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MK3 serial numbers begin with 30. All previous generations begin with 14.

The SV9 allows for a 90° rotation of the horn diffuser to mount the speakers horizontally instead of vertically.

If the cabinet were rotated 90 degrees to its side, you could then rotate the horn to maintain the 90 horizontal by 60 vertical pattern. If you want to have 90 degree vertical by 60 degree horizontal coverage with the box standing up, you can also rotate the horn for that. The speaker's shape includes only a slight angle, so it's not enough to make it useful as a "wedge" monitor. You could use it as a front fill in either orientation.

To go a step further, the horizontal off-axis performance is much better with the box oriented vertically. When you rotate the box to "lying down", the horizontal dispersion isn't as even, and rotating the horn does very little to improve the pattern. In short, because out ears are offset horizontally, it's best to stack the highs an mids vertically, so whenever possible, use them standing up...

Neutrik powerCON True 1 TOP IP65 rated waterproof connectors are on all MK3 models.

All BASSBOSS MK3 loudspeakers include Integrated comprehensive DSP. 96kHz sampling rate with 8 pushbutton presets. All presets include high pass filters, low pass filters, phase alignment, equalization and multi-stage limiting. Storage apacity for up to 100 presets (8 at a time).

Any combination of BASSBOSS subs and/or tops are pre-aligned. Different settings are meant to adjust the relative balance of highs to lows, but no adjustment to crossover settings are required regardless of chosen preset.

Peak output is calculated using “industry standard” techniques. These calculation methods create theoretical specifications that are inflated over what can actually be achieved. BASSBOSS real world output specifications are provided as “Maximum Sustained Output” ratings, which reflect actual measured, continuous output levels.

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Sustained power is the rating at which your system can be run indefinitely. As a tuned and well programmed system BASSBOSS speakers are designed to operate at sustained levels for aas long as you need to run your system.

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For more info on why the "Peak" rating makes no sense, check out this video by David Lee:

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All BASSBOSS MK3 loudspeakers have 110/120/220/230v universal power supplies. Pre-MK3 Subs are 120v, pre-MK3 tops are all universal.

Gain Structure and Setting Your Levels

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How to set the levels of Subs and Tops to achieve balanced sound and safe operation in 6 easy steps

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1) Set up your mixer and speakers with all power and signal cables appropriately connected. (OK, this is the hardest step.)  

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2) Power the speakers on and turn their levels to minimum. (All the way down.)

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3) Pick a very loud song or set of songs to play and start the first one on your player. Check the input level on your mixer to ensure it’s not being overdriven. Whenever possible, keep the input signal below clipping, below any distortion.

Boss Tip: Listen for distortion in the tops speakers with their inputs at a very low level. This requires turning up the outputs of the mixer all the way, turning up the speakers until you can barely hear them, then adjusting the input level up until you hear distortion, then down until you hear no distortion. This is informative because if there is any audible distortion with the speakers’ inputs at low levels, the distortion is being created before it’s reaching the speakers. Obviously any distortion indicates something is being driven too hard and that situation will need to be addressed, usually by adjusting the input levels on a preceding device or within a software application.

If you connect more than one mixer, as in a DJ controller and a secondary mixer, it’s critical to follow this same procedure for both and follow the same rule: Keep the inputs from distorting! Overdriving or clipping inputs on secondary mixers is the most common cause of distortion and can cause damage to loudspeakers. It’s a common mistake to run line-level outputs from DJ controllers into mic inputs on secondary mixing boards, with the result being the overdriving (aka clipping) of the mic-level inputs. In short, line-outputs should be connected to line-inputs, and sometimes adapters (XLR to 1/4” TRS) or attenuators (aka pads) or new mixers, are called for. Whatever the case, keep it clean!  

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4) Now that you have verified there is no distortion on the inputs, with one of the test songs playing, turn the output of the mixer* up until the output meters read at the top of the range where you’ll be using it. The specific labeling may vary between mixers. It’s recommended to pick a level at which the color of the indicator LEDs changes and to have the higher level indicator just occasionally flashing. The level indicated by the color change will permit the operator to easily recognize when the limit is reached.

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*In case there is more than one, the “mixer” in this step refers to the device connected directly to the speakers; however it’s necessary to complete all the previous steps on all the devices ahead of the final mixer first.

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5) Turn up the input level of the first speaker until the “limit” LED illuminates. (This will be loud, so step 4A might be to warn the neighbors.) When using BASSBOSS gear exclusively, we recommend starting with the subs. If you don’t have subs, or if you’re using a different brand of subs, starting with a top is appropriate. Repeat with the remaining subs, if any, for one side of your system. Next, if possible, rotate the balance to the opposite side. If you don’t have a balance control, disconnect the signal from the first side. Repeat for the second side.

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With this step completed, the mixer’s output indicators are correlated to the speakers’ output limits. In other words, when the mixer indicates signal above your chosen maximum, the system will be at its uncompressed limits. Pushing the mixer above that maximum will indicate that the system is being pushed above the level at which it begins to compress. That’s the point at which sound quality begins to deteriorate. It may still get somewhat louder but it won’t get better.

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(When using tops without subs, follow the procedure to Step 5.)

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6) If you started with subs, with the sub(s) of one side on at the previously determined level, turn up the top on the same side until the sound is balanced to your liking. (If you started with tops, bring up the first sub to suit your taste.) Repeat this process for the other side. (Yes, your subs should be in stereo.)

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When using tops with one sub or small subs, most often the system will sound balanced when the tops are below the level at which the limit light would come on. This isn’t certain because the relative level will vary according to tastes. The selected preset will also affect the level at which the limit light comes on. The more subs in use, the higher the tops’ level will need to be. In other words, it’s possible to use more than one sub with each top, so to realize the full potential of your tops may require using more than one sub with each. When that’s the case, the procedure is the same: Set the maximum level on the subs and then add level to the top cabinet until you get the balance you want. If the subs and tops reach their limits at the same time, the maximum output capacity of the system is matched to your tastes.

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Following this procedure, the levels should be left where they are. There should be no reason to change them unless a different mixer is used or a different combination of speakers is used.

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There is no reason to turn the levels down before disconnecting power from the speakers. It’s always best to power the speakers off before powering off mixers and other audio equipment in the signal chain. It’s also recommended to connect and power on all other audio equipment before powering on anything with an amplifier.

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~

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Here's a handy PDF reference you can download and save to your device for when you are out in the field and need to set your BASSBOSS gear up:

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Yes, you can daisy chain as many speakers as you want regarding signal, as that is simply a passthrough connection. It's recommended to daisy chain per channel, start with subs, and move to tops at the end. This is primarily for cable management reasons and not because it affects the signal.

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For daisy chaining power —

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Quick answer: Connect subs directly to outlets. Use power passthrough for tops if needed.

Long answer:

Every outlet has a maximum rated current. Every connector has a maximum rated current. Every breaker has a maximum rated current. The most that can be drawn down a single daisy-chain is limited by the item with the lowest rating.  In the USA that’s usually a 15A outlet or plug.

In each owner’s manual, the “average” current draw is listed for each product at 120V or at 240V. Based on the voltage you’re using, (120V is the norm in the USA) add up the ratings for each speaker you intend to connect and stop when you reach the total of the lowest rated connection. (15A using the standard 120V power cords.)  

In general, subwoofers draw closer to their rated current than tops. Tops often draw a very small fraction of their rated current. This fact leads to some practical choices: Always connect subwoofers directly to outlets and daisy-chain to tops last. If power is limited, always account for all the current required for subs first, distributing the subs over as many circuits and outlets as possible. Power for tops can be shared with subs with less concern for reaching the limits of the connectors than if subs are connected together.

For those who argue that amps can’t put out more power than they draw in, here is the expanded scope:  

Each amp has a “rated” maximum power output. This specification is related to the maximum SPL the speaker can achieve but isn’t directly related to the current draw under normal circumstances. What are normal circumstances?

Each amplifier has a “rated” power draw that is lower than the “rated" power output. This is because audio has dynamic range and the average density of music content is about 12%. Thus the average current required to run an amplifier that plays music is about 1/8 the current required to deliver maximum SPL.

And finally: Current draw safety requirements for power connectors and breakers are based on average throughput of current and that average is factored over much longer time durations than musical notes usually last. In other words, the connectors have “headroom”.

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There are so many factors to consider.

Without going deeply into the simulations and graphics I will describe a few related issues and some potential solutions that you could actually use.

Initially let’s consider an outdoor scenario, or at least lets ignore the reflections that occur indoors that would complicate the following discussion.

First, there is Proximity. Having the subs in the proximity of the tops allows their outputs to be aligned and summed in phase. This helps with “punch" and "impact”. If you stack subs in the center, you get the benefit of more even coverage of bass but you lose the benefit of that bass being phase-coherent with the tops over a relatively large coverage area. In big systems, when the tops are flown well above the subs, this alignment issue is a problem even when the subs are left and right because they will also be misaligned to a certain extent from front-to-back due to the height offset. Generally with these bigger systems there is enough equipment and therefore enough sources that the gaps get filled in well enough for the audience to get acceptable results in most locations.  

In smaller systems it can be worse because the tops aren’t able to deliver a sufficient portion of the low frequency energy to make the sides sound full when the subs are stacked at a center location. The center-stacking advantage of having even coverage of the bass, without the lobes associated with a left-right placement, is offset by the potential imbalance in the perceived level of bass. With the subs in the center and the tops left and right, the perception can be too much bass in the middle and too little bass on the sides.

The 'best' solution is to have at least 3 subwoofer sources. One left, one right and one center. Instead of having one power alley, and several valleys, you will create 3 power alleys. One in the center, between the left and right, one between left and center and one between right and center. Having three sources creates a situation where you’re almost never in a location where an absolute null can occur. It puts full-range, high-impact sound on the outer edges so the overall balance is better to the left and right. It puts power alleys where there used to be valleys. It also puts a bit of extra whomp right in front of the stage where the fervent fans tend to gather.

Now let’s consider the difference between indoor and outdoor. Outdoors you’re less likely to have surfaces that reflect sound surrounding your system. There’s the ground, but not much else. And if your subs are on the ground, the reflections are not generally destructive. Indoors, the reflections of low frequency sound contribute to the comb-filtering effect. Comb-filtering refers to destructive interference caused by an offset of arrival times. Reflected sound arrives later than direct sound and it, too, can be constructive or destructive. The difference in distance from the source to the listener versus the source to the reflecting surface and back to the listener will determine the frequencies that are reinforced or cancelled. The side walls, the back wall behind the stage or speakers, the far wall and the ceiling will all reflect sound. For the most part, reflected sound indoors will be perceived as an increase in the level of higher frequencies due to their shorter wavelengths. On the other hand, due to the longer wavelengths of low frequencies, the reflected low-frequency sound can create large areas with a perceived absence of bass. Due to these reflection-induced nulls, subwoofer placement indoors is less forgiving than outdoors.

For instance, if a subwoofer is placed with its back to a wall, there will be a reflection as the sound leaves the front of the box, wraps around the box and then reflects off the wall. If the same subwoofer is placed with its side to the wall, there is effectively no difference in distance between the source and the reflection, making for a smoother response in the room and a slightly higher level to boot.

When you have a null, no amount of level at the source can overcome it. The remaining SPL will always be the same. +1-1=0, +10-10=0, +100-100=0  The only solution is to change the relationship of the direct sound to the reflecting surface. In other words, to move the source,- because that’s usually easier than moving the walls and ceiling. A hard-reflecting, solid surface is effectively a secondary sound source presenting the listener with an inverted, delayed, attenuated signal. The delay is the path length difference from direct to reflected sound. The attenuation is the energy loss over the additional distance and the inversion is the result of the reflection. This problem is potentially worse for people with OCD because if you put everything at the same distances from walls and each other, all the positives and all the negatives add up the same. If you offset things slightly you can have one side null at different frequencies from the other, with the result being some less complete nulls and a more even average level. Hopefully that offers you something to try…

On carts is generally not as good as off carts - unless you’re doing the lifting at the end of the night. On a concrete or dirt floor, off the carts will result in a slight, very slight, difference in SPL and impact. The reason is that if the subs can’t move back when the cones move forward, more of the energy is transmitted into the air. You can strap multiple subs together with ratchet straps to link their total mass to improve performance. This helps whether they are on carts or not.

Other than minimizing heavy lifting, there are circumstances where on a cart is better, such as if you want the carts to isolate the subs from the surface they’re on, like a resonant floor or stage. A resonant floor or stage can make the subs sound muddy by contributing its own tonality to the overall sound. For instance, putting the subs on carts with the wheels unlocked and rotated front-rear will allow the boxes to absorb the cone inertia without transmitting as much energy into the floor.

I would tend to avoid putting subs on any stage that has mics on it unless that stage is poured concrete or packed dirt, and even then I would hesitate. The only reason I can think of for putting subs on a stage, with no mics, is in a room with a ceiling reflection that’s killing your low end energy. This would tend to be a very low, very rigid ceiling, usually less than 12 feet, such as in a basement. If you elevate subs in a room like that, you will probably get better summed response between the floor and ceiling, where the ears are.

This brings to mind the advantages of cardioid deployments. Outdoors, cardioid can keep bass levels down on stage and even keep LF energy from interfering with other performances or even adjacent neighborhoods. Indoors, cardioid deployments can improve frequency response in the audience area by minimizing wrap-around reflections that would otherwise cause destructive interference as they bounced off the back wall and then all around the venue.

In conclusion, I would recommend adding one subwoofer for center placement to a 2 subs and 2 tops setup to help even out bass response in most circumstances. Pay close attention to reflection surfaces and try to use them to your advantage if possible. And consider that a certain amount of asymmetry can work in your favor.

