gnarly

Hi uams, The two horn loaded subs i have are DIY Labhorns (the Tom Danley design on Prosoundweb), and JTR Orbitshifters. Both are FLH, with about a 9-10ft horn path. The Orbitshifters use a single long stroke 18" firing into the throat chamber and will definitely walk around when cranked up enough. The Labhorns use dual opposed 12"s firing into the throat chamber and do not vibrate much, if any, no matter how loud. I hope that's the principle you meant.....? The combo definitely improves with active tuning. You've probably seen B&C's recommended filters for use with their passive xover in the 215-DCX suggested build. When i say "traditional PA", i mean in comparison to high quality boxes like Meyer UPA-1P's and such (which i own and compare to). It's a good sound, just lacking a little definition and openness vs what i've been hearing with the Synergy builds. I've been working on Synergy iterations for 2+ years, slowly finding improvements that continue to distance their clarity from anything else I've heard. If it weren't for some builds going backwards, i might think it's just a case of continued confirmation bias Lol. I have every intention of returning to the B&C combo, and giving it the time it deserves. Maybe even trying the 215 design. I'd like to compare it to the DIY PM90/60's that i think are outstanding high-output boxes.. But happily right now, i've hit on a Synergy build i like so much, I'm in the process if building a couple more of it, for a LCR setup.

Hi Dave, i took the dcx464 / me464 combo active like shown earlier in this thread https://community.klipsch.com/index.php?/topic/200572-bc-me464-horn-opinions/&tab=comments#comment-2628703 I don't have the passive xover, so can't compare to "non-active". From raw measurements of the two CD sections, I'd be surprised if sounded very decent without some active help. Even with the processing i put in place, it hasn't been a combo i've felt was worth pursuing further, as i'm really enjoying DIY conical MEHs. I agree with your assessment of it sounding more like traditional PA. So happy to exchange thoughts via PM if you still desire, but i can't say i've had what i'd call really good results.

Hi uams, thx. The subs are dual 18 push-push slot loaded (clamshell style PPSL). I basically took a very successful single 18 bass reflex build, and built the dual 18 PPSL scaling up the single's volume and port specs. The cabinet vibration reduction vs single 18's has been awesome. I can get rowdy and walking subs have been a problem. Are you currently using the MicroWreckers i see in your equipment list or the reflex boxes i see in your pict? I bet both work great. I agree that diffraction slots have gotten an unfair negative reputation.... No doubt there's some bad implementations, and bad CD/horn combos, using diffraction slots; but here's some dang good ones too !

Yikes ! That's quite the imagination ! No problem at all, your request will be honored.

I'm sorry to hear i was part of the reason you quit posting at diyaudio. I don't understand why you would have a problem over approaches and methods that are different from yours. Nobody's approach is the end all, be all. Tis healthy to discuss various approaches and methods as long as respect and courtesy is provided to all parties presenting theirs. I hope you can realize how condescending that sounds..."(if there are, in fact, any)".... You're correct about my "questions that aren't questions". I didn't ask any questions in this thread, other than what you mean by zeroth order? (which I'd still like to understand) I didn't disagree or present any alternatives to your first order tuning methods. I presented observations of measurements that don't reconcile with timings you discussed, and my viewpoint (theory) as to why TOF's measure as they do. Thank you for appreciating my civility, and for when you provide civility in return. I need to ask : Why is it important we see to eye-to-eye ? I don't care if you advocate a particular tuning strategy....why do you care if i advocate a particular different strategy? What is there to agree or disagree about? Both strategies work just fine. (If i ever hear proof why my strategy doesn't work, i'll happily and very thankfully accept the instruction, knowing better audible results are forthcoming !! ) I'm sorry it appears you don't like it when folks do not adopt your viewpoints, approaches, and methods. I don't know if i'll keep posting here or not, as you seem to be disinviting, directing me back to diyAudio. Seems like a pity to me...I know you have a lot to offer, that there is a lot i could learn from you. And believe it or not, i have a lot to offer you too. Anyway, later.. for now. Sincerely, Mark Peace...

