Front-of-House speakers missing midrange?? Opinions?? - Page 1
Forum section: electronics / projects

author: CaptDon
author_rank: Super Contributor
timestamp: February 05, 2022, 05:41:51 pm
content: There are two local bands both using the same type of JBL 'Club Speakers' or if you will 'D.J. Speakers'. My observation also applies to similar cabinets. I don't have a part number or exact name, but they are the typical stand mounted cabinets with a single 15" and a singe horn with a mouth of around 12"h and 8"v. What I notice is that the 15" is way to slow to articulate vocals above maybe 500hz and the horn is horribly inefficient below maybe 1.5Khz leaving a big hole in the articulate vocal range. Even with a lot of EQ the woofer is still inarticulate and muddy. I can hear the low boomy muddy portion of the vocals, and the brassy glassy ear piercing sibilance, but I can't understand a word they are singing. Has anyone else observed this when going out to hear live bands using similar 'set it and forget it' techniques?? I dislike these style of cabinets and I can't say I have warmed up to the 'Sticks sitting on a sub' offerings yet either. I am still maybe stuck in the years of the dinosaur with Tri-Amp using 15" Low / 10" Low-Mid / 6" Mid-High with passive crossover to a bullet tweeter for highs. Essentially a tri-amp system with multiple 4-way cabinets. My cabinets are 'all-in-ones' similar in size to maybe a 13 cubic foot refrigerator, maybe a tad bigger. What is your opinions / experiences??? TNX!!

author: bob91343
author_rank: Super Contributor
timestamp: February 05, 2022, 07:52:46 pm
content: Well you have your woofer and your tweeter. You need a midrange.

author: themadhippy
author_rank: Super Contributor
timestamp: February 05, 2022, 08:45:25 pm
content: QuoteAka Just Bloody LoudQuoteIf you ever get the chance to listen to an inspired audio optima 5 system prepare to be warmed up,the general reaction when used with a twin 15" sub is "no ways that little things making that sorta noise"QuoteFor old skool hernia giving i prefer all active,15" 10" and 2",sat on top of a twin 18" especially if it comes with an eaw or renkus heinz badge

author: AaronD
author_rank: Frequent Contributor
timestamp: February 06, 2022, 12:04:28 am
content: "Set-and-forget" NEVER works. Some people (apparently) think it does, but I'm convinced that that's only because they did it out of necessity, ignorance, or parroting someone else, and they just haven't had the "revelation by 2x4" yet to the contrary. It also amazes me what people define as "good" and then proceed to measure everything else in relation to it. (like, "what a hired guy does must be good," despite the entire audience cringing, including themselves)There's also the "happy face EQ", which is a poor approximation of theFletcher-Munson Loudness Curves, and often "rubber stamp" applied to every output just as a matter of course. Because it's so common, I would not be surprised at all to see speaker cabinets designed to have that response *in addition to* the "rubber stamp" that people still apply as a matter of course. (so that those that don't have an output EQ still get that response anyway, and those that do get it twice as strong) As an immediate A-B comparison, it sounds more "exciting" that way, mostly because it kicks you in the chest that much more, but it also cuts out the midrange just like you describe.Get a skilled, active FOH Engineer, that doesn't act on emotion, in the middle of the audience (NOT in a corner!), give him free reign for real over the *entire* system (NO sacred settings), not just during the gig itself but for at least twice that time beforehand as well, with the band available for all of it so he can get to know you and how you play, and all of that again for a second gig, and then see where you end up. I'll bet that the exact same gear sounds a TON better after you do that! If it doesn't, then you probably got a dud engineer. Do it again with someone else.But when you go to a different venue, or change gear, or change the band's lineup, or even have an off-mood, it's going to sound different. That's why you need an *active* engineer DURING the gig. Not to mention the audience responding differently...Having let the above sit for a while and re-read it, I feel like I need to add this warning: Good audio engineers have left paying gigs because they were treated like machines instead of people. Just another piece of gear that management is responsible for. I've seen it time and again, and been there myself for some of them, where the management buys "the perfect system" from a salesman and gives that pile of **** to the engineer that they've had all along.If you already have the gear when you hire someone to run it, tell him that and keep it for now. He'll understand. Especially if you ask him for improvements and take them seriously.If you still need to buy gear, show him what research you've done so far and be ready for him to tear it apart. Work with your operating engineer to get something that's *actually* good instead of just "feel-good" from a salesman. (even if that salesman is your friend, he's still more skilled in *sales* than anything else)A good engineer with bad gear can run circles around a bad engineer with good gear, but that is NOT an excuse to buy bad for him. That's an insult to his skill, even if you think it's good. Use your existing "bad" gear to test his skill, then give him a (real!) budget for improvements that *he* can buy from.

author: IanJ
author_rank: Unknown
timestamp: February 07, 2022, 11:51:34 am
content: I did FOH for many years kicking off with Peavey gear and ending up with Nexo/Allen & Heath/Crown.All i can say is that it impossible to speculate what the issue is without exactly knowing what the cabinets are, how and what they are connected to. Also, any details of the room and positioning etc.Saying that, when we went from Peavey to Nexo cabs the clarity went up a huge notch.Set and forget is impossible unless you are a one man and his guitar and the venue is really small......you might get away with it 75% of the time then.Ian.