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For basic setups here is a video where David shows some simple examples.

The process of break-in isn't necessary. The recommendation is simply never go from off (cold) to full power. This applies whether the driver is brand new or 10 years old. Allow the heat to build slowly, which is to say for a few minutes, before pushing the levels up. Gradually heating the coil allows it to expand more slowly while allowing the adhesives and other materials, which heat more slowly than the metals, to come up in temperature without cracking.

Check out our video with David Lee for more info!

You can choose to run the signal through the subs to the tops because that's the easiest and most consistent way to get the signal to both sets of boxes. You won't forget to adjust the level of the subs when you adjust the level of the tops.  (seen it happen way too often)  If you mix your instruments with subs on an aux, that's an obvious requirement for aux-fed subs.  If you must have separate control of the subwoofer levels from the mixer, then you can run the subs from the auxes. (Always use two, never sum your subs to mono.) One thing you DON'T want to do if you run subs on aux sends is apply ANY filters to the aux outs. The signal going to the subs needs to be identical to the signal going to the tops, with the only separate adjustment being the level.

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Here is a handy infographic showing a recommended setup for DJs who primarily control their setup through a DJ mixer. Live sound engineers should follow standard gain structure rules through their mixing console.

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The preset configurations shouldn't be pre-planned. They should be tested on site because the relative levels, the environment, which models of subs and tops, the musical genres and personal tastes will all influence which preset combination is preferred. There's no wrong combination, just degrees of better.

Both the BB15 and DJ18 models have installed hardware to accept caster wheels.

You can find compatible casters here.

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When turntables are used in conjunction with powerful subwoofers, significant steps need to be taken to stabilize and isolate the turntables. In the more recent decades, turntables have been largely replaced with digital media. This is because the bass demanded in modern music can’t be recorded into or played back from vinyl records. The more bass there is in the music, and the more bass there is in the system, the more problematic turntables become.

Turntables operate on a fundamentally different principle from magnetic (ie tape) and digital media storage and playback mechanisms. Digital and magnetic media sources are totally isolated from the environment in which they are used. Unfortunately, that’s very definitely not the case when it comes to turntables.

Back to the stone age: A turntable’s pickup needle is a stone, usually a tiny industrial diamond, glued to the end of a tiny rod. On the other end of that rod is a tiny magnet inside a tiny coil. Moving the “needle” moves the rod, and the rod moves the magnet, which then produces a tiny electrical signal from the tiny coil. This tiny signal is amplified by the turntable. It is then amplified again by the phono preamps. It is then amplified again by the mixer. It is then amplified again by the power amplifiers. (When turntables were commonly in use, amplifiers were much less powerful.) There are several points to note from this.

1) The signal coming from the turntable needle is amplified by many orders of magnitude.

2) The signal is produced by physical vibrations.

3) Sound is also physical vibrations, so the output of the subwoofers can be picked up by the needle and re-amplified.

4) When the subwoofers are close to the turntables, there is a greater chance of this happening, which is one reason why I recommended moving the subs to the corner of the room, away from the DJ stand.

Drop the needle on the record: That stone is dropped into the groove of a vinyl record, the sides of which have textures pressed into them such that when the vinyl is scraped over the surface of the diamond, it moves microscopic distances back and forth at exactly the frequencies (notes) and intensities (volume) that are intended to be reproduced. These tiny movements move the magnet at the other end of the rod and those movements are precisely amplified to reproduce the music recorded on the vinyl.

There are important points to consider here:

1) If the needle moves at all, by any cause, that movement will create a signal, and that signal will be picked up and amplified by each subsequent step in the signal chain.

2) If the turntable, the turntable stand, the stage floor or the hand touching the record moves, that needle moves, and a signal is produced.

3) These movements happen at very low frequencies, below the frequencies that humans can hear and below the frequencies at which speakers can safely operate.

4) These movements are inevitably much larger in scale than the tiny movements that turntable needles are intended to encounter, so the effect is that the signal produced by these unintentional movements is far, far bigger and more powerful than anything that could be recorded onto the record(s).

Within the speakers, limiters are implemented to prevent over-powering the woofers under normal circumstances. In addition, frequencies below their normal operating ranges are filtered out. With tops boxes, the filters are high enough to keep the woofers in them out of trouble from these issues. This is one of the reasons it’s possible to determine that the source of the problems with your system are related to out-of-control signals at extremely low frequencies.

Even in the subwoofers, the very lowest frequencies are filtered out of the signal being fed to the woofers, but a balance has to be struck between providing plentiful low-frequency output and preventing every conceivable form of damage. If subwoofers didn’t produce any deep bass, they wouldn’t be subwoofers. (They definitely wouldn’t be Bassboss subwoofers.)

Vented (ported) loudspeakers can produce decent levels of low-frequency energy down to what’s referred to as the resonance frequency of their ports. Down to those frequencies, the resonance of the port does two things. One, it provides acoustical output, and two, it modulates (restricts) the movement of the cone. Below the resonance frequency of the system, the port is no longer restricting the movement of the cone. As the frequencies move well below resonance, the port begins working against the amplifier’s control over the cone. When a powerful signal that is well below resonance is put into a ported system, it’s possible for the woofer to move too far, damaging its suspension and/or its voice coil.

The filters that are used to reduce low-frequency output are best described as having a “slope” insofar as they reduce the level of frequencies below a chosen frequency more and more the farther the signal goes below that chosen frequency. These filter slopes are described in terms of “decibels per octave” with an octave representing the range over which a frequency halves or doubles. (40Hz to 80Hz is an octave. 60Hz to 30Hz is an octave. 500Hz to 1000Hz is an octave, etc.)

To put it in specific terms, the filter frequency in the DJ18 is 30Hz and the slope is 24 decibels per octave. If the subwoofer receives a signal at any frequency below 80Hz and above 30Hz, it passes the signal through to the amplifier unchanged. If it receives a signal at 15Hz, one octave below the filter frequency, it reduces the level by 24 decibels before passing it through to the amplifier. When music program is the source, this is sufficient to protect the woofer because there is no music content at 15Hz and the levels are consistent within the recording.  

However, if a signal comes in at 15Hz that is 24dB higher than the music level signal would be at 30Hz, the two signals would reach the woofer at the same level. If the woofer is driven by a signal at 15Hz anywhere near as hard as it would be at 30Hz and above, even occasionally, damage will result. Similarly, at 22Hz, the signal is lowered by 12dB, so a signal at 22Hz that was 12dB louder than the normal signal from a record or digital file would push the driver beyond its limits.  

We do our best to protect the speakers from every circumstance that may be anticipated to put them in danger, even well beyond “normal”, but excessive movement and infrasonic feedback caused by unstable turntables is beyond the scope of the protection required by the vast majority of customers. This protection can be implemented but it’s necessary for the customer to understand the dangers and limitations associated with implementing old technologies with new and powerful systems. When turntables were the primary source of music in clubs, bass was weak and DJs had “booths” to isolate the turntables from the speakers on the dance floor.

What can be done to protect high-powered, deep-bass subwoofers from turntables? The first, best, easiest and cheapest answer is to keep turntables out of the building.

If you do plan to use turntables regularly, it’s best to go old-school and build an absolutely, solid immovable platform for the turntables. This is best done starting from a concrete floor and building the platform from concrete and/or cinderblocks and concrete. When the turntable stand is concrete and anchored to the floor with rebar or all-thread so that you can stomp-kick the turntable stand without skipping a playing record, you’ve got a good turntable stand. No part of the stage, railings or anything at all can be allowed to touch the turntable stand. In addition, some of the deepest bass would still need to be filtered out.

If you must use turntables, and you won’t be building the above turntable stand, you have to give up on the deep bass. This isn’t such a big compromise as it sounds because records can’t store or reproduce deep bass. Essentially, if you’re using records, you’re not getting deep bass anyway, and you definitely can’t access any of what they do have without the aforementioned turntable stand.

If you want to keep deep-bass capabilities available in your system when turntables are not in use, the deep bass must be filtered out when turntables are in use. This requires additional equipment between the DJ mixer and the speakers, into which the appropriate filters can be programmed. A “Turntables” setting would need to be loaded any time turntables were going to be in use.

To further reduce the possibility of problems, the subwoofers must be positioned and/or programmed so that a bass-null is created in the area of the turntable stand. Relocating the subwoofers is beneficial in achieving this.

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If you see a light glowing inside your subwoofer cabinets, don’t panic! High-powered lamps are provided as a safety mechanism to keep the drivers alive, even if the system is being pushed really hard.

Heavy-duty, vibration-resistant, 500W halogen lamps help to keep the voice coils safe. Excess electrical energy that could overheat the voice coils is absorbed and converted to heat and light by the lamps before that heat can get into the coil and blow the driver. Most of the time the lamps aren’t even getting warm , so they’re barely on and never noticed. But, all lamps do burn out eventually, even if they are barely ever on.

As each lamp fails, the remainder will glow more brightly. The good news is that the more of the lamps that fail, the safer your woofers are. And these lamps are a lot less expensive than woofers!

To ensure the protection circuits were easy to service, we used readily-available,118mm, R7S 500W 120V halogen lamps, which you can buy from most places that sell lamps or work lights (Home Depot, Lowe’s, Harbor Freight, WalMart, Amazon, etc.) New lamps should last many more years, but give them a check every two years or so because some lamps last longer than others and some systems are pushed harder than others.

Here’s what you’ll need and how to replace the lamps:

Buy lightbulbs on Amazon.

Tools you’ll need:

#2 Phillips or JIS screwdriver to remove the grilles (Not needed for VS21)

Allen Wrench to remove the driver(s) or cover panel (Be sure to use the right size to avoid damaging the bolt heads)

MkI woofers use 5/32” drive for flat-head or 3/16” for socket-head bolts. MK1 VS21 covers use 7/32” drive

MK2 woofers and VS21 cover panels both use 5mm drive

Wire brush, or similar, to scrub any corrosion or carbonization from the lamp holders

Recommended: Grease, such as marine grease or dielectric grease, to prevent corrosion and carbonization

The lamps are located behind the drivers, so it’s easiest to access them by removing the drivers. Here are the basic steps to make the procedure as easy as possible:

1. Put the cabinet on its back
2. Remove the grille-screws and remove the grille (VS21: Remove the cover panel and skip to step 4)
3. Unbolt the driver and lift it out.
4. Locate the lamp sockets. Most cabinets have them attached to the inner port walls.
5. Carefully remove all the old lamps. The ends of the lamps can crack. Glass or ceramic shards may be present.
6. Clean the contact tabs on both ends of the lamp sockets with a wire brush.
7. Once they are clear of corrosion, etc., apply grease to the contact tabs. Also apply grease to both ends of the lamps. (You don’t have to worry about touching the glass because they are never as hot as actual lights.)
8. Insert the lamps into the sockets, one end first. The lamps are connected and held in place by the spring tension between the socket and the tube. The lamps should rotate smoothly when properly inserted. To check this, rotate the lamps, ie spin them on their contact axis between thumb and forefinger.
9. To test the contacts, flip the cabinet onto its feet and connect power and signal to the amp.
10. Short the red and black wires of one woofer together and play some bass into the sub’s amp. If you are performing this procedure on a double-driver sub, test each side separately.
11. All the lamps feeding those shorted wires should glow orange. (This will make the lamps HOT to the touch!)
12. If any don’t glow, note which ones and, when the rest cool, check that they are centered on the fixture and making contact. It may be necessary to remove more of the carbon build-up to get good contact.
13. Repeat the signal test to confirm all lamps glow when their woofer wires are shorted.
14. Disconnect the signal and power from the amp and flip the cabinet on its back again.
    

While you have the drivers out, examine them for wear or damage. It’s assumed that the woofers are working, but these examinations will give you some insight into their condition and life-expectancy.

• Check the cones, particularly where they join with the surrounds, to ensure they have not torn or separated
• Check the spiders, particularly where they attach to the coil-former, to ensure they have not torn or separated
• Check tinsel leads, particularly where they meet the cone, to ensure there is no evidence of burning or fatigue.
  

If you observe any of the conditions listed above, you may choose to take this opportunity to replace badly-worn woofers before they fail in the field. If there are no signs of impending failure, it’s time to put the woofers back in.

15. Connect the wires to the woofers, red wires to red terminals, black wires to black terminals, of course!
16. Set the woofers in place and re-insert the bolts, starting each bolt by hand to ensure the bolts don’t cross-thread.
17. Tighten the bolts in a star-pattern, similar to that used on wheel lug-nuts. Don’t over-tighten. 4-5 foot-pounds is sufficient for T-nuts.
18. Check the gasket tape and re-install the grille. (On VS21, check the gasket tape and re-install the front cover panel.)
    

If you need to replace gasket tape, it can be found here: Buy Gasket Tape‍

If you need to replace a non-working lamp socket, they can be found here: Buy Lamp Sockets Option #1 | Buy Lamp Sockets Option#2

Download PDF of information and process here:

We recommend the following poles for your BASSBOSS System. These are available at many online retailers.

K&M makes the most reliable and professional poles we've found! They offer a 5-year warranty with a 10-year guarantee of availability for most spare parts.

Check them out at Koenig & Meyer

Adjustable ring-lock poles for between smaller tops and subs: K/M Ring Lock Distance Rod 38"- 58" 21366-014-55 & 21467 Speaker stand »Ring Lock«

Fixed length distance rod: K/M Ring Lock Distance Rod 30" fixed 21360

BASSBOSS is currently developing a full line of custom covers. In the interim, Under Cover is able to make covers for the entire line.