No, i don't think that's it. Phase lag induced for every order of the crossover filters applies to IIR crossovers. I'm using complementary linear phase xovers which have the same phase lag (none), no matter what order used. Spectrograms, step, group delay are all just derivatives of mag and phase response....they all key off the same TOF measurements that place the woofer further away that the CD. (REW: TOF = Delay relative to loopback) 26 minutes ago, Chris A said: (This seems to have been edited away or something...???) My reply got hung up Saving...so this is a put back together attempt) I think you're having trouble using these extremely high order filters having sharp cutoffs in MEHs, that absolutely don't need those kind of filters--because the reason for using them is simply not there as it is in multiple-aperture driver/horn loudspeakers. The multiple aperture MEH's help mitigate the need for steeper xovers for sure. But MEH's can improve further still with their use. I have tuned them both ways and the steep acoustic rolloff of a large woofer crossing to a CD has been easier to get right using steep, for example. I use outdoor polars to determine whether to use steep or shallow. Working on such a project right now...

Hi Chris, i should first ask what do you mean by a "zeroth order crossover" ? Not familiar with that idea... Here's my two cents... So far, on every MEH i've built (maybe a dozen counting fully developed prototypes), the woofers have always measured further from the mic, than from the CD. This has occurred despite the fact the woofers are closer to the mic than the CD,....... as in the drawing you posted. And it occurs whether taking time-of-flight measurements with no filters in place, or with linear-phase low passes of any order (which don't have any group delay). I've come to the conclusion, perhaps wrongly, that the natural acoustic low pass of the woofer is the source of the delay. Well that, and also i believe that our FFT measurement programs use the Hilbert-Transform to make TOF determinations under the assumption minimum phase represents correct timing. At any rate, i know all my builds have had to add delay to the CD. For the first time recently, i tried adding small 4" mids between the Cd and woofer (even though not needed). The mids also need delay relative to the woofer. With the woofer at 0.00ms, mids delay is 0.32ms, and CD is 0.83ms & 0.90ms (hf & vhf dcx464 coax sections) You can see from the spectro, those delays put timing right on target. (other than i got a small bobble at 250Hz, xover between woofer and mid)

Thx, I saw the AES3 cards and was hoping you were using it for your digital signal processing. But like you say, anything can be done with the Cores, and maybe you had something else going on with it... I'm such a huge Q-SYS fanboy too... Here's the schematic I'm currently listening to. It's for a 4-way MEH on top a dual 18" sub. 500i fed via Q-Lan from a 110f. 16384 taps on all FIR filters! (just to keep from messing with different delays Lol) Now i just need the super amps you guys are talking about 😁

Langston, may i ask a quick question unrelated to thread topic.........what does the Q-SYS Core do? Thx, Mark

Hi Edgar, in my previous reply, all I did was push against pre-ringing being significant. And your question did ask what is important (to me)..... and as you said, i do go to great lengths to eliminate phase shift/group delay... So to try to explain further... Logically and intuitively, flat mag and phase make sense to me for reproduction. I think everybody is on board with flat mag (subject to house curves and preferences of course). But we all know how much phase audibility is debated. Personally, i want to do what i think and intuit, is technically right, whether audible or not. Seems like a can't loose, and will probably win, as well as learn something strategy. And so far, the flat mag and phase effort, along with extending it to smooth polars, has been paying big audio dividends. Subjectively I could wax on alot...but i don't really see the point...suffice it to say , tonality, clarity, transients, low end dynamics are making me a very happy camper. Of course, i can't really separate the processing gains from the speakers' acoustic design gains, but since almost all the considerably varied speaker designs i've attempted have shown similar sonic gains, i have to credit much of the sonics to a common processing technique. Anyway, if i may go back to the cost of potential pre-ringing vs the gains linear phase might bring... here's an example of the tradeoff (as i see it), that illustrates the order of magnitudes difference between the tradeoffs plus and minus. This is a measured not simulated, linear-phase LR 96 dB/oct electrical xover, where both the low-pass and high-pass sides have been summed together. For me, there isn't much pre-ring, due to complementary cancellation. What small pre-ringing there is, seems like a tiny price to pay for the outstanding Step Response. As i'm sure you well know, only first order can achieve a step response like that. And this is 16th order !! Gives so many degrees of freedom combining drivers acoustically. I feel like a kid in a DSP candy store...who walks out into a rocking live concert Lol