author: Bassman59
author_rank: Super Contributor
timestamp: February 07, 2022, 05:34:21 pm
content: Quote from: CaptDon on February 05, 2022, 05:41:51 pmA common configuration is the 15" woofer and a horn mated to a 2" exit/4" diameter compression driver. Obviously not ideal, but we used to run the tops down to something like 800 Hz or whatever was the low-frequency horn cutoff because 15" woofers get really beamy, and then there's the problem of a crossover point in the middle of your vocal clarity range. I'm glad I don't have to deal with this stuff any more.So the first problem is: you don't know anything about the rig or the person running it. It could be broken, misconfigured, all sorts of issues over which the person operating it has no control.Or it could be the person operating it. Likely not paid enough, knows that the gear is lame, so the amount of care put into the mix is minimal. Or maybe the operator is the bartender who doesn't care?Now based on your description, you're in a bar, and the budget for audio is exactly zero. They got something installed that was probably fine when it was new and now it's all beat up and unless the bar owner sees any reason to spend money on it, it won't happen.Money.Money.Money.If you are in a venue meant for performance, where tickets are sold, even the small clubs around here have fine systems that sound good. If it doesn't, then the venues are not long for the world, as booking agents won't put a tour through the place because bands and their crew tell the booking agents about every venue, and a club can't survive on local bands only.If you are in a bar then the live band is beside the point. Nobody cares about the band. Wear earplugs, or don't go out.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 07, 2022, 06:15:25 pm
content: Quote from: Bassman59 on February 07, 2022, 05:34:21 pmYes! It seems like practically everyone either forgets about or never learnedKeele's Constantin the first place. That article is mainly about horns, but the math applies just as well to anything else. Just replace "horn mouth" with "radiating surface". For a 15" woofer, for example, "d" in those equations = 15". Or you might include the baffle in that dimension and see what that does; the difference isn't really all that much for a minimal-size box to hold the drivers.The absolute, cannot-be-broken, rule from physics is that you can expand the pattern from what that math says, but you cannot shrink it. So the constant-directivity horn tweeter does exactly that, per the published spec, and the crossover is factory-set to where the woofer's beam angle matches the tweeter's. The woofer is entirely governed by Keele's Constant, and has nothing whatsoever to do with the published spec.Thus, a crossover in the middle of the vocal range, and midrange that already starts "wrapping around" the cabinet, not to mention mid-bass and below. Pay attention to feedback frequencies when using that type of box; most of them are probably below the crossover because the coverage pattern is wider there than spec...This also explains why there's not always a difference in bass response between the 10", 12", and 15" versions of what is otherwise the same box. They're not going for bass; the bigger woofer allows the spec'ed coverage pattern to apply to a lower midrange frequency.

author: CaptDon
author_rank: Super Contributor
timestamp: February 07, 2022, 06:38:17 pm
content: The venue is a bar, and typical of all of the animal clubs, Elks, Moose, Eagles, American Legion, Whatever. My observation is that the 15" plus Horn cabinets seem to sound bad no matter who is mixing the sound, and in my experience these types of cabinets belong to the band. No sensible club would run 2-Way cabinets for Front-of-House. People DO care about the band and its sound!!! That is how you get your bookings and garner a faithful following. In both of my recent cases represented in my first post the band was using very similar cabinets, but different brands. Both bands consisted of Excellent musicians, but sadly with a F.O.H. system that sucked!!! As one client who knows I do pro-sound said to me "For as loud as they are I can't understand anything they say". There in lies another problem, when your F.O.H. system sucks playing it at Molly Hatchet Concert Level becomes even less intelligible. Little 4 piece local bands never seem to grasp that idea. Also, a lot of small bands don't Bi-Amp their shitty cabinets relying on the passive crossovers which often put a dent in any type of active EQ you throw at the power amps. Maybe my original post should have said "Does anyone make a decent sounding 2-Way cabinet", but sadly I think I know the answer to that. And again, sadly, those 2-Way cabinets seem to be pushed onto the 'local band' guys who are tight on budget. A great band with a piss-poor sound system will never be recognized as a great band.

