Bill Fitzmaurice

Designing crossovers is no more difficult than designing boxes, thanks to software like XSim. https://xsim.software.informer.com/ The only issue is the need for FRD and ZMA files. They're available for most hi-fi drivers, with pro-sound you probably have to create them yourself. But it's no more difficult than any other part of the speaker design process. Coils are readily available, I haven't had to wind my own since the 1970s. They are expensive and quite large when used in a passive crossover lower than 200Hz, but that's also something I haven't done since the 1970s. Below 500Hz active crossovers and bi-amping is the way to go. The easiest method to get better mids and highs is with a sealed back midrange, like the PRV 6MR 300 SEAL. Crossed over at 800 to 1200Hz it will keep up with any woofer. As for using higher order high pass filters to run tweeters to 2kHz, I've been doing that for 25 years. The bass cab industry as a whole has yet to catch up... or should I say wake up? 😉

Because they're cheap, or at least the ones you see most often are. They cross over typically between 3.5 and 4kHz, because that's as low as cheap tweeters will go. They use barely adequate crossovers again because they're cheap. IMO the best high frequency option is a four to six inch cone midrange driver, crossed over between 1.2 and 2kHz. They tend to be good to 8kHz or so, which is all electric bass needs. You seldom see them because they're more expensive, as are well designed and spec'd crossovers. BTW, you can't lay the blame for this on speaker designers. Given free rein and a blank check most of us would do it the right way. This one falls on the bean counters.

I successfully gigged with a 50w Bassman once I started using two 4x12 cabs. It wasn't the amp, it was the low displacement speakers of the day. The 'benchmark' Jensen C12 had 49cc displacement. A middle of the road twelve today has 250cc.

Adding a balanced DI XLR output jack to a valve amp is easy if you know what you're doing. The + and - signal is tapped at the input/output of the power amp section inverter, capacitor isolated and resistor attenuated. It's been a while since I've done it but as I recall 0.1uF and 10k Ohms will do the job. It's even possible to use a stereo pot for level control. A good valve tech will know exactly what you mean when you ask about it.

Boundaries will enhance the lows when the sub is close enough..and 1 meter isn't... and will cause low frequency response notches when the sub and/or the listener is in the .75 to 2.15 meter range from the boundary. They have no effect on subs being directionally locatable, as they don't affect the harmonics. One way to both lessen the distance to the wall and attenuate harmonics is to aim the sub at the wall. Try setting the sub low pass to 80Hz. It doesn't necessarily have to be set at the same frequency as the mains high pass.

You're dealing with two separate, although related, effects. Speakers are omni-directional where the radiating plane, the baffle, is less than a wavelength in dimension. At 100Hz a wavelength is 3.4 meters, so all but the very largest pro-touring subs are omni-directional throughout their pass band. Being able to directionally locate a sound source requires a sufficient difference in the arrival times of a sound wave at the left and right ears. This is also related to wavelength, in that the distance between one's ears must be a significant portion of a wavelength for the arrival times to be sufficiently different. By and large 100Hz is the dividing line, coinciding with the pass band of subwoofers. However, the output of subs doesn't abruptly stop at their crossover frequency. They radiate significant harmonic content, especially when pushed hard. Those harmonics can make a sub directionally locatable. For that to be purely distance related said difference in placement difference would have to have been substantial. What's more likely is that the room boundary effects were responsible, based not only on the distance from the sub to the boundaries but also the distance from the listening positions to the boundaries. Getting sub placement perfect is very much an involved trial and error process, one that we seldom get the opportunity to undertake unless we can get into the room during off hours.

I've been saying so for decades. There is a place for fifteens, that place is in subs. There's no reason for them in mains. You lose boundary loading and introduce boundary sourced cancellations.

From the standpoint of frequency response perhaps, but...in order for the high frequencies to be heard main cabs must be elevated enough to place the high frequency radiator above the heads of the audience. In order to prevent feedback the main cabs must be placed in front of the mics. In order to get maximum low frequency extension and output the low frequency sources must be on the floor, close to a wall, and if there's more than one clustered less than 1/4 wavelength apart. You can't achieve these goals without separate subs and mains. That being the case there's no point in paying for and schlepping mains loaded with more than one twelve, while a higher quality ten loaded cab will function better than a lower quality fifteen loaded cab.

Other than JBLs the drivers used in Fenders at least through the 70s were gosh awful.