Under Cover products are made in the USA and they offer very high-quality covers.

MK3 Speakers:

Signal: XLR in and Passthrough
Control: etherCON in and through (For daisy chaining)
Power*: powerCON to 3 prong Edison in and Passthrough

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Pre-MK3 Speakers:

Signal: XLR in and Passthrough
Power*: powerCON to 3 prong Edison (Some with Passthrough)

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*Passthrough power should not exceed circuit connected to. 2500W amps should only connect 2 per 20AMP Circuit, 4 per 40AMP, etc. Any improper load balancing could lead to blown circuits.

Need replacement screws for your BASSBOSS Gear?

Replacement screws for grilles... You can buy a lifetime supply for just ten bucks on Amazon. 😉

We recommend these: 100PCS #6 x3/4" Truss Head Self Tapping Screws Phillips Drive, Sheet Metal Wood Screws, Black Oxide 304 Stainless Steel 18-8

Replacement screws for amplifiers

We recommend these: M5 x 20mm Button Head Socket Cap Screws Bolts, 10.9 Grade Alloy Steel, Black Oxide Finish, Metric Allen Hex Drive, Fully Threaded, 100PCS

Replacement screws for drivers –

We recommend these:

Please contact the dealer you purchased from for all order questions - BASSBOSS does not have access to inventory or order information.

Please check our Support Page for product registration. Thanks for joining the BASSBOSS Family!

Since each support and service case is unique, some issues can be resolved in as little as one business day, while others will take longer. Service cases requiring parts could take 4 to 7 business days in transit.

We make every effort to turn around service issues as quickly as possible.

If you have a smaller BASSBOSS subwoofer, it may walk away from you during a gig from the intensity of the bass output. This is especially a problem with slick floors. Any rubber mat or carpet square will keep the sub in its place, and can be cut to the correct size to be invisible under the sub.

We try to make products that never fail. That said, nothing in this world is perfect! If you need information on self-servicing your BASSBOSS product, we have a whole series of videos with David Lee to help.

Please check out our Service Videos page for more information.

Got questions about using your BASSBOSS gear to the best effect? Check out our extensive EDU page for videos on how to get the most out of your system.

There are always what one might call “practical limitations” to speakers. The DV12 and the SV9 are not subwoofers. They can produce a fair amount of bass, but the woofers also plays midrange, which all woofers do more easily and efficiently than they make bass. To put that another way, for the same amount of power input, any woofer will make much more midrange than bass. Most people listening to dance music want more bass than midrange, which is what necessitates subwoofers. Subwoofers are essentially additional boxes dedicated to increasing the level of the bass because people want more bass.

To give an idea of how much the demand for bass is above midrange, and the relative ease with which midrange can be made, the ratio can be as high as 4:1 of subwoofers to tops. Trying to get a level of bass that’s close to what’s desired in modern music from a single 12” or single 9” top cabinet is going to push that top’s driver past its limits. Two high-powered 12” tops could be used with 6 or 8 single 18” subwoofers and the tops would easily keep up. They will definitely “outrun” a single 15” sub. If the DV12 or SV9 cabinet is being run in full-range mode with heavy bass music, it may be that it’s giving all it can and that’s the limit. But let’s consider other possibilities.

One situation that we’ve observed has been with speaker poles. If K/M Ring Lock Distance Rod speaker poles are used, this doesn’t happen, but if standard speaker poles are used, the speakers may “bounce” on the poles, causing a rattling that can sound like the woofer popping. To eliminate this as a possibility, test them when not on poles.

Test the cables. Bad cables have been found to be the cause of issues that were initially blamed on the speakers. Intermittently bad cables can cause popping noises. The making and breaking of signal-level contacts makes a sound like the speaker is popping.

Signal level clipping could be happening. To evaluate whether this is the case, eliminate all electronics between the controller (or whatever the source of the music is, such as CDJ and DJM setup,) and the speakers. Turn the “Master” level all the way down, leave the EQs at flat, and turn the input level on the controller up all the way, both the trim and the channel fader. Connect the speakers directly to the master outputs of the controller and put them in preset 1. Turn the signal input all the way up on the speaker.

Next, just barely move the output level above "off" and listen to the sound coming out of the speaker. Is it crystal clear and perfect? If it’s not perfect, lower the input trim until it sounds perfect. Then, raise the output higher until the speaker shows the -3dB LED. All good or anything popping? If no popping, raise the level more until the limit LED is coming on. Any popping? If there’s any popping under these circumstances, there may be issues with an internal contact, the woofer or the DSP, in which case let us know and we’ll get that taken care of. If not, the issue might be elsewhere.

When going into any mixer from a controller, a line-input must be used. Running a controller into a mic input can cause that input to clip, causing distortion that might cause popping noises. If popping only happens when a secondary device is in use, look for the cause there.

Doing any signal processing in any other device could cause internal clipping or could cause the signal reaching the speakers to be so bass-boosted that it’s unrealistic to expect the speaker to reproduce it. Test with all the filters in the mixer bypassed. If the popping doesn’t happen when the source is connected directly to the speakers but does happen when other equipment is connected, then the other equipment is likely causing the issue.

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David Lee is BASSBOSS' founder and designer. Learn more about David in this video:

BASSBOSS' goal is to provide people with a fantastic musical experience and to facilitate performers in delivering their music in the most impactful way possible. Learn more about the why of BASSBOSS directly from our founder, David Lee, in this video:

Learn about BASSBOSS speakers' power requirements by watching this video:

David Lee discusses what BASSBOSS speakers you can use on a 20 amp circuit in this video:

David Lee talks about the benefits of deep bass in this two part interview:

Part 01:

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Part 2:

David Lee shares how to best distribute your system load to get the most out of the available power in this video:

David Lee discusses the importance of input sensitivity and shows you how to use it to achieve the best sound quality from your loudspeakers in this video:

AM&S National Sales Manager, Samia Scoda, discusses the importance of gain structure in professional audio systems in this video:

David Lee talks about mastering headroom and how to avoid  distorting and clipping in this video:

David Lee explains the advantages and disadvantages of vertical and horizontal speaker placements in this video:

Q: I am considering bringing a smaller system to burning man. I’m thinking I will bring my EAW tops and Two of the VS21’s as to keep the main bulk of the equipment in perfect condition. As this will be our first year as a sound camp I am worried about power consumption needs and would like to know some information if you can help me out here. I have a couple of these generators (https://www.harborfreight.com/3500-watt-super-quiet-inverter-generator-63584.html) and I would rather bring them then rent a large one from someone.

My question is how many of the VS21s could be run on one of these generators at a loud level but not maxed out. Just 1, maybe 2? I have quite a lot more stuff to power there and was rather confused by the 360watts at typical 1/8th power. I’m assuming here that I would be running them a lot harder than that being a dance music focused camp. I can run both of them in parallel as they are matching.

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I have never personally run our gear on one of those Harbor Freight generators but I have used the Honda EU7000 for a pretty big system. I can tell you what we ran on the Honda, where the limits were and why the ratings are what they are.

First, the 360 Watts at 1/8th power is based upon the average consumption over a period of time, such as 10 minutes or an hour, even when the system is turned up loud. The peaks will be much higher but the valleys still occur and the average, when all things are factored, is still quite low. These figures are appropriate to calculate power consumption under normal use circumstances and assume the power is being pulled from “the grid”. The reason that assumption is relevant is “the grid” doesn’t have an absolute current draw maximum. The maximum is set by the breaker feeding the outlet into which the amplifier is plugged. Breakers will pass more than their rated current for short durations without tripping and a brief peak draw from grid power will not cause a significant drop in voltage.  

When running amplifiers on a generator, the generator’s maximum capacity is the absolute limit because, unlike the grid, it has no excess capacity… no headroom. As soon as the current demand reaches the limit of the generator, the voltage drops, hard! Everything draws more current to do the same work on lower voltage, so it’s a death spiral. Breakers trip or equipment browns out or the gennie shuts down.  

Normally you can expect to be able to run 2 VS21s on a grid-powered 120V, 20A circuit as hard as you like and not trip the breaker. You should be able to run one VS21 on a 3500W generator if it lives up to it’s ratings. That’s 2400W with 1000W of “headroom” for peaks so as to limit voltage drops. You can run 2 VS21s on the same generator but at a reduced level. The question to answer is whether 2 boxes at a reduced level will provide more overall SPL than one box at full power. In theory, yes, they will. The problem might be called “temptation” or “Fader Creep” because there is a lot of temptation to push a little harder, so you might be able to get more sound from two boxes that could overload the generator - if you don't overload the generator - but you’re less likely to find yourself sitting in a temporary silence if you run just one.  

So, you can run two VS21s on two of those generators IF you don’t eat up your headroom with too may other electricity consumers, not least of which will be amps for the EAWs. You may not be able to get every last decibel out of them but they should give you plenty of whomp!  

From personal experience, we ran quite a bit on the EU7000, which is theoretically two of the Harbor Freight gennies. We had 3 x 4000 W amps powering 4 x double 18” subs and 1 x double 21” sub and two QSC amps running 6 line arrays and more amps running 2 monitors and 2 fill speakers along with a DJ controller and a DSP unit. When the system got cranked up and the gennie hit its limit, the DJ controller and the DSP shut down due to voltage drop. Even so, before over-exuberance pushed it over the edge the rig was pretty impressively large and loud for running on one EU7000!  

And as much as I would love to hear that system on the Playa, I totally understand and agree with the plan to keep as much stuff new and pretty for as long as possible! Besides, those two Harbor Freight generators couldn’t possibly run the rig you’re getting to any level that would do it justice!  

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David Lee dives into the best type of generator to run your audio system in this video:

David Lee discusses the advantages and disadvantages of using carts for subwoofer deployment in this video:

David Lee from BASSBOSS discusses the optimal ratio of subwoofers (subs) to top speakers (tops) in this video:

David Lee explains how to avoid ground loop noise in this video:

David Lee demonstrates how to move and stack ZV28 subwoofers in this video:

David Lee discusses how best to move and stack VS21 subwoofers in this video:

David Lee talks about what cleaning products to use on your system in this

David Lee discusses the advantages and disadvantages between 2-way and 3-way loudspeakers in this video:

Heat is always the enemy. The amps in the subs don't tend to get very hot but if the amp plate is left in direct sun in the summer it would double the problem. Keeping the amps shaded would be a good idea. Only 2 to 8% of the amp's output is converted to heat, so its cooling requirements are low. If it does get hot enough it will reduce output to prevent overheating. If the heat continues to rise after the level reduction, it will shut down. Fan cooling can be added if that becomes a problem. The air movement generated through the ports helps to cool the amp so I don't expect running it harder will cause it to overheat more. So far we haven't been able to get the amps above barely warm.

Ambient heat will affect how well the drivers cool but there's not much to be done about it, other than turning the system down, that will reduce the operating temperature. Amps run 90 to 98% efficient, meaning 2 to 10 percent of the energy is waste heat. Woofers run 1 to 4% efficient, meaning 96 to 99% of the energy going into them is converted to heat, the remainder into sound. Heat is the enemy, more so in the voice coils than in the amps. We have to limit the inbound power to prevent excess heat buildup but ambient temperature will change how long they can be run at maximum output. If the limiters are set for high-temperature conditions, then max levels in cooler temps would be compromised. Normally we expect things to be pretty hot so even in the desert heat, you should be OK, just remember that you do have control over the level.

Thinking of the volume knob as a 4 percent sound a 96 percent heat knob would be informative.

‍Q: What is the best way to align the VS21s for summing?

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The summing of the outputs of 2 cabinets offers the same benefits directly on-axis regardless of their orientation. It's off-axis that there are some differences. In a VS21, the lowest frequencies come from the ports and the upper frequencies come from the mouth, with the curved grille. Destructive interference starts to occur when the sources are separated by more than 1/3 of the wavelength being produced.  For this reason, the best coverage is achieved when the horn sections are together. The lower frequencies coming from the ports are still within the minimum distance at just under 6' apart. That said, putting them the opposite way, with horns out, won't cause any problems and, although the coverage above 60Hz might be a little less even, you might like the sound that way better, so don't worry, you can't get it wrong.

There are professional studios and producers using the SV9s for monitors and the reports we've heard have been very positive. We've had many phone conversations with folks that indicated they were very happy with their MK2 SV9s in their studios.

The MK3s are even better. They have a lower noise floor and higher output.  The lower noise floor is the biggest benefit for studio use and higher output is not usually a bad thing anywhere...

The SV9 MK3s are very good speakers.  It's difficult to understand what they will be able to do for you until you have them, but once you do, you'll know.

The intention of the SV9 MK3 is to make your creative work easier and more enjoyable, and thereby better.  Adding more steps, wore work and more processing doesn't serve that purpose and we do our best to provide products that need nothing more than power and signal.

The SV9s will make your efforts in the studio easier by providing you with high-resolution referencing, and it will make the sharing of your efforts easier by translating it to larger audiences and spaces without changing the fundamental character of what you've done in the studio.  In other words, the engineering behind these systems is intended to make it possible for you to play music rather than work in the studio or work on a show.

The dimensions of the radiating area of your subwoofer(s) determine the frequency range of their directivity. It's often said that bass is omnidirectional, however that's only true to for wavelengths longer than the height and/or width of your subwoofer or subwoofer array.  The longer the dimension, the lower the frequency at which the array can achieve some directivity.  This isn't a huge factor with just 2 single 18" subwoofers but the effects get more noticeable as subwoofer arrays get bigger...