author: CaptDon
author_rank: Super Contributor
timestamp: February 07, 2022, 07:03:09 pm
content: When I look at the frequency ratings on a lot of 15" woofer stuff I see them as being rated at maybe 3 to 6 Db down at numbers up to as high as 3 Khz. I have come to a conclusion that with a slow frequency sweep to generate the 'sales brochure' SPL curve the large cone is being actively pushed and pulled for several if not many cycles at a narrow frequency spectra at any given timeframe such as .5 seconds of the sweep. But I feel the '3 Khz' rated woofer sounds Muddier than the Mississippi at that same frequency with a broader spectra signal like vocals. You can push a kid really high on a swing, but not on the first or second push. You can probably get sales brochure frequency response at 2 or 3 Khz from a 15" woofer, but not with a 2 or 3 cycle burst at the same tone. Just my observation of why the vocals are pure mud on most small underfunded 'local bands'. The sales brochure frequency curve shows nearly flat response through the vocal range, and yet the vocals sound like a wet washcloth over the microphone.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 07, 2022, 08:01:43 pm
content: Quote from: CaptDon on February 07, 2022, 07:03:09 pmYes, "consumer" speaker and amp specs are both horribly inflated and dishonestly measured. Peak power instead of continuous average (Can you really plug your passive subwoofer directly into a wall outlet? 'Cause that's essentially what the inflated spec says...), special test signals, wacky math, etc. True professional systems that are designed by real engineers instead of by marketers, have much more reasonable and useful specs. So they may not look much better compared to an over-inflated consumer one, but there's a difference in scale of maybe 10x between real-world accurate and inflated marketing.Quote from: CaptDon on February 07, 2022, 06:38:17 pmI certainly agree with you on that! I've never been impressed with a woofer-tweeter box either.Something I've wanted to do for a long time now, is to build at least one, maybe even a set, of multiple-entry horns. They'd have to be big to handle bass(Note A1), but the idea is to have every driver dump into the same horn, with individual-driver EQ and limiting(Note A2), and delays to make the waveforms line up as they exit the mouth(Note A3). Compression tweeter in the throat, surrounded immediately by high-mids on the sides, then low-mids farther out on the sides, then "subs" farther out still. Each set of drivers should be closer than 1/4 wavelength to the throat for its entire range, which makes the high-mids kinda tight and the "subs" easy. If you get it right, then the crossover disappears, which makes it perfectly okay to cross over in the vocal range, or to run the "subs" a bit higher than usual. The overall effect is of a true full-range single point driver.Notes A:Minimum 1/4 wavelength long, which is ~9 feet for 30Hz, so I'd call it 10ft to allow for some "microweather". You can fold it, but the total acoustic length through all the folds must still be 1/4-wave or more.NOT system EQ and limiting at that point.Each driver is delayed by its distance out from the farthest-back driver.I think it could be done with large sheets of plywood or MDF for the square, flat sides that directly control the coverage pattern(Note B1), to which you attach some crude boxes to surround and seal the back of each driver(Note B2), and put a small hole in to connect that driver to the horn proper(Note B3), and a bunch of small DSP amp channels that all receive the same full-range signal. The DSP amps handle the crossovers, driver-correction EQ, driver-protection limiting, and acoustic wavefront delays, to produce a near-exact match from a measurement mic to the original signal. Once you have that, you can then add a system EQ to boost the bass or whatever, but NOT as a "rubber stamp"! Leave it flat at first, mix the band, then move the similarities in the channel EQ's into the appropriate submix or master EQ's.Notes B:Above the turnover frequency according to Keele's Constant, anyway. It still widens out below that, but you'll probably have a huge mouth compared to what you're used to, which will keep that control to a much lower frequency than you're used to.The correct 3D volume matters more than shape for this.This hole is quite a bit smaller than the driver behind it. This:Makes an acoustic bandpass, so that out-of-band distortion artifacts from the driver itself are difficult to hear.Further increases the air pressure that the driver connects with, for better impedance matching and therefore efficiency.Reduces excursion (because of the air pressure), which also reduces non-linear distortion at the source, so that there's less for the acoustic bandpass to filter out.Makes it more important to limit each driver electronically instead of by ear. There are stories of people who built bandpass subwoofer enclosures, for example, that were surprised to find that the driver had burned out. In fact, it *had* been distorting horribly, but the acoustic bandpass enclosure prevented them from hearing that.For tools and materials, two places that I like are:https://www.hometheatershack.com/threads/hornresp-for-dum-hmm-everyone.36532/for a good, free, though mostly non-graphical simulator. It doesn't have a CAD frontend, but it does have graphs that update in real-time as you tweak a numerically-entered design. So I switch back and forth constantly between that and a free, open-source, general-purpose CAD program, until I get something that both fits in the space *and* simulates well.https://www.thomannmusic.com/for better deals than in the US for the same gear, even after shipping from Germany to the US.I don't have a favorite for speaker drivers, but when you go shopping for them, make sure you get a decent spec. sheet! Not just SPL @ freq. (that number is going to be wrong anyway, as a horn driver), but all the fiddly bits of how it behaves so that the simulator can work with it.Build a prototype first, as best you can with just the simulator and CAD combination, measure it, reverse-engineer the simulation to match the measurement, and then build another one to account for the errors in simulation / construction.

author: themadhippy
author_rank: Super Contributor
timestamp: February 07, 2022, 08:35:23 pm
content: Quotetry giving a martin le1200 or L'acoustics fm115 a listen,even ye old ,much cloned martin le 400 can be made to sound rather nice.

author: Bassman59
author_rank: Super Contributor
timestamp: February 07, 2022, 08:38:54 pm
content: Quote from: CaptDon on February 07, 2022, 06:38:17 pmThose "Animal" clubs are not a performance venue that charges admission to the general public to see shows. Yes, they may host shows brought in by perhaps an outside promoter -- I've done enough of those gigs. But when I did those gigs, it was with a PA system we brought in. It was appropriate for the venue and it sounded pretty darn good.But I reiterate again: you're not dealing with "sensible clubs." You said it yourself: "pushed onto the local band guys who are tight on budget." Whoever is paying the bills has chosen to skimp on the sound system. This is the whole of the truth. The rest is commentary.

author: Bassman59
author_rank: Super Contributor
timestamp: February 07, 2022, 08:40:29 pm
content: Quote from: themadhippy on February 07, 2022, 08:35:23 pmOr some of the small EAW JF boxes, or Nexo PM, Meyer, or ... there are a bunch of great-sounding two-way boxes (and you need subs, of course), but ... they're not cheap. You really do get what you pay for.

author: Marco
author_rank: Super Contributor
timestamp: February 08, 2022, 05:56:00 pm
content: Hasn't room correction technology penetrated into this market yet? Why not put a zero latency phase+amplitude corrector on each speaker and just get stuff optimally flat for the entire venue with science instead of artful tweaking?

