Jump to content

SME

Members
  • Posts

    1,699
  • Joined

  • Days Won

    109

SME last won the day on December 29 2021

SME had the most liked content!

1 Follower

Profile Information

  • Gender
    Not Telling

Recent Profile Visitors

2,186 profile views

SME's Achievements

Super Bass Overlord

Super Bass Overlord (8/11)

350

Reputation

  1. Well it's not always true that 2-way bass will sound worse than 1-way bass or whatever. The end result depends on the whole system. Nevertheless, crossovers in the bass are rarely ideal and introduce flaws which are difficult to eliminate completely with additional EQ. Multiple crossovers make this much more difficult. Attempting to design a cabinet to a particular target curve is likely to compromise overall efficiency. The BP6 designs like the SKRAM and SKHORN have a lot more output at 100 Hz than 30 Hz, even though most applications need more output at 30 Hz than at 100 Hz. That's not a flaw though. It's fundamentally harder to get more bass at 30 Hz, and the higher efficiency at 100 Hz makes more amp power available for 30 Hz while keeping the voice coils a lot cooler. As far as target curves, I think they are best treated as a sanity check rather than a strict guideline. I've seen a lot of different "preferred curves" (including multiple allegedly recommended by Harman over the years). They all tend to suggest a gentle downward slope and an SPL difference between bottom and top of 6-10 dB. They vary with respect to which frequency areas are steeper, but I suspect these differences are not that meaningful, not because such differences are not audible (they very much are) but because different systems and rooms will probably sound their best with different target curves. And don't forget that measurement location has a big impact on the broad shape of the response. The other thing is that actual content can vary a lot in terms of broad spectral balance, even between different tracks in a given mix. I realize this doesn't answer the question of what ratio of subs vs. tops to provision for. The answer is probably a matter of preference in part, and may be affected by budget as well. For my home system using experimental DSP, I rely on an estimate of in-room *acoustic power* response to optimize my bass. I use a -3 dB/octave target for ~1 kHz and below because that yields a response for my speakers that's closest to measuring flat under anechoic conditions. This also gets the SPL response at my seats pretty close to the "preferred curves", though I measure particularly hot from 20-70 Hz, which probably has to do with most of my seats being near a partial wall (a high pressure zone). Subjectively, this seems to be a good reference for a "performance" system, but I know a few people would want even more bass, especially for a bass-centric genre. For a large room / outdoor PA system, there is little boundary support, and so it'll take a lot of sub to match tops while maintaining that -3 dB/octave power curve. I would want at least +6 dB more output per channel at 30 Hz than 120 Hz, and indeed +6 dB in terms of capability may not be enough for the content that has a steep power curve of its own. (This is very much the case with movie tracks!) And if a -3 dB/octave power curve isn't enough? For -4 dB/octave from 1 kHz down one needs an additional 5 dB at 30 Hz, which requires almost twice as many subs. On the flip side, if budget is tight and loud bass is not essential, one can use about half as many subs to target -2 dB/octave, and so on. To be clear, most of my dB figures above are in terms of *acoustic power* output, which is not something one typically measures directly, but on the other hand Hornresp simulations (for sub/speaker design) actually give results in terms of acoustic power, not SPL. SPL response only loosely tracks power response because of directivity effects and environmental acoustic factors.
  2. Drivers in series have higher impedance, which means that yes, they share voltage drop. And an amp in bridge mode typically grants 2X the voltage compared to each single channel alone, so it all balances out. Damping factor is kind of an anachronistic term from when amps (made of tubes) had non-negligible output impedance, so damping factor was high enough to significantly alter the bass response in ways that varied depending on pairing of driver and amp. I don't know if I'd call this "distortion" though, in that in audio distortion usually implies non-linear distortion whereas increased "damping factor" looks a lot like a shift in T/S parameters. It's likely something that can be completely corrected with EQ (except for the lost power, of course). The other thing is that more than 2 drivers in series may behave sub-optimally if they are different or if their specs are not very similar. However, I don't think I've heard of this being a problem in practice.
  3. I'm not sure what you mean by the "whole purpose" here. As noted above, it's not hard to cover 30-100ish Hz solidly with a BP6 like SKRAM, and many mid-woofer sections get down to 100 Hz no problem. From there, it's a matter of tuning things to get the best bass sound, and it's a lot easier to tune with only one crossover in the bass area. When crossovers are close together, the problems are significantly worse. With that said, I think your idea of using bass horns to cover 2+ octaves from 70-250 Hz is likely to work out a lot better than trying to use dedicated "kick bins" to cover only ~1 octave worth. Really, the "sub sound" lives in the frequencies below 250 Hz or so. That's because 2nd and 3rd harmonics contribute substantially to bass sounds. A 65 Hz fundamental has 2nd and 3rd harmonics at 130 Hz and 195 Hz. All of these are crucial to the most impressive, most tactile bass sound. (Higher harmonics matter too but are of diminishing importance.)To be