Now, that said, the ground is effectively a mirror to bass. This is called half-space radiation. 2 side-by-side subs are mirrored by the ground to provide the effective output of a block of 4 subs in full space.  The side-by-side subs present a larger dimension to the ground/mirror and the effect is slightly lower frequencies will be constrained and reflected by the ground.

The short answer is, side-by-side will provide slightly better performance at slightly lower frequencies.  Stacking vertically will provide slightly better performance at slightly higher frequencies. The corollary to that is that side-by-side will provide slightly narrower horizontal coverage and stacking vertically will provide slightly wider horizontal coverage.

If you have 4, stacking them in a cube will give you the best overall result. Putting them all side-by-side will start to narrow the sound field slightly, and won't seem as loud up close but it is likely that it will provide response that is _slightly_ deeper on axis and farther away.

In open space, two boxes in normal mode will provide more output up front than if one of them were in cardioid BUT if a reflection from a wall behind the subs was to cause a reflection that cancelled the primary output of the two subs, then placing one of the cabinets in cardioid mode to mitigate that reflection could result in higher SPL in front of the two boxes. In other words, you can use cardioid mode with just 2 boxes to good effect when it's called for.

Cardiod mode uses the output of one of the boxes (which is turned to face backwards) to deliberately "phase out" the sound going backwards behind the stack. If the sound going backwards hits a wall and then comes forwards again, that sound can "phase out" the forward-traveling sound. If you cancel the backwards-traveling sound before it hits the wall, it won't have the energy to do any cancelling out front.

Using a subharmonic synthesizer needs to be done with care and in appropriate circumstances. The Bassboss subs reproduce the music you put into them. The subs don't benefit from the enhancements, the music is what's enhanced, and it's the audience that can benefit. The subs just do their job...   But there is a limit...

The key is, as you say, to only enhance SOME recordings. Often older releases can benefit from adding some of the deep low-end that's more common in recent releases to make them sound more current and less dated... BUT...  adding subharmonic processing to tracks that already have deep bass, and lots of it, is asking for trouble.

Bassboss subs do very well at reproducing the low frequencies in even the most bass-heavy releases, and therefor do very well reproducing the enhanced low-frequency content of older recordings, but when you combine the subharmonic synthesis with music that's been produced with that kind of deep bass already, you start to ask your subs to reproduce frequencies that go well below the capabilities of almost all subwoofers.

If you run 60Hz through a subharmonic synthesizer, it will produce demand for 30Hz. 40Hz will demand 20Hz and 30Hz will create demand for output at 15Hz! The reason why older recordings don't contain as much LF content as current recordings do is because the technology to produce, record, and ultimately reproduce those notes didn't exist, at least not in any practical form.

The reason why current recordings contain a lot more bass information is because the technologies to record and play them back are now a reality thanks to digital recording and playback equipment. The loudspeakers to reproduce deep bass are also now available, although not yet common. Tracks that contain a lot of content at or below 20Hz are still very rare for several reasons, starting with because the frequencies can't be reproduced effectively for reference in most studios. Those notes also cause the majority of systems that would be used to reproduce them to distort, or fail, making the songs sound bad, so the producers filter them out.

If you then add them back in, you'll likely cause the distortions and failures that filtering them out was intended to prevent. Unless or until you have a loudspeaker system that's specified to reproduce frequencies below 20Hz,  I'd recommend using any subharmonic synthesizer only to enhance bass-lacking recordings or live instruments, and be very careful how much you use. You might think of it like adding salt in a recipe. A dish without salt can be bland, so adding some will improve it, but adding a lot of salt to a salty dish will definitely ruin it.

Get it right and you'll be cooking with bass! Get it wrong and you'll cook the bass...

In short, you don't need or want to add an EQ. The system components offer you enough controls and options to adapt to various environments without additional processing. In addition, all but the very best (read most expensive) EQs will deteriorate the signal quality and graphic EQs are an outdated design based on fixed frequency points that are never centered where your issues are or where natural harmonics occur.

You're better off knowing where to place speakers to minimize the reflections that are most often the cause of "room issues" for which EQ doesn't really help. You can't EQ out a reflection because most of the issues they cause are related to arrival time, not signal amplitude. If, after you've used the system for a few months, you decide you must have an EQ, you should get a parametric EQ. You'll do more good and far less damage with a 4-band parametric EQ than a 31 band graphic EQ. They also offer a better opportunity to learn what does what and how to get better results.

There's a guy I know who loves to take his iPad analyzer out and show it off, make pronouncements and "tune" systems based on what he sees. I know of another guy who goes into clubs with an RTA and "tunes" systems with it. What they both don't realize is that what they see, regardless of the quality of the mic in use, is only relevant in that exact location and will not tell the same story as little as 6 inches from that spot.  What they also don't know is that they are measuring the sum total of all the acoustical energy bouncing around in the space with no reference to time so there's no way to tell the difference between the primary and reflected energy. The one guy has good ears and the other one is a good salesman. Neither one is helping as much as they could if they knew how to use a transfer function FFT system. And to be frank, the guy with the good ears does more good using them than his RTA and thankfully uses it more to show off than for data acquisition. The other guy uses the RTA and, from what I've seen, does mostly damage.

The good news is that the BASSBOSS speaker system you've bought will need nothing in order to deliver outstanding sound quality under most conditions. The bad news is there isn't much more that can be done to improve its performance in the field. To put it another way, you've just bought a Ferrari. You won't need to be re-mapping the fuel injection. ;-)

In order to prevent access to the amplifier's controls and connectors, there are smaller, simpler and cheaper options than building a box. The simplest, from a customer's perspective, would be to cut a piece of plywood about the size of the back of the cabinet and drill some holes in it. Cut a notch near the middle of the board on each side then drill 4 holes that line up with the cabinet feet. Remove the foot bolts and, leaving the rubber feet in place, add the board using longer M8 bolts through the feet.

The notches will allow the cables to exit at the bottom of the board and the spacing provided by the feet will allow the cables to access the back of the cabinet and allow enough air flow for cooling. (Most of the cooling happens inside the cabinet thanks to the air-flow through the ports.)

Example below is for the ZV28:

Q: Do you have any advice on subwoofer stacking or placement when using two ZV28s? Should they be vertical side by side or stacked horizontal? Or another config?

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There’s not a single best way to position subs because different setups yield different results.

It helps to think of the ground as a sound mirror. The larger the surface you put against the mirror, the more sound gets reflected. If you put a cabinet horizontally on the ground, less sound escapes around the box and more is reflected. Due to the relationship between cabinet dimensions and wavelengths, the longer the surface adjacent to the ground, the lower the frequencies are that get reflected. Two boxes horizontally side-by-side will deliver more of the lower frequencies forward than if the boxes were stacked or standing up. The sound will also be focused forward up the center between the boxes with less energy available to the left and right. If you stack two boxes, the surface adjacent to the ground is half as long, so the lowest frequency the reflection supports is higher, (going from about 40Hz to about 80Hz at 1/4 wavelength directivity.) Since the radiating area is about square at that point, 80Hz is similarly supported both vertically and horizontally. This results in a more “punchy” sound.

Standing the boxes side-by-side has the same 80Hz directivity but the difference is in how the drivers are loaded. When the boxes are stacked, both drivers in each box are equally loaded. The bottom box is loaded differently from the top box because the bottom box is radiating into less space. The ground below doesn’t yield to pressure and the box above provides pressure against which the bottom box must also push. The top box has pressure coming from the bottom box, but that’s not as unyielding as the ground, and above it there is no additional resistance to pressure. The differences are slight but the effect is that each box behaves in a relatively stable fashion, with both drivers presenting the same resistance to the amplifier’s output voltage. When the boxes are standing, the lower driver is in the higher pressure environment and the upper driver is in the lower pressure environment. Each driver presents a slightly different load to the amplifier. This can allow the upper and lower drivers to behave differently, resulting in a different frequency response from when the drivers are equally loaded, and the observed result is that the sound is perceived as being even more “punchy” although this is a trade-off for reduced output at the very lowest range of the boxes' response.

The short answer: Punchiest is side-by-side standing up. Deepest is side-by-side lying down. Stacked is splitting the difference.

And, if the ground is not the only reflecting surface, you may need to change your priorities and adjust placement to compensate for those reflections.The reflections from walls and the ceiling can do a lot more harm to the system response in the listening area than which way the boxes are stacked.

Q: Being that the MFLA is rated to sustain 136dB, would invert-arraying a pair of DV12’s get me to 136dB from 126dB as a non arrayed DV12? Would arraying 2 MFLA cabinets together add 10dB and would adding 2 more MFLAs effectively double the volume once again?

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Arraying a pair of speakers ordinarily results in an increase of 6dB if their output is coherent, which an array of DV12s, an array of MFLAs and an array of most subs would be. An array of 2 "point-source" speakers (such as AT212s, DiaMons, AT312s and most other non-line-array speakers in the world) would result in a 6dB increase at low frequencies, where the output would be coherent, and an increase of only 3dB where their output would tend to be incoherent, in other words, where their outputs would begin interfering with each other. (aka comb-filtering.)

The woofers in the MFLA are more efficient than the woofers in the DV12, so they can sustain a higher SPL. Two DV12s would get you to 132dB. (The low-frequency drivers are usually the limiting factor in the output SPL, which is why some manufacturers claim such high SPL capabilities for their line-arrays and tops: They measure the maximum SPL that the tweeter can provide, or at 1kHz, where both woofers and tweeters are contributing, without considering that if the woofer(s) can't keep up at the bottom end, the sound character changes and the result doesn't sound good at all.)

The biggest risk in adding another device in your signal chain is signal level clipping. I'll be brief and list the points to watch out for.

Do not run line-level signals into microphone-level inputs. A DJ mixer can and will clip a microphone input on a mixing board.

Do not use the secondary mixer to increase the signal gain going into your speakers. The maximum output of your DJ controller/mixer is enough to get the speakers to their maximum output capacity. If the system isn't loud enough with the DJ controller turned all the way up, then you need more system, not more signal gain.

Don't use your eyes to gauge the sound levels. Your DJ mixer's outputs are VERY unlikely to clip. It's much more likely to clip their inputs, and even more likely to clip the inputs of another mixer to which the DJ mixer is connected.  In other words, turn up your DJ mixer until the speakers are as loud as you want them to be, regardless of the indicated level on the DJ mixer. Reducing the indicated level on the DJ mixer by using a secondary device to add more gain is simply providing yourself with a visual illusion.  All professional-grade signal-level devices produce effectively the same maximum output level: Nominal +4 dBU, maximum +22 to +24dBU, equating to approximately 12VAC from the signal outputs. More than that would clip the inputs of any line-level device. (Considerably less than that is enough to clip a mic input.)

Amp-level clipping can be prevented in the internal processing, so is no longer a cause of driver failures. Signal-level clipping can't be prevented in the amplifier, and so the amplifier does what amplifiers do and it amplifies the clipped signal. This can do the same damage to drivers as amp-level clipping, but the limiters can't prevent it.

The best way to add mics to a DJ system is to mix them before going into the DJ mixer and add them to one of the inputs on the DJ board. Some mic mixers even have mic-level outputs, which can be connected to the DJ mixer's mic inputs.

In conclusion, if you choose to run an additional mixer between your DJ mixer and your speakers, take very good care not to clip the inputs to that mixer. Be aware that the speakers have a maximum output level that is unchangeable regardless of how much signal is sent to them. Further, keep in mind that the speakers have limiters and that the sound quality deteriorates when the limiters are active. It's best to have the indicators on your mixer tell you whether the system is close to maximum output rather than hiding that information by adding gain at another stage.

For the most part, it is normal. In louder environments, it's not noticeable. At a party or event, it won't be noticed. When high-gain electronics are connected to high-sensitivity loudspeakers, sometimes the lowest level of the amplifier's output, it's "idle level" is audible because the speaker is very sensitive. The reason it's audible from the horn and not from the woofer is because the horn is much more sensitive than the woofer.  The same amount of millivolts are going to both drivers but the compression driver and horn are much more efficient at converting those millivolts into sound pressure.

There is a maximum dynamic range between the lowest level an electronic circuit produces and the highest level it can produce. For pro-audio gear this is usually in the range of 110-115dB. The output of one of those horns is ~115dB with 1W @1 meter. The amplifier channel can produce 1500 watts. Essentially the horn can produce 62dB from only 0.01 millivolt. We make compromises in order to get loud, and that compromise could be the inability to get perfectly quiet.

It may be possible to make it a little less noticeable in quiet spaces, and this would also improve sound quality anywhere you would take them... (This is what everyone should be doing anyway, but it's particularly relevant here...) Try turning down the input level on the cabinet. First, turn it all the way down and see if the noise is reduced. If it is, then we're making progress. Next, put some music through your mixer and bring the mixer's output level up until it's at 98% of maximum. Yes, really high. Now, bring the level on your speakers up just enough to reach the maximum sound level you'll want in the room. With that done, you'll no longer be amplifying background noise for no reason. You won't be amplifying background noise from any source, amp, mixer or whatever. Your mixer won't struggle to produce enough output to play the speakers as loud as you want. Mixers have way more output than you need and they tend to sound better when they are delivering more output rather than less.

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Q: I need help understanding what power cords are best!  Especially when traveling long distances. I have been using crappy power extensions not considering proper power. Thinking about getting 10 gauge extensions. Is that necessary?  Will 12 gauge work fine too?

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Things do tend to run hotter when voltage drops. When voltage drops, the demand for current rises. There are standards for cable gauge vs distance, and the further you have to go, the thicker the cable should be. The thicker the cable, the less voltage drop there will be at the end, so your amps will have access to more power and, all other things being equal, the cooler they will run.  