author: AaronD
author_rank: Frequent Contributor
timestamp: February 08, 2022, 07:37:02 pm
content: Quote from: Marco on February 08, 2022, 05:56:00 pmWHERE in the venue? Get one spot *exactly* right, and everywhere else is awful. The best you can do, even with a perfect transducer, is "close", with the definition of "close" varying wildly between venues. Acoustics are insanely complicated to simulate in detail, and it doesn't help that most spaces are like a funhouse hall of mirrors, with dimensions comparable to the wavelength so that interference and standing waves are also significant.Besides, the ultimate goal is actually the result of our ear-brain processing. A measurement mic is really an intermediate point in that chain. As far as I've seen, no one's been able to describe our biological processing in technical terms so as to write a DSP algorithm to copy it. If there was, then it would be AWESOME for hearing aids! And it could also be fed from a "foam head" stereo mic, to (almost) accomplish what you describe above.But as it is, we just have a single mono mic or an array of mono mics around the space, all of which confuse the direct sound with reverb, and a relatively dumb autotuner that tries to make *that* match the original signal. The result is...suboptimal because our biological processing inherently separates direct ("what") sound from reverberant ("where"), and the autotuner with bad inputs has essentially optimized the reverb at the expense of direct. For an outdoor rig with no reverb, that works; but for every other venue, it's chasing the wrong thing.You also need about 10x the acoustic power (+10dB) to deliver the same SPL to the audience anechoically compared to the funhouse. In addition to system specs in the first place, that should also give you an idea of how difficult it is to remove that reverb from an indoor system measurement.There is also an "X-Curve" and probably a few other "rubber stamp" EQ curves that are supposed to be applied after the autotuner is finished, in response to the bad but consistent result that the autotuner gives, and I'm sure that that step is built into a lot of autotuners now. (does Smaart do that?) But it doesn't really solve the problem. It's still a "rubber stamp" on top of a "garbage in, garbage out" kind of result.This article is specifically focused on movie theaters, but the concepts apply everywhere:https://sound-au.com/articles/cinema-sound.htmAll of that being said, most powered speakers today (amplifier built-in, so they take a constant-power cord and a line-level signal input) also include a non-user-accessible DSP, which handles the cabinet EQ, crossover, per-driver EQ, and per-driver limiting, so that an anechoic measurement on-axis (notice how specific that is) already matches the original signal plus optional non-flat EQ, and you can't blow it up by casually overdriving it. (it still sounds worse at high volume though, because one driver limits before the other) If I had a separate system EQ on a powered rig, I'd either leave it flat, or use it as an artistic thing. I would not tune the room.For passive speakers (no amp, just a passive crossover and the drivers, so they take a single combined power+signal input that should be respected as if it were wall-power,especially if it has a guitar-style plug on it that exposes that hazardous voltage), they're really not all that bad sonically...ignoring of course, the inherent problems with the box design, and assuming that it isn't designed for a "happy face" response in the cabinet itself. It can help to put a system EQ on a passive rig and tune it anechoically, but if you can't do that right, then don't at all. They're really not *that* bad to start with, given the aforementioned caveats. And I would still not tune the room; only the speakers.If the room misbehaves, FIX THE ROOM! Leave the PA alone. Don't mess up the perceived sound to fix something that today's machines get confused by.If a machine can produce a signal that actually matches our perception (see my second paragraph), THEN you can use that machine. Until then, your ears should be the final authority, using actual program material for the "test signal", not noise, and also being aware of "fatigue", where everything sounds the same anyway after a few minutes. So take frequent breaks to "unfatigue" if it takes you more than about 1 pop song or so to tune it by ear.(though if you have to tune it live with an audience, the audience fatigues along with you, so...The trick then, is to have it right for the *second* audience. Maybe a second engineer that isn't fatigued yet?)

author: Marco
author_rank: Super Contributor
timestamp: February 08, 2022, 08:59:14 pm
content: Quote from: AaronD on February 08, 2022, 07:37:02 pmThat's why I said optimal flat in the whole venue. There's a little arbitrariness in deciding what to optimize for.Hall of mirrors, interference, standing waves ... it's all just linear. You need long filters to deconvolve some of them, but long filters are not a problem with zero latency FFT based filtering. Any way, it has not really penetrated into the market then I guess.

author: themadhippy
author_rank: Super Contributor
timestamp: February 08, 2022, 09:40:59 pm
content: Not only were ,but when?no use applying your automated eq on an empty barn,and the last thing you want as you step up to the desk just before the band wanders on stage is for the systems eq to change .with a manual system you can set it before sound check,making allowances for the lack of breathing sound absorbers,when you get to show time your in control of how much and when the tweaking takes place .

author: CaptDon
author_rank: Super Contributor
timestamp: February 08, 2022, 10:02:09 pm
content: For my own personal small to medium venue (apprx 250 persons indoors) I have a Rane RE31 31 band eq with real-time analyzer. At one time I thought using the 31 band eq and calibrated mic to 'tune' the system to the room was the way to go. I kept trying to convince myself this must be right, but time after time the sound ended up being lifeless and no where near pleasing to the ear. I also found moving the mic 3 inches changed everything. Now I start off flat, use the eq to reduce perceived resonances and/or fill hollow spots. The real-time analyzer just kind of ends up being eye candy for people who like to 'hang out at the mixer board' and shoot the breeze during breaks. Always the same questions, "How many watts", "How loud can this thing get", "Can you announce my girlfriends birthday", "Whoever owns the blue Toyota...Your lights are on", but the one that always makes me smile and actually feel good "You guys sound great tonight, Wow, your setup is really cool".