clear, this doesn't mean you want "flat response" across the bass. A power response that rises toward the bass sounds most natural. Power response concerns total sound output in all directions rather than SPL measured at some point. Almost all speakers which measure flat on-axis in an anechoic chamber exhibit a downward sloped power response, starting at a point below which the radiation is omni-directional. Below this point, in-room SPL response typically rises if the trend in power response is being maintained (hence the "house curve"). This tends to happen naturally in small rooms, but in a live/PA setting where bass-boosting boundaries are fewer and farther away, this can require a lot of boost for the lowest frequencies. Hence, I'd say that if you're looking for more mid-bass "kick", you probably don't need more mid-bass SPL unless you're already working with a whole stack of SKRAMs. What you probably need is smoother response across the mid/upper bass range, which is best served with better tuning rather than more brute. It doesn't take much SPL at all to slam and throb if things are really tight. Though of course it's all relative, so if you need your mids and highs at ear-blistering levels, you need some serious bass to keep up, and (except in small rooms indoors) the last octave is almost always the most demanding of SPL.
  4. I very much lean toward fewer crossovers being better. Crossovers are rarely precise enough to not degrade sound quality. I would also add that SPL isn't everything. Better sound quality can provide more apparent loudness (and more listener enjoyment in general) at lower SPL. So while you may be sacrificing low-end on paper, if fewer crossover resonances means you're "hearing more bass", you might still come out ahead.
  5. The way I read this, your described workaround is functionally the same as just engaging "bridge mode" in the first place. This makes perfect sense, especially if the loads on each channel share a common ground. Which is to say that one should just enable bridge mode anyway because it's easier. But this does not necessarily allow for *asymmetric* loads across multiple channels, which is basically the point of having multiple channels in the first place. Asymmetric loading between different subs can be useful in some situations, but not all amps can reliably do this if subs are involved. And I would say the "worst case scenario" isn't having the amp shut down. The "worst case scenario" is designing with an amp that is not adequate. OP is wise to carefully choose the impedance to match the intended driver and amp configuration. And as noted by others above, damping factor is not a problem with proper matched wiring.
  6. Don't worry about damping factor, but do keep in mind that many amps are not designed to power subs unless they are used in their bridged mode, in which the amps for two channels work together to power the load. The bridged mode typically doubles the impedance requirement, so an amp rated into 4 ohm loads will typically require an 8 ohm load when bridged. You could use this to power a single 8 ohm sub (with a lot of power and one amp per sub), or you could connect pairs of 4 ohm subs in series and run each pair off a bridged amp.
  7. My speaker optimization requires way more than 1024 taps. More like 128k taps, as I noted above. ARM support would mean running the DSP on the CPU itself rather than a dedicated DSP chip. Though I'd like to look into supporting dedicated DSP chips as this could help lower latency further, it's likely to involve some heavy R&D investment and is a low priority. You make a good point about FOH tolerating more latency than monitors, and I appreciate you sharing these figures, which seem very reasonable to me. Of course we're talking about how much total latency is acceptable, not how much is acceptable for a single device in the chain. I'm thinking that for devices intended for live performance monitoring, I want to aim for under 5 ms if possible. If I can do this, it will probably be for a much smaller number of channels than would be possible with higher latencies. That's probably an OK compromise. I should also be clear that I do not believe in DSP optimizing a speaker and sub(s) separately. They must be optimized from measurements with the crossover in place or from precise simulations of the crossover. Otherwise, the result is completely spoiled by the crossover. If the mid-bass is screwed up, the whole bass response will be subjectively *much worse* for it.
  8. Thanks much! Business-wise, I'm feeling very risk-averse since the pandemic made a mess of everything. The real killer app that I am developing is for sound quality optimization, and I believe what I can do is a game-changer for audio reproduction, certainly in home listening rooms. From a business standpoint, that's actually kind of a problem. I could probably get paid a lot of money to allow the tech to be buried forever. I don't want that at all, but in the game of markets, I might have that choice made for me if I'm not extremely careful. The other thing is that I could probably occupy myself for a decade or more with more R&D including more formal scientific work---theory development, listener experiments, etc. If I could self-fund it and find the right advisor for myself (someone not in audio most likely---I need someone with strong statistics / information theory background), I might do a Ph.D. I should at least mention here that there is at least one Linux + open source DSP program that is quite capable. See BruteFIR. I opted to make my own DSP for various reasons, but with a bit of tinkering, BruteFIR may be usable to some. It's complicated. My personal setup has a pretty high delay because I don't really need low latency. I'm using 3 x 1024 sample buffers at 48 kHz, which I think works out to 107 ms