my head. Yeah, that was a while ago.