author: themadhippy
author_rank: Super Contributor
timestamp: February 08, 2022, 10:13:17 pm
content: Quoteyou missed out "can i get a set list/plectrum/drum stick"

author: Ed.Kloonk
author_rank: Super Contributor
timestamp: February 08, 2022, 10:34:01 pm
content: I know peeps on here prolly know a lot better than most. However...When you say that you can't hear mid-range so well, I gotta ask.. it's not your hearing is it? Classic symptom.Like all sensitive test gear, you sometimes need to check the calibration of your own ears.

author: Marco
author_rank: Super Contributor
timestamp: February 09, 2022, 12:53:09 am
content: Quote from: themadhippy on February 08, 2022, 09:40:59 pmBut who can manually control a hundred thousand tap FIR filter? In theory echos and delay matching imperfections can be removed in a static situation, ignoring if it would be robust enough to deal with change in audience density, a human could never control all the variables period. After the setup, delay and EQ are about all the human can handle, EQ isn't room correction.

author: CaptDon
author_rank: Super Contributor
timestamp: February 09, 2022, 01:30:39 am
content: As both a ship captain and aviator I get my hearing tested every time I go for a U.S.C.G. or F.A.A. or CDL physical. None of the physical exams are accepted by any of the other agencies however the drug screening is. I go broke proving I am in good health with good hearing and eyesight along with low risk of heart attack to three different agencies. I am however nearly stone deaf to frequencies above 14Khz but now in my mid 60's that is kind of the norm for men. I hear 12Khz like a kid, 14Khz is down 10Db, 15Khz up GONE with the wind, or maybe tinnitus? Cheers mate!!

author: magic
author_rank: Super Contributor
timestamp: February 09, 2022, 09:58:49 am
content: Somebody here hangs out at A$R and believes everything he reads there

author: Bassman59
author_rank: Super Contributor
timestamp: February 09, 2022, 03:37:48 pm
content: Quote from: Marco on February 08, 2022, 05:56:00 pmAs Sam Berkow, who wrote the first versions of the audio system analysis program Smaart, likes to say, "The only way you can tune a room is with a bulldozer."What he means by this is that there are no electronic solutions to architecture problems. If there's a suckout at some frequency caused by reflections, there is no "equalizer" (a word I use in the general sense, maybe "corrector" is better?) that can fix that problem. Move a meter in any direction and the depth of that suckout changes.We can, and do, use DSP to manage the loudspeakers. Modern line arrays wouldn't work without it. But then there's the real issue of hanging that line array in a venue and the interactions with the room.Now there are things like the dbx DriveRack which have a microphone input that feeds an RTA, and it can do some equalization based on what the mic hears. But it's not very effective because it hears only at one point in space and you have to optimize the sound system for the entire room.Oh, and there's no way to do that kind of DSP with zero latency.And for the third time: we are discussing bars with owners that refuse to spend a penny on better speaker cabinets, so to expect them to get a DSP system and (gosh!) hire someone with skills to set up the rig properly and then run it every night is pure fantasy.

author: Marco
author_rank: Super Contributor
timestamp: February 09, 2022, 06:02:21 pm
content: Quote from: Bassman59 on February 09, 2022, 03:37:48 pmLets do a thought experiment for a moment. Lets say the architecture creates a nice anisotropic suckout around a given location in the room, what would happen if you put one extra anisotropic speaker in that exact location, could you then correct most of the suckout? Obviously yes, no matter the reflections that speaker is stronger around that spot, it can overrule them. It's a question of degrees of freedom. Just applying EQ to the source obviously doesn't have the degrees of freedom to correct for bad architecture, but enough speakers with long arbitrary FIR filters are a different matter.HiFi room correction can correct for multiple listening positions nowadays.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 10, 2022, 03:07:49 am
content: Quote from: Marco on February 09, 2022, 06:02:21 pmI think you've just described the front fills in a large professionally-designed system. But I doubt that they're as complicated as you describe. It's just easier to design the main PA to start a bit farther out so as to not spill onto the stage so much (acoustic feedback!), and then put some smaller speakers on the stage facing the audience to fill in the hole that that leaves in front. If there's a hole in the middle of the main coverage pattern, and you can't get the main rig to cover it, just leave it. (maybe put a "volcano" fog machine there or something) You might succeed in fixing that hole, but the sound from that added speaker doesn't stop there. The "overspray" may well interfere enough to create other holes or audible weirdness somewhere else, and so it's just not worth dealing with.What you REALLY don't want is for theHaas Effect or Precedence Effectto end up working "backwards" in some parts of the venue, where a blind person would insist that the band is in that little fill speaker instead of on stage.There's already a fair amount of engineering to make the front fills play nice with the mains: likely a speaker EQ copy/pasted direct from the manufacturer with no changes at all, plus some high-/low-pass filters to account for the radiation patterns changing based on frequency (if the mains already spill into that space at a given frequency, then the fills are filtered down at that frequency; no sense in covering it twice), plus some delays and/or all-pass filters to make the edges of both patterns line up in time and phase in the parts of the audience where they have similar SPL at similar frequencies. But not much more design than that. (it's already a lot, despite only caring about the audience and not the entire 3D space, and only about *relative* phase between speakers for the purpose of interference and not at all about absolute phase from mic to speaker)Complex filters with weird responses just aren't used here, outside of a gimmick...which does occasionally happen, but doesn't last long. They're just too fiddly to set up differently and tear down all the time, and the 5% or so that they might legitimately add just isn't worth that. The broad sweeping strokes on top of an array calculator (that takes *a desired pattern* as its input, *not* a seating architecture), with someone that knows how to run those tools and what GIGO looks like in that context, will already get you maybe 80-90% there depending on the venue, and that's more than enough for the non-tech public that defines "good" by "what a professional does".Quote from: Bassman59 on February 09, 2022, 03:37:48 pmIn terms of the OP, that's absolutely right! But there's also a side of it that is exploring the possibilities and what is normally done when there *is* a decent budget available. Maybe when a small amount of money comes up for that purpose, someone will remember this and not squander it on an audiophool magic rock or something like that.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 10, 2022, 03:38:31 am
content: Quote from: CaptDon on February 08, 2022, 10:02:09 pmYeah. When I was learning / teaching myself, one of the systems that I ran had a dbx DriveRack with a measurement mic. I used to "auto-EQ" per the instructions before every gig, and every time it would come up with something different, despite being the same rig in the same room. I thought it was odd when a guest engineer put the measurement mic close to a speaker instead of out in the room like the instructions said to do, and I didn't know enough then to understand his explanation either, so I don't remember what he actually said.Many years later, I found the article that I linked to earlier about "room correction" and the stuck-record mantra of"You cannot fix time with amplitude!"THEN, I understood why the room mic didn't work and what the guest engineer was doing. (or at least trying to do) I still didn't understand how noisy and just *how* easily influenced that kind of measurement is, even with test tones instead of actual noise, so I used a close mic with short tone bursts to calibrate a completely reworked PA in a local youth center. (only the flown passive mains and ground-stacked passive subs were still original) A 1/3-octave (31-band) analog GEQ with a cover, still ended up with some sharp corners in it, which I doubt are real.And every once in a while, I think it'd be a good project to make an analog RTA from a bunch of bandpasses followed by LED bargraphs. Hook it up to the Solo/PFL/AFL/whatever your favorite mixing desk calls that output... Sigh...one of these days, along with the other analog audio projects that have been piling up since DSP became cheap...