What Motorola should have said was that piezos could be used without a crossover, which is true. Being a capacitive load they act like they have a built in first order highpass filter. But be it a piezo or dynamic tweeter first order filtering doesn't make the grade in professional use. I use them extensively, with a second order filter that combined with their inherent filtering gives third order results.

By the same token no matter what the signal a highly colored speaker will highly influence the result. This is particularly of interest with respect to vintage sound, as the tone of the electric bass was very much defined by the shortcomings of the available speakers of the day, and they weren't very good.

They key to minimizing feedback from drum mics is to place them as close as possible, ideally with rim mounts. Of course that means using more of them. Overheads are for picking up cymbals.

TLAH Pro uses eights. I didn't create a ten inch loaded version as when you get that large it's more practical to stack smaller single or dual woofer cabs. All of the DR, Jack and OmniTop designs have that ability, and being horn loaded have much higher sensitivity.

There are only two reasons for them thinking that. One is 'That's how everyone does it'. The other is 'It's the only way we know how to do it'. Neither is a valid reason to keep doing it. That's why one person should be in charge of PA, that being the person who is the most knowledgeable.

Where it comes into play with respect to speakers has to do with the potential for damaging the amp. With sufficient phase shift between current and voltage the current and voltage can both be at a maximum. Combine that with low impedance and the result can be magic smoke.

Ask them what the speed of sound is. Every acoustical engineer, pro or amateur, knows that number as well as they know their own name, as it's the prime factor in calculating wavelengths. If they don't know that number and don't know what the importance of wavelengths is they don't get to offer an opinion. They may have one, but they should keep it to themselves.

If it's too close you'll know, by the feedback. Max Weinberg and his band were in town a while back, he had a sub set up maybe six feet to one side. When he played there was a constant droning feedback that drowned out the entire band. I mentioned it to the guy in the booth, he was too busy playing games on his phone to care. That's when I left.

+1. Against the wall is the equivalent of doubling the sub count, in a corner the equivalent of quadrupling it, compared to well out. It also eliminates boundary cancellations that occur when not tight to a wall or corner. But 'adjacent to the drum kit' is a no go. Not only will you have low frequency feedback into the drum mics but you'll also have acoustical coupling of the sub and drums that creates another feedback loop.

The issue with respect to speakers is that impedance isn't a constant figure, it varies with frequency. Volts x amps= watts, but since amps vary with impedance which varies with frequency power also varies across the spectrum. Even if you have the gear required to simultaneously measure both voltage and current across the audio bandwidth the result looks like this, a 2.83v signal into an 8 ohm speaker. That's nominally 1 watt, but as the chart shows there are only four frequencies out of fifteen thousand where it's actually 1 watt.

Perhaps that's what was claimed, but I'd need to see proof before I believed it. I've seen too many examples where the sum total of the engineering involved in speakers supposedly designed to work together was to make the cabinet corners interlock. Having designed more than a few commercial cabs I know from personal experience that far too often the design priorities in order of importance are profit margin, looks, and coming in a distant third, sound quality. For instance, in that Trace 2103x the fives, and preferably the tens as well, should be placed vertically, not spread apart as they are. I take one look at that configuration and think 'comb filtering'. If they didn't get the driver placement right I wouldn't trust them to have gotten anything else right.

Subs at the back of the stage won't feed back any worse than at the front as the output of subs is omnidirectional. Placing subs at the front, and for that matter more than a half meter from the wall, costs output and low frequency extension, so that should be avoided anyway. What matters in this case is the distance from the subs to the mics. But since subs cannot be directionally localized you can move them well over to one side or the other and the audience will never know the difference.

They never have. Way back when Fender put stickers that said 'Special Design' on the driver magnets. The only difference between them and off the shelf was the stickers. 🤪

Unless grossly mis-matched that's true. As a for instance a 410 plus 115 will almost always sound better than either on its own. But a pair of identical 410s is better. Maybe. If true the tens would be guitar drivers. But even then if not bi-amped with an electronic crossover the tens are seeing lows and the fifteen is seeing mids and highs that they shouldn't.

How drivers work is primarily defined by their Thiele/Small specs, not the magnet material. It's those specs you need to compare. However, in general it's easy finding Waldo than OEM driver specs. Where low frequency output is concerned the most important spec is excursion, xmax. If they're not very close you can have one cab comfortably cruising along while the other is distorting badly.

No comment on particular models, but what makes a mixer good for the studio is having as many output channels as you plan on recording with. That makes a two channel mixer that's adequate for the stage not so great for recording. But if you have a mixer with eight or more outputs sooner or later you'll find a use for them on stage.