If you were running all (say) 6 of your boxes on one extension cable, that may have been the cause of your issue. Essentially the more copper you have carrying the electricity, the better.  It's not as simple as 12 gauge vs 10 gauge because the current demand and the length have to be considered.

Voltage drop calculation

Ohm's Law is a very basic law for calculating voltage drop:

V_drop = I·R

where:

I: the current through the wire, measured in amperes

R: the resistance of the wires, measured in ohms

The resistance of the wires is often measured and given as length-specific resistance, normally in the unit of ohms per kilometer or ohms per 1000 feet. Also, the wire is round-tripped. Therefore, the formula for a single-phase or direct current circuit becomes:

V_drop = 2·I·R·L

where:

I: the current through the wire

R: the length-specific resistance of the wires

L: the one-way length

But, to shortcut the math, you can look at a chart.

To provide 20A at 100' you'll need 8 gauge wire. That's some heavy cable, so the next best thing is to run multiple cables of a lighter gauge and connect less load to each of them.  With four subs combined could draw peaks of 40A, so for long runs you should run 4 extension cables of the appropriate gauge for the length. The tops don't draw much but they will suffer the loss caused by the consumption of the subwoofers. Each of the tops can share a circuit with a sub for instance.

All BASSBOSS subs already have all the processing built in, so they don't need, and should not have, any additional processing added to the signal chain. If you need an external processor to control your non BASSBOSS tops, you could run the signal through a processor BUT without any filters on the outputs to the BASSBOSS sub. That way you could control their level without compromising their response and use the outputs for the tops to do what you need to do to them. (I'd recommend spending the cost of the external processor on a better loudspeaker, though.) Next question is, what do you need to do? And why do you need to do it?  

The BASSBOSS tops all have the necessary high-pass filters included, and are always coherent with any of our subwoofers when used together.  When combining different brands of subs and tops, inevitably the challenge is to get the subs and tops coherently in-phase. (That's often also a challenge when combining the same brand of subs and tops, but that's a different discussion...)  

Bear in mind that just because your filter frequencies match (Eg: 100Hz High-pass and 100Hz Low-pass,) that doesn't mean that the speakers are going to be coherently in-phase. The filter frequencies are only one of a large number of variables that have to be taken into account. A partial list includes: The filter slope, (18, 24, 36, 48dB/octave,)  the filter form, (Butterworth, Linkwitz-Riley, etc.,) Equalization filter delay, (not inserted time delay but the filter delay, essentially phase shift that is the result of EQ filters,) the influence of ancillary filters, (such as the subwoofer's high-pass filter,) the processor latency (which is different from one DSP to another and from one sampling rate to another,) and, often forgotten, the inertia of the cone and resonant system as a whole.

Combining speakers that haven't had these things factored into the design and processing is not something to undertake without a measurement system and the skills to use it. Getting it wrong won't make the system completely unusable but it can result in a loss of level, a lack of impact and, in extreme cases, shorter service life for the drivers.  Whatever the case, the system almost certainly won't be optimized to the degree that it could be, or in my view, that it should be.

The true value of a system often comes down to how it makes you look to your customers and how it makes you feel. If you're confident and comfortable with the performance of your system, that's valuable. If you can get set up and packed up in record time, that's valuable. Adding more hardware and more places for things to go wrong doesn't generally make your life better...

All of our gear is full processed to provide a very level and consistent sound across the frequency spectrum there are designed to cover. Because of this, our gear illuminates the quality or lack of track quality in a very apparent way. Considering the HIFI quality of output and tuning involved with our gear, we advise no processing be applied to the channels supplying signal to the BASSBOSS gear. Processing can affect reliability and subsequently may void the warranty. Additionally, I am confident user applied tuning applied to our gear would degrade the sound quality.

Variation between different tracks is typically a clear indication of a quality issue of the tracks themselves. 320Kbps MP3s are the minimum quality that I will suggest using. AIFF and WAV even higher quality (lossless) options as well. Some record pools and bootleg media can even have doubly compressed MP3s. I have also seen folks edit or remix an MP3 and export the mix as an MP3, resulting in double compression. Even doubly compress 320s are going to greatly suffer in the sound quality and dynamics department. Best practice is to export to WAV to avoid a second layer of compression.

As for the best practice on gear setup:
- if only one set of XLRs is dispatched to the system, hit the BASSBOSS subs first and the next manufacturer / tops second.
- you have a second output available, run the tops on one set of XLRs and the subs on the other. This will give you separate volume control and rapid adjustment to balance out the mix for lower quality tracks.
If you lack multiple outs, you could run the booth out to the tops and the master to the subs or vice versa, if booth monitor speakers are not being used.

Generally speaking, when mixing brands of speakers you could be encountering phase and time alignment issues that may be impacting the quality of sound. Also room reflections, nulling, comb filtering could also results in output diminishing cancellations.

I would check and adjust the LF roll off of the tops and the HF roll off the subs (EXO knob) to see if the output improves. You may have a gap between the subs and tops in the 80 to 100Hz area, which can be more apparent on certain tracks.

Additionally, moving the sub itself can improve issues caused by reflections and cancellations. Cancellations are more prevalent in left/right positioning, which lends itself to comb filtering. Sometimes a little bit of repositioning can make a dramatic improvement.

Subs clustered together in the center typically yields the best result, most output, and little to no comb filtering.

Ultimately, if the phase and time alignment between brand X tops and the BASSBOSS subs is identified as the most likely issue, a processor can help.
I am not a fan of the dBX Drive Racks. The Xilica XD4080 is a powerful processor featuring high quality D/A converters to preserve sound. https://xilica.com/products/xd/ This unit should allow the inversion of phase and delay adjustment to the tops. By ear/feel you maybe able to get it tuned well or to be proper, you could incorporate an RTA mic and measure output and get things really dialed in.

Another option would be a digital mixer such as the Behringer X32 Compact. The X32 would give you the ability to mix a show with a band/multiple DJs/multiple performers, manage mics more effectively, have discrete output for tops and subs, and process those outputs for phase and time alignment on the top.

At the moment, my money is on the quality of music and/or a gap in roll off between the subs and tops.

Q: Do you guys considered to open a smaller assembly factory in the EU?

The speakers are B&C, the amps are Powersoft, so shipping these parts to middle-east EU (Hungary or Romania) and CNC the plywood, put all parts together, have the same QC, and sell them within the EU and probably some neighbouring countries could be a very wise move.

No shipping components back and forth EU-US-EU, paying duties every time, cheap but well trained, very smart Hungarians in both mentioned countries. Not to mention the EU market is large and hungry for new things, amazing amount of money and so on....
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With the proper control, few additional safety protocols, and its ready to go!

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We have considered these factors for many years.  We concluded that we would need to establish demand in the EU before it would make fiscal sense to begin to produce there. We haven't yet had the opportunity or resources to begin to market products in the EU. Given the relatively small volume of our production, it makes more sense to concentrate all our production in one location, realizing as much benefit of that integration as possible, and shipping the same products to the various markets throughout the world.

Indeed, the drivers are B&C, the amps, however, were not all Powersoft, and are no longer Powersoft at all. Nevertheless, those components are very small in terms of shipping volume, so a very small percentage of the cost goes into their transportation. Many finished loudspeakers can be built from a single container full of drivers. In addition to which, ground transportation of components through Europe is at least as expensive as ocean transport of the same components from Europe. The finished products are far larger and therefore more expensive to ship. Making finished products in the EU for final consumption in our primary market of the USA would be more impractical than shipping small components from the EU.

We appreciate your suggestions and hopefully one day we will need another facility just to fulfill demand in the EU! Until then, we are getting our production streamlined and optimized so that we can produce the highest quality products at the most competitive prices possible and delivered to any market in the world.

Q: Looking at the specs of the diamon 112 it shows the max sustained spl is 123 and the max output is 133db. I would have thought 6db down from max would put the sustained output around 127db. Is this box capable of that output at all or really maxes at 123?

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The DiaMon doesn't attempt to be a full-range box, so it can provide plenty of SPL in it's role as a mid-high cabinet. I expect the RCF TT-22 offers response that goes a little deeper given its size and weight, which would explain why its maximum output would be slightly lower.

The background on the specs for the DiaMon is that although the peak output was very high, it's sustained output was limited by the DSP to prevent excess heat build-up in the coil.  This is why the peak output and the sustained output are more than the standard 3dB apart. It can do that peak output, but it can't sustain that output long-term safely. In order to provide customers with realistic expectations, we rated it at the limited SPL for sustained output.

The key is not to have the cabinet running into the limiter all the time. It will be able to provide great dynamic sound quality if the dynamics remain in the signal. That done, the peak SPL is achievable and the sustained output is significantly less relevant.

Q: I'm wondering the pros, cons, and theoretical tradeoffs of flown subs as well as what's possible with bassboss subs.

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Boundaries  and proximity are the relevant points here. Subs on the floor are near the boundary that is the floor. Outdoors, subs can be moved away from that boundary by flying them. The theoretical advantage of subs on the floor is that the boundary that is the floor prevents energy loss into that hemisphere of space.  (Hence the term half-space.) The measurement of the output of a source that is taken in open space will measure 6dB higher if the source is against a boundary, ie the floor, than if the source is also in open space. If, on the other hand, the microphone is against a boundary and the source is in open space, the result is that the measurement is effectively in half-space. In other words, it will read 6dB higher than if neither the mic nor the source were against a boundary. Humans are hard to fly, so they are usually on the floor. At the operating frequencies/wavelengths of subwoofers, humans are effectively against that boundary, which puts us in half-space anyway.  Whether a subwoofer is against the floor or flown, since the audience is usually on the floor, the difference to the audience is minimal.

What does make a difference to the perception is proximity. If someone is hearing subwoofers that are 8 feet away on the floor, the level at that distance will be higher than if the subwoofers were flown 30 feet away. To achieve the same level in the same place would require more SPL from the flown subwoofers, usually requiring more subwoofers.

Proximity also contributes to the myth of throw. If subwoofers are downstage center, the distance between the subwoofers and the listener in the front row is a very small compared to the distance from the subwoofers to a listener in the back row.  (Perhaps a theoretical 8 feet to a theoretical 128 feet, so a difference of 120 feet.) If the subwoofers are flown, the front row listener may be 30 or more feet from the nearest subwoofer. The back row listener may be a few feet farther from the flown subs but the difference is far less. Let's say that the front-row listener were to experience 120dB at their location about 8 feet from the subs. That would mean the back row listener would likely experience 96dB. But what if the subs were flown? If the front-row listener were to experience the same 120dB at 32 feet from the subs, the back row listener would experience 108dB. Now it would take a lot more output from the subs to achieve 120dB at 32 feet, 18dB more at the source, so the need for more gear becomes evident. The practical take-away here is that flying subwoofers doesn't increase their "throw", it decreases the relative difference between the levels at the front and the back of the listening area.

Introducing another boundary, such as an infinitely tall and wide wall, would decrease the range of space into which the sound radiates by half again, so 1/4 space. The measured/perceived SPL in that region would increase by 6dB. This does assume that the source is immediately adjacent to the wall and the floor, in other words with no distance for reflections to be generated. If the source is far enough from the boundary for reflections to be generated, the increase is inconsistent at best, and the reflections can be destructive.

Add a second wall, (third boundary) and the remaining slice of space is 1/8 of a sphere, aka 1/8th space. If the source is at the intersection of these 3 boundaries, the result is another increase of 6dB in measured/perceived SPL. If the source isn't immediately adjacent to these boundaries, the resulting reflections can cause destructive interference, resulting in little or no perceived increase in SPL.

A ceiling is also a boundary, so subs flown near, or not near enough, to a ceiling can seem worse than subs flown outdoors.

The reflection from a boundary presents as a polarity-inverted mirror-source. The frequencies at which destructive interference occurs between direct and reflected sound vary with the distance between the source and the boundary. The distance from the source to the boundary influences the time at which the returning signal arrives at a given point, thus whether it is in-phase or out-of-phase with the direct signal. Some energy is lost over the distance traveled to the boundary and the distance returning to the point of reference. The amount of energy lost over that distance, combined with the rigidity of the boundary, influence the amplitude of the reflected energy. In places where the reflected energy is out-of-phase with the direct energy, the level is attenuated. It's also possible for the reflected energy to be in-phase with the direct energy, however it will always be delayed, so while the level may be increased at certain frequencies by the reflected energy, the quality and accuracy of the signal is still compromised.

Setting up cardioid subwoofer arrays can help to reduce the number of boundaries contributing reflections, or at least reduce the magnitude of the reflected energy, thus reducing their destructive effects.

A second source presents as a polarity-matched mirror source. Everything about arrival time and energy loss over distance is the same as with reflections. Where the distance between the listener and both sources is the same, all will sum positively. When the listener is farther from one source than the other, the difference in distance can be correlated to a wavelength at which destructive interference will occur. Thus the greater the difference in distance, the lower the frequency at which the destructive interference occurs.

Keeping the gaps between subwoofers (or arrays of subwoofers) to less than 1/2 wavelength of the highest frequency at which they're being used will help minimize the worst destructive effects. (That's about 7' for BASSBOSS subs.)

Now, to answer the really easy questions! The SSP118 and SSP218 are both equipped with fly-points. The VS21 is also equipped with fly-points and shares a dimension (36") with the MFLA, so they can be flown together using optional linking flying hardware.