Gotcha. Thanks 🙂

What demonstrably audible pre-ring are you referring to? I hope not that FabFilter youtube....that thing's ridiculous imo. I mean, what is the high Q filter in it, min-phase? or lin-phase? And what is the point of it even being in place? If it's min-phase it should be fixed with min-phase....applying lin phase to it is a joke. If it's lin-phase, it's even more of a joke... Pure contrived marketing imo...... How do I choose which audible effect to worry about, and which to ignore ?.... Good question...🙂 I guess it's as simple as the audible effects i can hear in my real world builds, and my real world testing. No hear, no worry. Can't really say i feel i need to read about, what to worry about...

Yeppers, the 0.01% hit that pre-ringing might maybe bring, is indeed 10x larger than the 0.001% hit to audio bliss that audiofools typically worry about, huh? ......haha

The sound degradation in the FabFilter youtube is very easily heard, so i'm not sure what you mean by saying it needs to be lossless to be heard. I think the worst MP3 would make such a contrived comparison obvious. That's my point really, it's contrived crappola for marketing purposes.... It's not representative of any real world pre-ringing issues, imho.

Yes, I've seen and heard many of those studio type presentations regarding pre-ringing. They all have a simple common trait, imo..... they all use a very high Q electrical filter supposedly linearized (the idea of which is nonsense) ..... carefully constructed to prove their point. Fabfilter is one of the worst....pure marketing on a fish hook. Lol Bogus amogus big time! 🙂 imho.

Hi again Steffen, i think you've asked the real question....and i kinda ducked it my previous post. "What does pre-ringing sound like?" I truly wish somebody would nail it down ! 😉 The only time i've ever thought i heard pre-ringing, was when i was doing some super long FIR files (64k taps) using j-river convolution, with the same long FIR file size on every output channel. Kinda sounded like a pre-echo maybe, when there was silence before transients. But that experimentation was so short lived, without any real confirmation, it's almost stupid of me to mention it. The thing with FIR in my mind, is that it is a VERY powerful tool.....which requires VERY judicious use. FIR questions me, "I don't give a shite what you want to fix. Do you really know what to fix and why?" Pre-ringing only happens when we can't answer FIR's question, i think..... Being a lover of "hold-to-target down to -30dB" may be as much BS as pre-ringing 😁

Great, glad that made sense. I don't think differing from ideal curves matters too much to potential pre-ringing. As long as the difference is relatively minor, and not near the central xover frequency. With complementary lin-phase xovers , if mag and phase look good thru xover region, pre-ringing is simply not worth thinking about. -30dB is my standard because I've found it easy enough to do...probably overkill in terms of ring. What I think matters more to potential pre-ringing, is how complementary is the acoustic coupling both on and off-axis. The idea that complimentary linear-phase xovers cancel pre-ringing, where the driver on the low pass side negates the driver on the high pass side, depends on acoustic symmetry. So good old geometric lobing rears it's head again. Which again gets back to why I like steep xovers....as they minimize the region of potential lobing. My take on pre-ringing is it's mostly the same type of audio overconcern/overthinking, as with so many other audio things buried deep under the noise floor. Don't mean to offend anyone, but i believe in ranking marginal impacts, maximizing SNR, which sometimes means living with a little more noise i can't hear to gain a whole lot more signal i can hear. That said, i do think many of the experiments where folks have tried to linearize phase globally on top of existing passive or active designs, have had the capability of bringing pre-ringing up to the surface.. The success of a global phase correction overlay depends on how well the speaker had already achieved acoustically complementary xovers, and also level in-band mag and phase response for each of its drivers. And again, for both on and off-axis. Otherwise the global correction will be valid only to the mic location.....and complementary ringing cancellation could be sucking wind everywhere else.