round-trip plus whatever the interface adds (single digits ms, I think). I'm using larger buffers because I'm running 12 x 128k FIRs, 12 x 64k FIRs, 8 smaller FIRs, and a few hundred PEQs. It's also running on a single core mid-level 2009-era CPU. I have many big opportunities for improvement. My new code-base supports multi-threading and multi-rate processing, which will improve compute efficiency a lot. I'll likely have to get away from USB to get the best latencies. I can't use thunderbolt on my Motu16A in Linux and don't know about other hardware, but the 16A does support AVB. AVB is arguably the best technical standard for Ethernet audio transport, though it's not the most widely supported because it requires specialized hardware capabilities and quite a bit of supporting software as well. I've only spent a few hours on it, but I haven't gotten AVB to work on my Motu Ultralite AVB yet. It passes audio but is very glitchy. I think I can get it to work with more time, but CPU overhead may end up much higher than I prefer. OTOH, the AES67 and compatible protocols (which the Motu stuff do not natively support) are simpler and a lot more likely to work with typical Ethernet interfaces. Looking just now, it appears that Linux support for AES67 has matured a lot! When I am further along on the software and am ready to spend more time looking at hardware, looking at AES67 capability will be a high priority for me so that I can support a wider variety of hardware. Just a side note about delay in general is that the filters themselves can introduce non-negotiable delay. Linear phase filters and crossovers require additional delay depending on how low in frequency, how high Q, and how much accuracy is desired. Even delaying a speaker in order to better match it to a sub with a long horn expansion or in a different part of the room is contributing very real delay. Fortunately, I don't think linear phase filtering or crossovers are critical for outstanding sound. It's something I need to revisit, but as far as I can tell, phase is not terribly important as long as the shift is not too dramatic. ======== If you all don't mind me asking you all, for live sound purposes, how much latency do you all think is tolerable in a DSP appliance? Lower is better, obviously, but at what point is it really a problem?
  9. I'm not aware of any outboard appliances that offer much in the way of modularity, which is a real bummer. I've been using my own modular PC-based DSP software for 5+ years now. I wrote it because I wanted something Linux-based that I could experiment with. I had been using MiniDSP devices and hated having to use their pointy-clicky Windows-only interface to get my filters installed, especially being that I was already computer-generating the filters. I'm now working on the successor software, which could become part of standalone appliances some day. I'm targeting multiple hardware platforms---not just PC but also potentially ARM-based stuff like Raspberry Pi. I'd like to be able to use it to build fully integrated (DSP+amp) speakers, and the functionality you're looking for is something I desire for that purpose among others. A key difference with my processor vs. a VST-based processor is that my architecture is tightly integrated. As latency is pushed lower, extra layers of indirection contribute non-trivial cost if the process graph is complicated. The downside is that I don't support external modules, which is what VST is all about. In theory, such support is possible, but it's not my priority now. I know my work doesn't help you right now, but I thought I'd mention it because I know I'm not the only one who's wanted/needed these kinds of capabilities. Integrated and outboard devices are even less useful when you need to process many channels and want some kind of matrixing capability. For example, how do you optimally bass manage 5+ speaker channels (at different room locations) + "LFE" to 4 subs (also different locations) in a home theater? Answer: with even current state-of-the-art solutions, you don't, really. Ideally you want separate sub delay (and maybe EQ too) for each signal between each input and output channel pair. This is especially important in larger rooms or where you want to (properly) do something like a double-bass array. Ideally, each speaker and the sub(s) its crossed with behave as a coherent source throughout the crossover. In practice this is very hard to achieve if you can't at least apply separate delay to each combination of input channel and output channel.
  10. On the wire ratings, sorry my mistake! I reviewed the posts, and my display is working fine. For my home subs, I used 4-conductor 10 gauge cable (5.25 sq mm), one cable to each cabinet for a total cross-section of 21 sq mm for the 12 kW amp, and the loads are closer to 4 ohm.
  11. The Sanway "clones" may perform better or worse than the similarly named Lab Gruppen brand amps. I'm not even sure how closely the Sanway amps resemble the original designs. The "clone" may just be in the branding. It actually wouldn't shock me if the Sanway amps actually perform better. Of course, I doubt anyone will be buying and measuring one of the original ones any time soon. That said, while 130 ms might be OK for rock kick-drum, dub-step and related genres typically have bass with a lot more compression. In my example track from above, the "average power is 1/4th of max" applies when I analyze the whole song, which has a few brief breaks from the sustained bass. If I exclude those brief rest intervals, the average is closer to 1/3rd the max for passages that are a minute or longer! This suggests that one should provision *a lot of spare headroom* into systems that play this kind of music and pay much more mind to long-term power capabilities in components. For example, the 21SW152s may have an edge over the DS115s for such systems, and it's definitely important to know how much output an amp can sustain for those minute long sustained bass passages. :) I wish there was an easy / straight-forward answer, but a lot really depends on details including the particulars of the content. One of the nice things about over-provisioning an audio system if one can afford to do so is worrying less about accidentally taking it too far and breaking something. Regarding wiring, I assumed the 4mm figure indicated by @domme referred to diameter and not cross-sectional area, which would have units of mm^2. A cylindrical wire with 4 mm^2 of cross-sectional area is about 2.25 mm diameter.