author: AaronD
author_rank: Frequent Contributor
timestamp: February 10, 2022, 03:59:58 pm
content: Quote from: Marco on February 08, 2022, 08:59:14 pmIf it hasn't "penetrated into this market", is there another industry that does use it? I wonder if the problem is better defined over there than "a random found/rented space", or if the wavelengths are sufficiently different from the geometry that they can all be treated the same way, or if the receivers (like most electronics) are dumber than human auditory perception, etc.Or is your thought process more theoretical, which includes the awful human ability to estimate the scale of things, including the number and nature of real-world problems?

author: Marco
author_rank: Super Contributor
timestamp: February 10, 2022, 04:35:24 pm
content: Ehh,a little googleshows it's starting to be adopted already. Though in this case based on simulation rather than measurement, not really a product suitable for a bar. My intuition ain't that bad it seems.Human craftmanship in speaker setups only adding delays and EQ will always leave something on the table for more advanced filtering, ie. arbitrary FIR. Human craftmanship can't design the optimal filters.PS. the google search went from "venue correction speaker filters" to thisAES paperwhich hada patentin its citing links which though not really applicable looked like it was from a company on the same wavelength as me ... and it was.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 10, 2022, 07:46:55 pm
content: Yes, there are all kinds of things being tried and patented. But neither patents nor research reports are required to work, and even if something does offer an improvement, it needs to be almost foolproof to be adopted at this point in the game. A new thing with most of its warts still present is going to be compared to the refined version of what came before.Frankly, we really don't care how the result comes about, except to do as little processing as possible to get there because everything has side-effects. If a "black box" is at least as easy to use as what we have already, and consistently produces better results, then we'll...s l o w l y...adopt it.That said, the cool graphic on the AFMG page looks an awful lot to me like a designed pattern that an array calculator is already capable of producing with only delays, all-pass filters, and gain (could be negative) between many boxes in a simulated anechoic space; which was then overlaid on the architectural drawing that the human engineer used to create that pattern to feed the calculator. And I'm sure it was cherry-picked and simplified (probably a single frequency) to have the entire audience end up at (almost) the exact same SPL. I wouldn't be surprised either, if the balcony and the main audience were opposite polarities (either a negative gain or a 180deg phase shift, depending on some other details) to make the hole in between.I'd like to see a GIF of that same image, where each frame is a different frequency with the exact same processing, and a 3-D acoustic simulation of the actual venue that includes the civil engineering structure, architectural decor, seats, uneven distributions of people, etc. A venue sim with that level of detail is probably still limited to a time-rented supercomputer. Get that kind of computing power into a laptop, and we might actually see some widespread improvement. But until then, we're stuck with "open-loop" designs with minimal feedback because a human has to provide that feedback.If the (usually closed source) array calculators already use the techniques that you describe, then that's fine. It's a "black box" that is offered to the market to produce "a result", given some input. If the tradeoff of work for result is acceptable, in full detail for both, then it's used.Keep in mind though, that there are still some fundamental rules that absolutely cannot be broken. For a FIR filter, one of those rules describes the tradeoff between phase and latency. "Friendly phase" is absolutely NOT "low latency"! But for a PA that is already far from the audience, maybe you don't *need* low latency processing. At 48kHz, for example, one sample is just a bit over 7mm in "standard air". Delaying by, say, 1k samples to make the phase nice, has a similar effect as pushing the speakers directly away about 7 meters but without the SPL loss. If they're already 10's of meters away, then maybe an extra 7 is okay. If all the speakers are delayed by the same amount, then the math for interference is unchanged...except for their interaction with a mic'ed acoustic source.(But if the PA is much louder in the audience than that source is, then maybe that doesn't matter either, except for the Haas/Precedence Effect.)Another thing to consider is delay towers, which are essentially an additional PA farther back in the audience, so that the mains up front don't have to throw that far. They have their own design, part of which interacts with the mains' design so that the physical crossover region works, and they're delayed of course, to match the speed of sound. If the delay towers are not directional enough in *all* dimensions (a line array is only a 1-D structure), you can hear them from the front section as a distinct echo...