Hi Steffen, How far we want flatten to is set by the driver's acoustic target order we want to achieve, and how deep we want to match the target. For example, say the acoustic target is LR 24 dB/oct. And say the goal is hold-to-target down to -30dB. Here's a 500Hz LR4 lowpass. As you can see, flattening to -30dB would need to go out to about 1200Hz. Here's the acoustic target of 500Hz with a much steeper LR96 dB/oct, and goal is the same hold-to-target to -30dB. Flattening to -30dB needs to reach only about 620Hz. Ok...how far can we actually flatten. The raw measurements speak. Here's the raw woofer/mid section of syn7, a pair of 10 inchers that reach up to the CD. (with round ports in horn centers) I think you can see that flattening response up to about 700Hz is pretty easy. That out-of-band minimum phase frequency flattening to 700Hz, also takes care of the section's out-of-band rising phase (to 700Hz) The in-band minimum phase frequency flattening (which almost everyone does), takes care of in-band phase flattening. So when a steep linear phase xover is applied on top of the all the minimum phase flattening, the result is flat phase across the driver's bandwidth down to -30dB. When the same flattening process is applied to the CD's low end, and uses a complementary linear phase high pass (500Hz LR 96 dB/oct), the result is very flat frequency and phase response across thru xover region (and across the board). Hopefully, our woofer/mid ports are located where the 1/4 wave notch frequency is well above the -30dB down frequency, which makes the notch mostly immaterial after the steep lowpass. When I try a shallower acoustic target for the woofer/mids like the LR4 first pictured above, it ends up taking alot of parametric work,along with shelving filters to flatten out to -30dB (or about 1.2kHz). I always end up saying screw this, this is nuts......cause steep works 😁 I've used this technique on everything from MEH's, to co-axials, to PA traps, to line arrays, and all of them with subs (crossed to with same technique). Iow, anywhere there's a xover ..Lol Always a cost in audio though, huh ......Price is FIR latency, and need/expense/complexity of multi-channel processing / amplification.

If you mean being unable to specify bi-quads and such for IIR work , then I'll prefer to stay lost. 😂 If you mean unable to specify FIR coefficients for a FIR processor, i of course obviously agree. I take 24-32 bit floating point, 48-96kHz, with sufficient processing, as a given. So for me, I/O channel counts, I/O input and output types and their type connectors, are the first screening factors. Then comes whether I want FIR, and how much. Then comes the breadth and depth and counts of IIR filter types, and orders. Precision of delays, gains, limiters if needed...etc. etc Iow, What can it implement quickly and precisely. Ease of use is an ever present factor.... whether setup/controlling from software, or from the unit's front panel.

Sure, there is definitely alot of hand waving. !!! Here's some more !!! 😄 For me, the fact that the transform can be done mathematically in a number of different ways is almost entirely immaterial to choosing a processor, and figuring out exactly how I'm going to use that processor to implement (what the math says can be done). The math alone isn't going to do any processing is it ?

Hey, here's one that might meet your requirements....i forgot about it in my zeal for Open Architecture stuff. Linea-Research ASC-48. (Or also available rebadged as a Danley SC-48) https://linea-research.co.uk/asc48/ 4 in-8 out. analog/AES3. 96kHz. High spec / super clean device. (I have the Danley version.) One of its really cool features is what it calls LIR xovers, which are essentialy 24dB/oct linear-phase Linkwitz-Riley's. What makes them cool, is that the LIR xover frequencies can be dialed in on the fly, just like any usual IIR xover..... AND the processor automatically adjusts internal delays to keep phase alignments spot on I haven't seen that anywhere else....

Hi all, apologies for quoting my own post, but i wanted to build on it and try to get some feedback. I want to compare the technique of out-of-band fattening combined with complementary 'named crossovers', to building 'un-named' xovers using combinations of parametric EQ's, shelving EQs, perhaps along with high or low pass filters, etc. Because i believe the two techniques give identical "total filter" results, for a given acoustic order/slope target. By "total filter" results, I mean a transfer function of the whole electrical filter package of each approach. The out-of-band flattening when combined with a named xover, becomes the actual xover...and as as such, the combination formed also can't be named. That combo should look just like the "un-named filter approach", again assuming they both produce the same desired acoustic target. Imo, the really cool things about this equivalency are: that it lets us use the named xovers in our processors, and i find it is much easier to EQ response up to a horizontal flat line, than it is to try to nudge one curve on top another (which is what we have to do when using combinations of parametric EQ's, shelving EQs, etc.) The vertical distances between curves can be quite deceiving. But flat against flat is easy to judge. Anyway, this technique works so well for me, i keep trying to share it.... (credit for it goes to POS of rePhase)