  12. Line resistance is basically a linear effect, and when it's small, its impact will probably not be more than that from normal T/S parameter variation in the driver one's using. One can even input the linear resistance directly as a parameter in Hornresp, to see what happens! This also means that the effect can be compensated for relatively easily using minimum phase EQ. Of course, the wasted power can't be recovered, which limits max output. An important question to ask in the above example is whether it's acceptable to be dissipating "1200W of power" (see below) into those lines. That depends on how much wire length the dissipation is spread over. The more general problem concerns how much *current* an electric wire can handle for long periods of time without overheating. This dictates electric building codes that require a minimum diameter wire for a given current rating. Now, no one's going to run their sub amps at full blast forever. Few if any amps will even do that, and the music program rarely requires that kind of power for the long-term. What your average power looks like, assuming you're not running hard into clipping, depends on the music. A nominal value of -9 dB ratio of long-term average vs. short-term average is a kind of industry standard. This -9 dB is equal to 1/8 power. For dub-step and other music with long, sustained bass, it's possible the average power could get higher, maybe 1/4 power? I don't know, but I just checked a few tracks I own that have loud sustained bass and, after applying a low-pass filter at 100 Hz and normalizing, the loudest one came in right at 1/4th power! So take 14400/4 = 3600W long-term average into 2 ohm => I = sqrt(3600/2.0) = 42.4 A A conductor ratings chart indicates that a 4 mm wire is rated for only 37 A. So if this system is going to be doing "pedal to the metal" for hours on end, a second 4 mm run (or a single larger diameter run) is best so that the wires won't overheat. This will also cut power waste in half, which in the above example means getting 0.5 dB back. Not much, but cables are cheap compared to drivers and amps. ---- If I may suggest you consider amps other than the FP14400, keeping in mind that amp power ratings are often based on some unspecified "burst" capability, which doesn't even last long enough to be meaningful for subs. The other thing to keep in mind that is that most Class D amps need to be run in a "full bridge" or "full bridge tied load" configuration in order to not become unstable when driving subs at high levels. Some amps like (e.g.) Speakerpower and some ICE products are full bridge on each output channel. These are essentially already bridged internally. Other amps like Powersoft and probably the FP14400 are half-bridge on each channel. These amps typically provide optional bridging functionality, which I strongly recommend when running subs off of them.
  13. If you measure the tops at one meter and in front of the mid-woofer, then you are probably at a pretty steep vertical angle, which will have diminished output compared to a measurement directly in front of the top. So when you boosted to compensate, you were likely running the tops too hot. Be sure to measure the top from directly in front if you want to use this data to match the different sections. Also, if that bump at 4-5 kHz shows up at other measured angles, you might want to try knocking it down with some PEQ (if you have it) because that resonance is likely to be a real ear-grinder, not to mention stealing a lot of attention away from the rest of the good stuff.
  14. Last I checked, the vast majority of DnB and Dub Step music go below 40 Hz. Plenty gets down to 30 Hz if not lower. A sub synth processor can also add more bottom. If done well, I believe adding extension is often impressive enough (in perceived intensity) to justify the SPL that was sacrificed. The biggest issue is that too much content is filtered too high.
  15. SME

    Bassboss Kraken

    I wonder where the vent resonance is. That may be what limits the max usable frequency. 21DS115! I have a couple of those I bought to make some compact vanilla vented subs for a kind of semi-pro demo system for my audio processing, but I got distracted by other projects and then the pandemic happened. Now I wonder if I should put them into a pair of SKRAMs instead, for what purpose? I don't know. The apocalypse? I'm already very satisfied with the capabilities in my living room, and I don't even know where I'd store them. This Kraken looks real snazzy but is obviously humongous and awkward shaped. I don't see the point of going beyond a dual-opposed configuration like a SKHORN, and as evidenced by its relative popularity, the advantages of the SKRAM seems preferred by most builders over the advantage of dual-opposed configuration in the SKHORN. I like that the SKRAM has more port area per driver, which also makes it more usable with a single port plugged for a bit more extension. So the Kraken gets points for going just a little lower than 30 Hz, but the SKHORN and SKRAM are configurable and can be tuned much lower when the application calls for it. A way better deal if you ask me.
×
×
  • Create New...