In technical terms what we’re describing with the word “throw” usually is a matter of SPL at distance. Barring reflections and interference, sound loses energy at a rate of 6dB per doubling of distance and that affects all subwoofers equally. Individual, portable subwoofers can’t be “long-throw” by virtue of the fact that they are not large enough to produce meaningful directivity throughout the subwoofer frequency range. Significant directivity is achieved when a dimension of the radiating area is greater than or equal to the wavelength being produced. Therefor it occurs at higher frequencies before lower frequencies, so it’s possible for a 36” wide box to provide reasonable directivity at 187Hz (not subwoofer range) and marginal directivity at 94Hz (not quite subwoofer range) but effectively no directivity as frequencies extend much below that.

A subwoofer that operates from 80 Hz down to 25Hz, as most of ours do, are radiating frequencies with wavelengths from 14 feet up to 45 feet. (4.2m to 13.7m) Thus one of our larger cabinets could provide marginal directivity at 80Hz but no appreciable directivity on down from there. Even at 80Hz, there would not be sufficient directivity to be called “long throw”. And our subwoofers aren’t particularly large or small relative to other portable subwoofers, so the same rules would apply to any box of similar dimensions.

If a box is designed/tuned to be particularly loud at a particular frequency, it will be perceived to be louder at any distance at that frequency. If that frequency is high enough to be within the range where it can produce limited directivity, it will benefit from an increase in measured/perceived output, on axis, in that range. (Consequently it will be perceived as having less output when off-axis.) Every sub box that I’ve ever heard to be described as “long-throw” was particularly more sensitive at higher frequencies than lower ones. Boxes that are particularly effective at upper-bass frequencies can also benefit from the limited directivity they can provide at those frequencies, making them seem to be very loud and “long-throw” but at the price of flat frequency response and low frequency extension.

A box that produces a broad range of frequencies at the same level may produce less SPL at a specific frequency (compared to a peaky box as described above) but it will transmit the same range of frequencies at the same level at any distance. Flat frequency response is a design choice that precludes tuning for maximum output at a specific frequency. In the design process, the more you’re willing to give up in low-frequency extension, the more output you can achieve further up the spectrum. Also, the narrower the targeted operating bandwidth, the more peak SPL you can achieve. Deeper and smoother are fundamental goals for BASSBOSS subwoofers, so any one-note-wonder has the potential to produce a higher SPL at it’s one note forte.

The only way to achieve directivity at subwoofer-appropriate frequencies is with an array of boxes. It would be inappropriate to describe the subwoofers in such an array as “long-throw”. The array may qualify, but the individual parts cannot unless they are impractically large.

There are two possible conclusions to be drawn from the physics:
1) Portable/touring products described as long-throw subwoofers are not operating in the subwoofer frequency range, so they are not subwoofers, so they don’t exist.
2) Portable/touring products described as long-throw subwoofers that operate in the subwoofer frequency range cannot be long-throw because of the rules of physics, so they don’t exist.

There’s no such thing as a long-throw subwoofer.

If you have only one sub it should always be used in Normal mode. In Normal mode, the sub is phase-coherent with the tops when they are facing the same way.

When you add the second sub (that’s necessary for cardioid mode) it does face the rear and it does go in Cardiod mode. The forward-facing sub remains in Normal mode.

The output of the rearward-firing subwoofer in Cardioid mode is used to cancel the wrap-around energy coming from the Normally positioned, Normal-mode sub.

When it’s in Cardioid mode, the forward output of the subwoofer is not phase-coherent with the tops, so if you put the sub in its normal, forward-firing position and switch it to Cardioid mode there will be a significant gap in their combined response.

If you'd like to read more see our blog post on the subject:

https://www.bassboss.com/david-lee-blog/what-is-cardioid-in-subwoofers

Q: Your specs look a little inflated.  Why do you use non-standard sensitivity ratings?  Like 1 watt / 1meter in an anechoic chamber?  Looks like yours are rated on the floor, with no mention of the input level or amp power for said rating.  Please elaborate.

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Thanks for your question. I understand and encourage skepticism. I know that in this industry it can be difficult to know who to trust because there are a lot of inflated numbers out there. I'd be happy to clarify anything that appears to be unclear.

The practice of using 1 watt at 1 meter is a hold-over from days gone by, when 1 watt was being made by a tube amp and bigger power was difficult to generate. The information it provides is relevant to determining a loudspeaker's sensitivity for comparison to other loudspeakers when each is driven by the same amp. The primary relevance of this is related to comparing passive loudspeakers to see which one would produce more SPL output at 1 watt input. From there one could calculate the theoretical maximum output given the power limit of the amplifier. This doesn't take into account issues like thermal compression, port turbulence, excursion limits or distortion, all of which are extremely low at 1 watt power levels.

Because we sell powered loudspeakers, there is no need to reference the sensitivity of the loudspeaker independent of the amplifier, which is why the specific sensitivity at 2.83V into 8 ohms isn't part of the specifications. Our amplifiers, drivers and processing are matched for ultimate performance as a system. I've seen many specifications of sensitivity that, if you investigate them, reference output SPL at frequencies outside their intended operating band. I've often seen sensitivity and peak SPL numbers for subwoofers based on levels at 120, 160 or 200Hz, which is well outside the range I would consider using a "subwoofer".

It's difficult to have honest numbers be competitive with inflated numbers. Marketing departments and the general public all want to see the biggest numbers possible. Big number for power amps encourage the use of "peak" watts specifications. Our subwoofer amps are rated on continuous RMS power. We don't even list peak power. People who know a little more can see the difference but it's still possible for people to be misled. Part of the problem is that there are misleading standards. For instance, 1 watt / 1 meter measurements aren't always what they seem. The standard recently changed. Technically, the result of putting 2.83V through 8 ohms is 1 watt. If you want to measure an honest 1 watt into 4 ohms, you must put in only 2V to get 1 watt. The current practice of using 2.83V as the standard output voltage regardless of impedance can produce misleading numbers because 2.83V into 4 ohms is 2 watts and 2.83V into 2 ohms is 4 watts. To be strictly accurate, I use the voltage required to produce 1 watt into the loudspeaker's nominal impedance to calculate its sensitivity but it isn't published because that information isn't relevant to the customer's use of a powered speaker. Strictly speaking, input sensitivity of a powered loudspeaker is adjustable via the input attenuator and is driven by a pre-amp output.

While it is useful to test loudspeakers in an anechoic chamber because they isolate the loudspeaker and microphone from wind, weather and reflections, the purpose of the anechoic chamber is to simulate an uninterrupted acoustical field, in other words, outside, suspended above the ground far enough to be able to measure to the lowest operational frequency of the loudspeaker without the influence of reflections on the measurement. An anechoic chamber is an idealized environment in which to measure a loudspeaker, and one that can be used year-round in controlled conditions, but it isn't a natural environment. Loudspeakers developed and tested in anechoic chambers will never be used in such an ideal environment again. In addition to that, the environment is an attempt to simulate an outdoor measurement on a clear, windless day.

For mid-high boxes, an anechoic chamber of sufficient size is possible, however an anechoic chamber will have a low-frequency limit at which the room is too small to not influence low-frequency measurements. Anechoic chamber measurements of subwoofers are very rare because to achieve anechoic conditions down to 20Hz you need to have an extremely large anechoic chamber, something the size of an aircraft hanger. That's why 1/2 space/ground plane measurements of subwoofers are very common. Full-space performance of subwoofers can be reasonably calculated by subtracting 6dB from the 1/2 space measurement.

Our output and frequency response ratings do specify distance and conditions,  such as 1 meter in half-space, aka ground plane. Since the subwoofers are most often used on the ground, the specifications tell you exactly what you will get with the subwoofer on the ground in a very large space or outdoors.

Q: I'm trying to compare your products with other manufacturers and am having trouble figuring out real performance from all the different specs, SPL, wattage, etc. Would you explain how you derive your specs and how the differ from other manufacturer's? I've seen elsewhere on the site that you think specs can be misleading.

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BASSBOSS products should be compared in-person where you can hear, feel and understand the difference. If you compare based on the published specifications alone, you're not going to realize the significant differences that result from the BASSBOSS method of specifying compared to others'.

BASSBOSS: What we tell you:

Minimum sustained performance.

What we don't tell you:

A bunch or irrelevant, hyperbolic nonsense intended to appeal to people who talk a lot more than they know.
Meaningless acronyms intended to make plain physics sound like something we invented.

What does that mean to you?

BASSBOSS products will deliver their specified performance for the length of your gig and every gig.
BASSBOSS products will deliver their specified output throughout their specified frequency range.

Others: (Typical mass-market products.)

What they tell you:

Absolute best case scenario performance.

What they don't tell you:

The specified output performance can only be achieved for a fraction of a second and if you try to achieve the specified peak output for any longer than a fraction of a second the result would be rapid failure or massive distortion followed by failure.

The specified peak output SPL is achieved over a narrow frequency band that may or may not be within the operating frequency of the associated loudspeaker.

The specified output SPL very likely cannot be achieved throughout the specified operational frequency range.

What does that mean to you?

Buying products based only on their published specifications is kinda like proposing marriage based on an online dating profile. You might not get exactly what you were promised. :~)

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Breakdown:

Don’t believe the common audio-industry practice of cherry-picking the highest possible numbers at unusable frequencies and levels that can only be achieved for a few milliseconds. We quote our specs by what can be achieved for extended periods of time and over a wide range of conditions, unlike specs by many manufacturers. Outrageous claims abound in the speaker industry. In the blizzard of hyperbole it is difficult to know what to believe. Many of the speakers on the market are "paper tigers", with performance numbers that can be achieved for mere milliseconds, if at all. This is at its worst with subwoofers.

Sensitivity and peak output numbers are quoted at frequencies well outside the operational band anyone would or should use a subwoofer. Frequency response is quoted for speakers with bass-boosted EQ. Sometimes you can find the truth in the fine print, but OFTEN not.

Claiming a peak output of 137dB looks great on paper but at what frequency can the quoted speaker do 137 dB, and for how long? Can the speaker also do 137 dB at any other frequency, let alone the frequency claimed to be the low frequency corner? Technically a loudspeaker that claims to have a peak output of 137dB and a frequency response of 30-100Hz should be able to do a minimum of 131dB at 30Hz. The fact is that most would not because they are built to achieve that high peak number, not solid, deep bass, so they won't get close to reaching peak performance at their low frequency corner.

‍Assuming the 137 dB was scored at a peak and the box is +-3 dB. If the box in question achieved the flat response to 30Hz through an EQ boost of 3 dB then it will only reach 127dB at 30Hz. This is a 10dB difference between the claimed output and what you get at the low end of the spectrum. That's not just misleading, it's downright fake.

BASSBOSS subwoofers are designed to deliver peak performance where you want peak performance of a subwoofer to be: in the low end! A BASSBOSS subwoofer that is specified to be -3dB at 30 Hz will deliver peak output at 35Hz, not at 135Hz where others seem to be at their best. Our numbers don't look nearly as big on paper as they sound in the real world because real numbers translate to real performance. We could publish amazing, cherry-picked and misleading numbers like the other guys but instead we choose to publish conservative, representative and realistic numbers. Try measuring other manufacturers' products against their specs and you'll be disappointed. Measure ours and they will often be conservative.

You need to be able to achieve the specifications of a product - at the same time, indefinitely, and over years, for them to mean anything at all. Don't believe the hype.

Q: From my understanding, it's best not to use a power conditioner for the BASSBOSS subs and tops. What would be an appropriate power strip option to connect power for BASSBOSS? Would a Furman SS-6B Surge Block be okay? I don't see that it says it's a power conditioner.

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What's best depends on what you need it for. If all you want is more outlets, there are extension cords with multiple outlets that will do that. Those would be my first choice. If what you want is surge suppression, then you just need to choose one that offers the level of surge suppression you feel is necessary for your location. As rare as it is, your electrical system might get struck by lightning, or you might live in Florida! Most UL-listed surge suppressors also include 15A circuit breakers because they have 15A wall plugs, so be aware of how much you plan to draw through it. Single-driver subs won't draw enough to worry about but double-driver subs or multiple subs could push the limits. If you have two smaller subs, use two surge suppressors. RF filtering is not likely to be necessary so don't pay too much for it. I don't see any reason not to use the one you mentioned but there are better and there are worse and far too many options to choose a "best".

What I want is a clear path between the power source and the equipment with minimal resistance and maximum reliability.  In short, a power strip should be well-built and solid, and the less it has on it besides sockets, cable and a plug, the less there is to go wrong.

1) So just to clarify it is extremely possible, specifically talking about subwoofers, when frequency spectrum and quality are similar between two products, it's possible that a 700w sub can have a dB output that's louder or just as loud with the same quality as a 2400w sub? If so, would that be shown on paper as the Max peak SPL? (Main point I'm getting at is if I run into a decision between buying a 700w speaker with 128dB @ 1m, and a 2400w speaker with 128dB @ 1m, the best decision would be go with the 700w speaker, or are there other factors that would make one better than the other?)