Yep, been my concern too. I've come to believe complementary linear-phase xovers cancel each sides preringing, and are completely safe to use. So even for the low frequencies, like for a sub-to-main xover at 100Hz, i feel comfortable with a linear-phase xover. Where i don't have comfort with linearizing-phase, are on the spectrum ends, 20Hz and 20KHz. Because there is no complementary offset. So I've been using IIR for sub high-pass and 20kHz+ low-pass (which for me has helped clean up any tiny digital impulse oscillations) I also don't feel comfortable linearizing a driver's phase from its natural frequency response rolloff. There, I like to flatten the driver's out of band response with minimum phase EQs, hopefully to the point that the added electrical xover order becomes the driver's measured acoustic order. That's a big part of the reason I like steep linear-phase xovers....it makes the out-of-band flattening much easier (and protects against lower-end over excursion from the min-phase flattening). Steep also has helped me minimize the width of lobing region, making it easier to measure smoother polars thru xover region. And i think that thereby also minimizes the freq range where complementary linear-phase xovers might not fully cancel pre-ringing acoustically. Any comments / concerns welcomed....🙂 Gotcha. Like said, I'm clueless about video.

Ok, very cool....maybe this is the right thread to continue on with phase some. I very much share your views and subjective comments about phase audibility. It's a very narrow tightrope to tune to, but when right, for me it's like washing a windshield (to use a visual analogy). Phase audibility is what has driven me to find the best processor i can, to be able to both implement linear-phase and then easily A-B it against traditional IIR. Q-sys was the first affordable platform (used) i found that allows instant A-B, FIR vs IIR. In case folks are interested in the Q-sys Cores' FIR capability, I've found a Core110f can run 8 channels of 4k taps per channel (@48kHz), on stereo 4-ways. Different tap allocations per band are of course possible, but I haven't been able to put more than 6k on the sub FIRs. Due to that, i recently acquired an older Core500i, that allows 16k taps per channel on at least 16 channels. 16k taps is the current Q-sys maximum, raised from 8k a year of so ago... Of course like you say, such latencies are a no-go for home theater without video delays. Do such long video delays exist....? I have no clue as I'm audio only, (and use computer logitech speakers for video. lol) Anyway, I feel I've had super success with phase linearization, both audibly and with measurements. I've used it on a number of different type 4-way builds, always on a driver by driver tuning approach, where each driver gets it's own FIR file and amplification channel. Here's a spectrograph example from a flat-phase MEH build, that uses a CD straight to 10"s. (I used the spectrograph settings you showed in the linked phase flattening thread.) It's steeply high passed at 100Hz. Here is a dual 18"s push-push slot loaded sub's transfer function, that I'm using with the MEH. It was taken outdoors at 4m. (I don't bother with indoor sub measurements.) Green is raw, blue processed. It was with 6k taps. Looking forward to dragging it back onto the driveway and retuning it with Core500i. I think the processed phase trace is already pretty good, but 16k taps will raise frequency resolution to a little better than 6Hz (from the 6k's resolution of 16Hz), to maybe gain some bass definition....who knows 🙂

Awesome help to folks, like you've been doing. I wish i wasn't such a slow writer and could try to help more too.... (i ducked every writing class possible in undergrad and grad,.....to my current chagrin ...) Anyway.... back to to audio..... I can not stop with A-B comparisons, on what i think you call phase growth. They keep redefining what I think i know/hear. Are you simply talking phase lag, and re-coining it into 'phase growth' , or are there subtleties I'm missing with your term phase growth? I keep playing with phase rather intensely. I'll post some of my typical measurement results, and the tuning techniques that have given best results, on the thread you linked...for critique. Seems like they would fit better there. I'll try to stay away from subjective results.....cause we all know well they work 😅