author: Marco
author_rank: Super Contributor
timestamp: February 11, 2022, 02:40:06 pm
content: Phase is the wrong way to look at a correction filter, it's far more natural to think of the problem in the time domain than the frequency domain. It won't have a nice phase any more than random reflections have a nice phase, otherwise you could never cancel them. Instead of one delay with a given (negative) gain you can have lots of them for a single speaker.Compute is not a huge problem unless you want to iteratively improve speaker positioning. With a fixed setup getting the (directional) impulse responses for every speaker to a set of listening positions is a once and done calculation. Even if it takes a couple days of cloud computing time (much less actual time) it's not all that relevant. Bigger problem with simulation is trying to accurately model anything. Easier to measure than model, even if you need an audience.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 11, 2022, 03:47:36 pm
content: Quote from: Marco on February 11, 2022, 02:40:06 pmI was thinking about interference between speakers, not the sound of one speaker by itself. If you're designing an acoustic crossover between mains and front fills, for example, then phase makes a difference. But if you only have one speaker per signal(*), then nobody cares about phase at all. (except possibly audiophools, and that's only after you tell them) Our ears certainly don't.(*) A mono source panned center in a 2-channel stereo rig, is not one speaker per signal. It's two speakers producing one signal. NOW we have interference patterns in the direct sound. (as I've said before, reverb doesn't matter as much to our perception) That's mostly what the big deal is with a 3-channel left-center-right (LCR) rig, for those like me that advocate it: everything critical goes to the center channel only. If you're in the round though, that's not really practical; for that, go back to "stereo" (with half the audience only hearing one channel anyway, and half of the remainder getting a reversed image), or even mono.Quote from: Marco on February 11, 2022, 02:40:06 pmSo...you rent a stadium, which costs the same regardless of who's in it and what they're doing, and you spend more time measuring and calculating than you do running the show??? Maybe even pay an audience to do nothing but fill the space and have their conversations (that you can't stop) interrupted by test tones?Not to mention that I'm pretty sure the kind of mic that you describe is physically impossible. Either it picks up everything without telling you where from (omni), or it changes frequency response based on both distance and direction (cardioid and pretty much every other directional pattern), still without directly telling you where from. That's a physical tradeoff, not an engineering one.I'd be surprised if a professional show actually measured the venue acoustically, in the amount of detail that you describe. Way too much time and effort. Instead, take the architectural drawings, design an anechoic coverage pattern to match that, set up the PA to produce that pattern, and then spend an hour or two at the end of load-in to fudge it from there. Then hand it off to the FOH Engineer, who fudges it some more during sound check(*), and again during the first few songs with an audience.(*) Usually virtual now if there is one at all, playing a multitrack recording of a previous performance, through all of the live processing, instead of the band actually playing.For a fixed installation like you mentioned, you'd technically be right, but since the vast majority of sound installations are temporary as I described, the entire industry tends to think that way, even for permanent things. Thus, line arrays for everything without a second thought, for example, despite their primary design goal being *portability* at scale, with sound quality farther down the list. And since the industry is already optimized for that and audiences are used to "that sound", it also becomes the cheapest way to get what people like. Hire the same crew to set it up the same way, and just don't take it down.Pretty much every testimonial (on a public forum, not connected to the manufacturer or installer) from those who installed horn-loaded point sources instead of line arrays in their permanent space, has been blown away by the seemingly effortless sound quality. But those are still very much the minority, with most installations being the "industry standard" that just gets tossed up according to the "standard practice" for touring gigs.Essentially, if you want to change the sound industry, even if you're only focused on permanent installs, you need to make it work on tour first.

author: Bassman59
author_rank: Super Contributor
timestamp: February 11, 2022, 05:08:02 pm
content: Quote from: Marco on February 09, 2022, 06:02:21 pmI mixed a show at the Pritzker Pavilion at Millennium Park in Chicago. It's an outdoor shed, and it has a LARES system for active acoustic corrections, set depending on the type of performance. They do everything from orchestra to jazz to rock bands (this last was me) so the entire facility was designed with these differences in mind. The show was pretty fabulous and the PA was tops.I'm sure this all cost millions of dollars. Money absolutely well spent.And again, for the nth time, the OP was talking about a bar somewhere, with $0 budget for audio. Expecting to add any kind of active correction system fed from a processor that has to be set up by a professional is ... unrealistic. The venue wants two speakers on sticks.If you haven't been in the world of touring/club audio, then I humbly submit that you listen to those of us who live it.

author: Marco
author_rank: Super Contributor
timestamp: February 11, 2022, 05:33:59 pm
content: Or you could have answered my question and told me AFMG was already publishing software for the concept but it had little penetration as of yet. AFMG is a big player.