‍There is always a trade-off of one kind or another. If there were two subwoofers that actually had identical frequency response characteristics and output characteristics and one of them achieved the performance with 700W and the other with 2400W it would be almost inevitable that the one using only 700W to achieve the output would be much larger. On the other hand, if the two were the same size and the only available comparison was the CLAIM of 128dB, it's not likely that they would both make good on their claim.‍

‍Something is likely to turn out to be an exaggeration. 128dB from 700W? At what frequency? I'd be more inclined to believe the 128dB from 2400W from a subwoofer. At that point you would need to dig deeper into the specifications to find data that supported the claims. Is the 2400W referred to as "peak" power? Probably a red flag. Is the 700W referred to as RMS power? Probably more realistic if the two share a similar price. 700W RMS is often then called 1400W "program" and some companies might even stretch that to 2400W "peak". Look for the *asterisks*. ‍

‍That's where you find the fine print about how they got to those numbers.  If both speakers can actually deliver 128dB and the specification of the amplifier is exaggerated in one case then you can start to see which company is more or less likely to be telling the truth in other areas. The question is, can they actually deliver the 128dB? ‍

‍The next place to look is at the 128dB specification. Does it specify whether that is a measured output or a calculated output? Is there a sensitivity specification? Does the sensitivity specification indicate the load impedance? If it does, you can calculate how much power would be required to reach the 128dB from the rated sensitivity. ‍

‍To achieve a calculated output of 128dB using 700W would require a sensitivity of 100dB at 1 meter with one watt. That's not impossible for a horn-loaded system or a double-18" subwoofer but very unlikely for a direct radiating single 18" subwoofer. To achieve a calculated output of 128dB using 2400W would require a sensitivity of 94.5dB, which is realistic for a single 18" subwoofer. ‍

‍Measured output takes more factors into account. The factors that occur in the real world, such as heat, turbulence and friction. In order to achieve a measured output of 128dB with 2400W, the sensitivity of the loudspeaker would have to be closer to 96 or 97.5 dB at one meter with one watt, depending on how well the driver dissipates heat. ‍

‍This result requires a better quality driver and an amplifier that actually delivers every bit of the 2400W... And the speaker has to survive the testing. I see plenty of specifications that would almost certainly prove fatal to the speaker if anyone tried to achieve them in the real world. ‍

‍So which one would I buy? I would be inclined to believe that a single driver sub with a 2400W amp could produce 128dB. IF the amps are rated using the same method and  the SPL and frequency responses were measured and specified exactly the same way, and they were the same price, etc., the 700W product would appear to be a superior design. On the other hand, if the 700W product is claiming 128dB peak and the 2400W product is claiming 128dB continuous, I would choose the 2400W product.

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‍2) You mentioned "One could build a ruthlessly efficient system that was 112dB at 1 watt but did nothing at 30Hz. One could also build a desperately inefficient system that had 60,000W of power that would move more air with the fans in the amps than with the speakers."Similar question to question #1, are you saying that the efficiency (output dB @ 1 m) and effectiveness (amount of Watts it takes to expend that dB at 1 m) is what a consumer should look at when figuring out what system to buy (assuming main goal is loudness and clarity).

‍Efficiency in this case refers to the conversion efficiency from electrical watts to acoustical watts.‍

‍Yes and no... I'm saying that you can't just look at electrical Watts and ignore the acoustical output that results from those watts. First let me clarify. Efficiency isn't just output dB @ 1 meter.  It's necessary to include three factors, Sound Pressure Level (dB) Distance (1m) and input power (1W) The specification of XdB (SPL) measured at 1 meter with 1 Watt of input power, usually but not always 2.83V into 8 ohms ( aka 95dB 1W1M) This specification indicates the system "Sensitivity", from which we can calculate the system's conversion efficiency as a ratio or percent. i.e. 95dB @1w1m corresponds to a 2% conversion efficiency. While this is very surprising, why is it important, you may ask? Because if your electrical to acoustical conversion efficiency is 2%, the other 98% of the electrical energy (aka Watts) is converted to heat! And heat is what blows woofers. ‍

‍The higher your conversion efficiency, the less power your amplifier needs to deliver, the less heat you have to dissipate from the voice coil and the less current you need from the wall outlet. And a speaker with very low conversion efficiency will get a lot hotter when producing the same SPL as a speaker with a higher conversion efficiency, so there is a tendency for lower conversion efficiency speakers to burn/blow more easily if the demand is for high SPL. ‍

‍From a sound quality, clarity or product purchasing point of view, it isn't absolutely better or worse to have a high or low conversion efficiency on its own. It's something you have to consider in context. For instance, a standard 120V 20A outlet can supply a maximum of 2400W continuously. If a speaker "A" can produce 128dB with 2400W and speaker "B" can produce 133dB with 2400W, then you might want to consider speaker "B" if you need to double your output without requiring another circuit. On the other hand, if speaker "B" is too big to move, you're going to need another circuit. ‍

‍Amplifiers also have conversion efficiency specifications. If amplifier A produces 2000W and is 95% efficient it will require 2,100W to make the output. If amplifier B produces 2000W and is only 70% efficient, it will require 2850W to make 2000W of output. You can see where this may also be a factor to consider. But this is a conversation for another day...‍

‍In the early days of loudspeaker development, loudspeaker efficiency was critical because the early tube amplifiers were only able to deliver Watts in the single digits. If your speaker system wasn't very efficient, you just wouldn't hear it. For most of the history of sound reproduction, amplification was the most expensive element of the system. ‍

‍It still is, but now the cost of the amplification is much lower, especially relative to the costs of drivers, materials, labor and shipping. The weight of the amplifiers has also gone down dramatically, making it a tiny percentage of the weight of a loudspeaker cabinet rather than the amps outweighing the speakers. ‍

‍As the cost of amplification went way down, the market moved to smaller, less efficient boxes using more power to make up for the reduced efficiency. This made the power and power handling specifications more relevant for comparisons. Once the herd had turned and headed in the direction of bigger power numbers, well, let's just they still haven't turned back. ‍The real problem is that they ran off a cliff.

The real problem is that in the madness of chasing the big numbers, the definition of a Watt coming from and amplifier got somewhat diluted. The methods by which amplifiers were rated for output power became vague and, to a large extent, misleading. ‍

‍Here's the definition of a Watt from Wikipedia: "The Watt is defined as 1 joule per second and can be used to express the rate of energy conversion or transfer with respect to time." So 1 joule per second, blah blah, blah, with respect to TIME! There's the key part that got what I call "creatively interpreted".

‍‍If one Watt is 1 joule per second, how many Watts will you use in 1/125th of a second? And if you cycle on and off 4 times in one second, making the duration of the "on" cycle 1/32nd of a second, how much power will you produce/use?  1 joule for 1/32nd of a second is 1/32nd of a Watt. ‍

‍Not officially, but what some amplifier makers want you to believe is that if they can produce a 98V spike for 1/125th of a second, and repeat that 1/125th of a second spike 32 times per second that they are providing 2400W. So, their claims are true for 1/125th of a second but in the long term, the continuous power available from a system rated that way is closer to 600W. ‍

‍You can effectively de-rate these amplifiers by frequency. 1/125th of a second is the cycle duration of 125Hz, so at 125Hz the amplifier can deliver full rated power for one cycle. After that, according to the specification method, it needs a 4-cycle break. At 250Hz, you get 2 cycles, at 500Hz, 4 cycles and at 1000Hz, 8 cycles etc. ‍

‍Going the other way it's not even that pretty. At 60Hz you have to de-rate by half, so 1200W and at 30Hz, you will have no more than 600W. In audio dynamics, a second is a long time, especially for acoustic instruments, but with keyboards and electronic instruments it's not uncommon to see a sustained bass note last well over a second. If your amp can't deliver for more than a few milliseconds, your subwoofers are going to sound gutless. ‍

‍The takeaway is that in order to have a loud, clear and reliable system you need good conversion efficiency and good power. You can't expect low conversion efficiency with big power to be reliable. You can't expect solid low frequency performance from exaggerated power claims. You can't expect extremely high conversion efficiencies from reasonably sized portable enclosures. You can't expect the powerful amplification and the top-quality components that are required to produce the desired results for the cheapest price. What you really need is a properly engineered solution that balances the demands of SPL, reliability, sound quality, depth, weight, cost and electrical consumption. What you want is a ruthlessly practical solution to the demand for excellence.

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‍3) What else matters besides efficiency?

‍See above! 😃

You said "Efficiency is the ratio of work done to the effort expended. Watts are how we measure the effort expended and dB are how we measure the work done". Does efficiency correlate to effectiveness?

‍Not quite. See below.

‍If I read that right, that means it takes less watts for a highly efficient system, to achieve the same dB, as a higher watt/lower efficient system does.

‍That is correct.

‍When comparing two speakers, when a customer's main goal is how well a sub will shake a building, the two things to look at are the efficiency (dB output @ 1 meter) (E.g. 93dB@1 m = 1% efficient; 96dB@1 m = 2% efficient) and peak watts the system has?

‍Actually, if your goal is to shake a building, the most important things to look at are the frequency response and the peak SPL. You need high output at low frequencies. Nothing shakes a building like extremely low frequencies. The deeper the frequency response, the better. But don't look at -10dB figures. The -3dB figure is what you need to consider. Beyond that, many times I see response graphs that contradict the specifications. Look at the response graph. 30Hz is where you start to really shake buildings. ‍

‍You should look for subwoofers with a -3dB specification below 30Hz. And with high maximum continuous SPL. And make sure the high SPL number is measured. And make sure the high SPL number is IN BAND, in other words a subwoofer than will do 140dB is worthless if that 140dB is happening at 140Hz. You will never use a subwoofer at 140Hz, and if you did, you wouldn't be shaking the building, you'd be shaking your head.

‍You explain the dB @ 1m with 1W, calculation very well. That shows how efficient a speaker can be, but then how do you add in the max speaker wattage? Do you take that efficiency rating and use it to compare speakers? (E.g I have two speakers with the same efficiency; one is speaker A is 100w and speaker B is 105watts. Does that mean speaker B will be louder since it has the same efficiency?

‍The short answer to that is yes.‍

‍The second part of the answer is that you won't notice the difference between 100W and 105W. You'll barely notice 150W. You'll notice 200W. ‍

‍The third part of the answer is to point out a technical distinction. You said "speaker A is 100w and speaker B is 105w"...  Technically, the amplifier in A would be rated for 100W and the amplifier in B would be rated at 105W.  ‍

‍Why is this relevant? Because loudspeakers (transducers) are rated for power HANDLING, not power. Their power handling rating is a measurement of heat dissipation capacity. It would have no influence on the output of the speaker. If speaker "C" had a 100W amplifier and a 200W (power handling) speaker it would be no louder than speaker "D" with a 100w amplifier and a 100W (power handling) speaker provided the sensitivities were the same. Speaker "C" would simply have excess thermal capacity, meaning it would probably last longer or prove to be harder to damage.

‍For example: my old Alto Black 18" subwoofer had 2400 Watts and it's Max peak SPL = 133dB @ 1 meter.
Frequency = 27-160Hz

Your ZV18 has 2400 Watts and it's Max peak SPL = ?
Frequency = 21Hz - 95Hz

‍The ZV18 is an extreme example. It's very inefficient in order to get very low. It's not a box that's even intended to get loud. It's not really a fair comparison for either box but it's an interesting exercise to see the extent of the differences.

‍‍First of all you quote the specs for both but you missed some critical differences. This illustrates A: Why these companies use these misleading number and B: Exactly what you need to pay attention to when looking at spec sheets.‍

‍Alto: 2400 watts is peak, only 1200 watts continuous.‍
‍BASSBOSS: 2400 Watts continuous RMS. Thats already double the power. ‍

‍Alto: 27-160Hz. Those are their -10dB specifications. ‍
‍BASSBOSS 21-95Hz +-3dB. That difference will probably result in a 7dB advantage to the ZV18 at 27Hz. ‍

‍The Alto spec sheet says 133dB is the "maximum peak" SPL. Most companies specify peak SPL at 6dB higher than continuous, making the estimated continuous output of the Alto 127dB.

‍‍BASSBOSS has the ZV18 maximum sustained SPL at 127dB and short-term peak output at 130dB. The questions you have to ask yourself are, at what frequency will the box achieve its maximum output? Which one is more likely to deliver the output that it promises? Which one will deliver the most output at the frequencies where I really want it? ‍

‍I would be very happy to subject an Alto Black 18 to the same tests as our boxes in order to get directly comparable data. I'm certain that the peak SPL will be measured at a higher frequency than any of our subs, probably cloe to their 140Hz upper limit. I fully expect that the peak SPL will also measure at a lower number when using the same extremely unforgiving test procedures we use on our boxes. ‍

‍The BASSBOSS response requires no EQ, no boosting the low end.  I also noted that the specification you chose for the Alto was 27-160Hz, which is their -10dB claim. Their -3dB claim is 29-140Hz, which I also don't believe the box will be able to maintain at high output levels. The frequency response graph published in their spec sheet shows clear evidence that equalization is applied in order to achieve those response figures. ‍

‍The benefit of the boost evaporates as soon as the volume is turned up because either the amp will run out of power or the driver will run out of excursion. The Alto box looks like the perfect example of a paper tiger. Make all the numbers look good on paper so it looks like a tiger, and when you get it home it's really just a pussy-cat.

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‍4) If the dB for the ZV18 is the same as the Alto Black, then that would theoretically mean that they are equally as efficient and effective, correct?