author: Bassman59
author_rank: Super Contributor
timestamp: February 14, 2022, 11:38:15 pm
content: Quote from: Marco on February 11, 2022, 05:33:59 pmto whom were you addressing your comment?Anyway, SysTune competes with Rational Acoustics' Smaart. And I know many installers use EASE for speaker/system/room modeling.

author: Marco
author_rank: Super Contributor
timestamp: February 15, 2022, 08:37:25 am
content: My initial question was if room correction technology hadn't penetrated the industry yet. It didn't have to become a huge discussion about applicability, the answer was simply yes but...FIRmaker is clearly similar to room correction, just not much uptake yet and only simulation based so only useful for large venues (getting a 3D representation isn't even the hardest part, good material models for surfaces more so). AFMG thinks can work, it just isn't applied much yet. Electronics get ever cheaper, a modern microcontroller can already implement multisecond FIR filters for audio.IMO someone is going to implement something like LARES based on room correction techniques based on multiple measurements through the venue, it's the clear evolution of the concept. It also doesn't need to be expensive.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 16, 2022, 02:08:02 am
content: Quote from: Marco on February 15, 2022, 08:37:25 amIt doesn't matter who likes it, what their qualifications are, or how much money they throw at it. Electronic room correction from a localized PA still remains physically impossible. (for this purpose, even a line array is localized) You need speakers literally everywhere throughout the space, all coordinated, to even have a chance at it, and nobody is doing that. You're right about a distributed PA like that needing to be designed algorithmically, because no human could ever hope to manage something that complex with all of its interactions. But even still, "You cannot correct time with amplitude," and so the acoustic design of the room itself will always be important, regardless of how much electronics you throw at it.Anyone that claims to "tune the room" electronically from a single source of sound, loses a ton of credibility with me, because they're essentially claiming to have rewritten physics. You can't do that.It's easy enough to make a measurement mic look flat on a screen, or even an array of them, but that's not what we hear... (this thread has been there before, so I won't repeat it)

author: Marco
author_rank: Super Contributor
timestamp: February 16, 2022, 04:37:20 pm
content: Quote from: AaronD on February 16, 2022, 02:08:02 amWave Field Synthesis is essentially a line array taken to the extreme and can virtually position sources any where in the space. Now the spaces where WFS are used are designed mostly anechoic, but the same principle applies in a less than ideal space with less speakers. The number of speakers give you degrees of freedom to shape sound in one place without affecting sound too much everywhere else. The more speakers on average which can reach useful volume for positions in the venue, the more you can shape.QuoteBut that's not what you're doing when every speaker has it's own FIR filter. You are correcting time+amplitude with time+amplitude. That's what a FIR filter is, a set of amplitudes and delays (instead of a single delay).QuotePut the microphone in a binaural dummy head and make sure group delay is flat too and it's pretty damn close.PS. even ignoring room correction, with arbitrary FIR filters you can also do beam forming a hell of a lot more accurately than if you just got a single delay and (negative) amplitude to work with.

author: AaronD
author_rank: Frequent Contributor
timestamp: February 24, 2022, 06:44:08 pm
content: Quote from: Marco on February 16, 2022, 04:37:20 pmI think you finally said something that makes sense. Thank you!Though I still doubt that it would be practical for a time-rushed setup and teardown that the vast majority of the audio industry has. For that use-case, it's more than enough to pre-design a coverage pattern that hits the audience and nothing else, toss it up at the last minute per that design, and just go with that. Not spraying the walls and ceiling goes a long way by itself, even if there *is* a small amount of overspray and incomplete attenuation through the people, off the hard floor, and back out to the room.If I remember right, I think Pink Floyd experimented with a quadraphonic concert, with a PA stack in each corner. It was a good experiment, but nobody else copied it. WFS seems to me like that kind of a rig on steroids, regardless of the actual number of channels or the processing behind it. Too much work to set up and run, compared to the audience experience.From my own experience as a FOH Engineer, a stereo rig can be just as much of a cognitive load as two separate, simultaneous mono's. It's just a different set of things to keep track of, with a similar total. Expand that to quad or practically infinite, and the workload goes up even more. Not to mention that you have to explicitly look for a mixing console that can smoothly move a single source across more than 2 channels. (at least, that's how it is on my end of the price spectrum) Of course, you can have more processing channels than engineered channels, but it's hard to realize much of a benefit with too few engineered channels.

author: themadhippy
author_rank: Super Contributor
timestamp: February 24, 2022, 08:16:27 pm
content: QuoteBut that was more for theatrical effect than audio quality,when they first tinkered with it even stereo live sound wasn't common.The grateful dead's wall of sound was an interesting idea for arena scale rigs

author: AaronD
author_rank: Frequent Contributor
timestamp: February 26, 2022, 06:00:52 pm
content: Quote from: themadhippy on February 24, 2022, 08:16:27 pmYou might be right on that. In fact, even today, I don't think stereo is worth it for a lot of cases because of how the venue is set up. Most of the audience only hears one channel well anyway.I've actually used the Grateful Dead's vocal mic trick: 2 identical mics equal-distance from the PA and aimed the same (GD used omnidirectional reference mics just for the matching), one polarity-inverted, and mix them with equal levels and identical processing. (digital board, so I could guarantee that the processing really was identical) It would have been better to wire them in reverse-series on a custom cord feeding a single preamp, but I didn't have time or materials at the moment to make that cord. It's not an exact science - the acoustics, mics, and analog preamps all have slight differences - but it did get me about 5dB more gain before feedback.