‍Efficiency isn't the same thing as effectiveness. Efficiency can be defined and measured. (It could be specified at every frequency but we average to simplify.) Efficiency is like miles per gallon. A sports car and a mini van could get the same miles per gallon but knowing their miles per gallon spec won't tell you how effective the vehicle will be for taking kids to school or for carving through corners on country roads. Same efficiency, different effectiveness. ‍

‍Effectiveness is subjective, it implies effectiveness for a particular purpose. In order to compare effectiveness we would need a metric of measure or a definition. For example, the Alto is more effective as a budget-priced speaker. To answer which one is more effective as a subwoofer requires that we know what you want a subwoofer to do, what your definition of an effective subwoofer is. To get a good sense of which one is more effective at making deep bass, you could look at the published response graphs side-by-side and evaluate which one is likely to produce more and deeper bass. ‍

‍As a general rule, published response graphs will be provided for professional products and not available for products intended for consumers who will never look at the graph, nor understand it. There is no standard for publishing graphs so you have to pay attention to the frequency lines and the SPL lines and note the differences at clearly indicated points. 20Hz, 30Hz, 40Hz and so on. You could go so far as to re-draw them on the same scale. I've done that for some models that were not available to measure. It helps to be handy with graphics software or Excel if you're going to do this, or you can go really old-school and use grid paper and a pencil... ‍

‍The key, though, is to learn to smell/see the BS. When you've looked at enough graphs, you can tell a lot about what's going on. Look for the notes and asterisks about how the data is presented. Is it smoothed? Is it processed? How and where was it measured? Check the scale. How much detail can you see? How many dB per vertical section? 3? 5? 10? Those things make a HUGE difference in how the data is perceived. If the marketing department made or edited the graph, it could be made all about manipulating perception. ‍

‍You have to learn to see past the presentation and find the meaning. From what I can tell, specification sheets are a lot like online dating profiles. Everything is shown in the best possible light, from the best angle and with make-up or air-brushing applied. The genuinely hot products have less to hide, and they are harder to find.

‍If you put them together would they sound any different in regards to ability to shake a building If I'm wrong, could you please educate me further? Like you said, it takes years to become an expert on this stuff, but I'm trying my best.

If I'm right though, as someone who is going to try to promote and sell your product, what justifies the extra money being spent? Durability of the speaker assumed being nearly the same.

‍As I said, the ZV18 is not the best example for comparison to the Alto unit. {Funny, BASS (low) boss -vs- Alto (high)}  There would be a profound (pun intended) difference in how they shook a building. If your definition of effectiveness was "the ability to shake the walls and fixtures of a building with low-frequency sound," I'm confident the ZV18 would win. ‍

‍We also have other boxes that would do that just as well or better. All of our subs would win against the Alto. The SSP118 is a better comparison for the Alto. It’s roughly twice the output and also very durable and reliable. I'd expect you to get almost 4 times the life expectancy, maybe more. And a better warranty. And better service. Theirs says designed and tuned in the USA. Any guesses as to where it's made? I can’t tell from their sheet.‍

‍BASSBOSS is designed & tuned in the USA. The cabinet is made entirely from 18mm Baltic Birch. The BASSBOSS amplifier is recessed into the cabinet to protect it and the connectors from damage, and even spills, and features a locking Neutrik PowerCON connector. Theirs has bluetooth... And a switch to turn off the logo LED... One of these products is built for serious professionals and serious bass-heads. You tell me, which one do you think will be a more effective subwoofer?

😃

Q: Is there a way to tell when it’s being pushed too hard or peaking? I see a protect light wondering if that’s it or not?

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There are multiple, redundant stages of limiting in the processing and the amplifier. The amp won't clip. This means you don't have to worry about keeping an eye on the back of the subwoofer.

There is no clip light because the amp won't clip. It will compress, but there is no light to indicate when the limiter is active. When the volume stops getting louder, then you've reached maximum output.

Follow proper gain-structure procedures and you won't overdrive your subs. If you don't put clipped signal into the subwoofer, the driver will survive. Here are some dos and don'ts that will help your subwoofers have a long and happy life:

Don't use high-pass or low-pass filters on the signal to the subs because the filters are already built-in.
Don't add gain to the point of clipped signals through secondary mixers or processors.
Do use high-pass filters on inputs for vocal microphones.
Don't use turntables unless they are used to control digital files. Turntable technology was developed before high-output, low-frequency subs and they tend to cause feedback and over-excursion because they are never adequately isolated.

The protect light will flash a coded sequence if the amplifier detects a problem. The one you're most likely to see that indicates some kind of limit is the 6-6, which indicates that the amplifier isn't getting enough power to run at maximum capacity. Usually the remedy is to eliminate power strips, light gauge and/or long extension cords or eliminate other power drains from your subwoofers' circuit.

Here are some of the codes, all of which can be found in the owner's manual...
Long flashes followed by short flashes:

1-0  GROUND NOT CONNECTED: Amp still runs. Blinks 10 times on power-up then stops. Check power cord and outlet wiring.
1-1  LINE HOT AND NEUTRAL REVERSED: Amp still runs. Blinks 10 times on power-up then stops. Check power cord and outlet wiring.
1-2  GROUND NOT CONNECTED IN 240V MODE: Amp still runs. Check power cord and outlet wiring.
2-0 GFI TRIP: Amp shuts down. Power cycle to restore.
3-0 OVERHEAT SHUTDOWN: Amp shuts down. Restarts when temperature falls.
3-3 THERMAL LIMITER ACTING: Gain reduced to limit heat buildup. Amp still runs.
4-7  BROWNOUT AT POWER-UP: Amp shuts down. Power Cycle to restore. Check supply Voltage.
4-8  UNABLE TO START AMPLIFIER: Power Cycle to restore. Check or eliminate power conditioner.
5-0 OVER-VOLTAGE: Amp turns off or won’t start. (Seen on 120V versions when connected to 240V outlets.)
6-0 UNDER-VOLTAGE: Amp turns off or won’t start. (Seen when Generators are overdrawn or long extension cords are in use.)
6-6 LINE LIMTER ACTING: Amp still runs. Output reduced to limit current draw.

Ordinarily you should see only a green LED in Normal operation. Your ears should tell you everything else you need to know!

Q: I took out my subs and as the night went on I heard it hit less and less does that mean that it will compress and lower in volume? It def wasn’t hitting as hard as the beginning.

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The short answer to your question is no. The processing will not compress and lower the volume as the event goes on. But, like so many other things in sound systems and acoustics, there's never only a short answer... There are plenty of other things that could contribute to that perception.

The relative output of tops versus subs. If you start out with the blend of subwoofer and tops levels to your liking at lower volumes, and then increase the volume as the night goes on, it may be that your subwoofers will begin limiting before your tops do.  People tend to want 6 to 12dB more SPL at subwoofer frequencies than at tweeter or midrange frequencies.  Sensitivities remaining equal, every octave lower in frequency requires 4x the power to produce the same SPL. In other words, it takes 4x more power to produce 110dB at 60Hz than it does to produce 110dB at 120Hz. If it took 100 Watts to produce 110dB at 120Hz, it would take 400 Watts to produce 110dB at 60Hz and it would take 1600 Watts to produce 110dB at 30Hz.  

Add to that the fact that people want or expect 6 to 12dB more SPL at subwoofer frequencies, which requires doubling the power for each 3dB, and you get to a demand for 12,800W to achieve 122dB at 30Hz in our hypothetical simulation. The usual result is that the subwoofers tend to limit themselves for self-preservation purposes long before the tops need to, and the perception is that the subs lose impact, when in fact they don't, it's just that the tops have an easier time achieving higher SPLs.

The next factor is called "thermal compression". As power is passed through a voice coil, the temperature of the coil rises. The rise in temperature causes a rise in resistance/impedance. With that increase in resistance, even though the amplifier is putting out the same voltage, the total power passing through the coil is reduced. The resulting output is therefor also reduced. The safe extent of thermal compression tends to be below 2dB, so the limiting is set to prevent temperature increases that would result in more than 2dB of thermal compression.

Another thing that could create the impression of a lowering of level is the increase in people, who tend to soak up sound.

A third factor is adrenaline. You're less sensitive when you're excited.

A fourth factor could be the increased demand on the electrical system in the building, from ACs, refrigerators, lights or ice machines.

And finally, the most universal effect that causes a perceived reduction of level is temporary threshold shift. Your ears just get used to the increased stimulation and become numb to it.  

You can verify all these effects using a Z-weighted SPL measurement. (For clarification, A-weighting measures SPL essentially above 1000Hz, C-weighting measures SPL above 100Hz and Z (zero) weighting measures SPL from 8Hz and above. In other words you need a Z-weighted meter in order to measure the output of subwoofers.)

Q: I'd like to buy just BASSBOSS speaker cabinets, or with drivers and use my own amplification. Do you sell only cabinets?

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BASSBOSS products are sold exclusively powered and processed. We offer both in-cabinet and rack amplification, but our amplification solutions are integral to our product performance and reliability.

Our driver choices are carefully sourced and tested, and we are continually on the prowl for better solutions for our products. Because these elements are so crucial to our designs and sound, we only sell completely finished products.

Q: I've known about you for years and your name used to be BASSMAXX. Why did you change it to BASSBOSS?

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If you've been familiar with BASSBOSS from before 2013, you know that we were formerly called BASSMAXX. We've heard a lot of hilarious anecdotes about why we changed our name! Trust us, there were no nefarious intentions.

Simply put, when we applied for a trademark, it came to our attention that another company owned the trademark to MaxxBass. There's even a BASSMAXX for fishermen!

In order to have a trademark all our own, we rebranded to BASSBOSS (and frankly, like it a whole lot better.  😃 Same products, same designs, same folks behind the grill cloth. Bwah-ha-ha!

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Outdoor festival production faces many challenges but one of the most difficult to manage is noise pollution. Bass, aka low-frequency noise propagation, generates the most complaints from local residents. Festivals don't tend to invest in the management of these issues due to the temporary nature of their sound output.

It often falls to the individual sound companies to address these issues to the best of their abilities. Long-distance low-frequency sound propagation is a also significant concern for industrial and transportation sectors. Thanks to their extensive resources, software is available to analyze and predict the acoustical paths and potential remedies. An excellent environmental acoustic assessment program that can assist in noise control design and strategy during planning and construction can be found at olivetreelab.com/Terrain

Other software specific to the concert production industry can assist in predicting and illustrating steps that can be taken to minimize a festival's impact on the surrounding residents. These programs can be used to predict propagation over shorter distances but also offer valuable information related to how to orient the noise sources in order to dramatically reduce the noise propagation that tends to disturb local residents.

Two softwares that are free to the end user are Meyer MAPP XT and AFMG Ease. MAPP predicts the performance of Meyer loudspeaker systems but the laws of acoustics apply to all systems equally. MAPP predicts planar response, which is to say 2-dimensional propagation. You can, however, create a vertical and horizontal prediction of each simulation. Ease allows you to create a single 3D simulation but gives you 2 dimensional views of the propagation.

All these programs deal with propagation in air. Propagation of sound system energy through the ground is not considered. The speed of sound in solids is significantly faster than in air but the energy required to move the mass of the ground over large distances is generally greater than can be generated by common loudspeakers. Trucks and trains weighing many tons rolling over bumps generate far more energy into the ground, which is why the transportation industry is a great resource for noise mitigation information.

With regard to primary isolation, in a multi-family residence or shared commercial space where people share walls, floors and ceilings, isolation is critical. Preventing the direct transmission of energy from solid to solid, such as from a wooden speaker box to a concrete floor, makes a huge difference in how much energy reaches the neighbors' spaces. For instance, placing a speaker on a sponge rubber mat allows the mat to absorb the vibrations rather than transmit them directly into the concrete. On the other hand, over greater distances where structures aren't physically coupled to each other, separating speakers from the earth isn't of significant benefit.

It is known that sand and soils dissipate energy when subjected to cyclic loading, so sand has long been used as an excellent low-frequency absorption medium. Putting speakers on sand is an effective de-coupling mechanism on its own. As far as propagation to distant neighbors are concerned, pallets and blankets appear to have no beneficial effect at the offending frequencies.

There are much more effective and relevant options to consider. To illustrate what can be of benefit, I ran simulations in Meyer's MAPP XT of a speaker on the ground and of a speaker elevated 5 feet above the ground. (See below.)

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The simulation indicates that the elevated speaker (upper simulation) actually propagates more energy over greater distance than does one directly on the ground. (Bottom simulation.) With the speaker elevated, the sound reflected from the ground acts as a second source.

These simulations cover a an overall length of 1500 feet with the stage facing from left to right and the propagation distance of 1125 feet from the speakers to the farthest (right) edge of the simulation. The patterns will hold true over greater distances but wind and the terrain do have an increasing effect as distance increases.

In looking for ways to provide a more comprehensive and scientific approach to solutions, it's useful to note some additional factors that aren't often considered or understood about sound propagation.

Many people have heard that low end is not directional, which is true when a single radiating source is involved, however when there is more than one source of low end, the interactions produce very significant directional effects.

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Image 2 (above) shows the propagation pattern of a stereo pair of subwoofers placed 20' apart. This pattern is viewed from above. It's clearly evident that the interaction between the two sources creates areas of reduced sound propagation, specifically 45 degrees off axis from the direction of orientation.

It's evident that orienting the system(s) has a greater influence over how much energy reaches a particular area than elevating a speaker or any de-coupling concept. A simple plan of orienting systems 45-degrees away from the nearest residences or potential complaint sources would allow the event organizers to minimize the noise levels at the nearest potential complaint points as well as minimize the interference between adjacent systems.

Simply drawing a line on the ground or tying a ribbon between two stakes to show which way a system needs to be oriented allows the participants a guide for system placement without restricting their creativity and all that's required to create sufficient directivity is a simple left-right speaker system.

If a participant wishes to use a large line of speakers, not in a left-right configuration, (shown in image 3, below)  it's still practical to orient such a system so that neighbors both near and far are minimally effected. The area of least radiation for a line of speakers is 90 degrees to their axis but they radiate nearly as much energy backwards as forwards. Pointing a big line of speakers "away from" the neighbors isn't nearly as effective as pointing the speakers crosswise relative to the neighbors. If the organizers know where the neighbors are and what the system configuration is, they can provide the information to the participants about which direction to aim their system, again by providing nothing more complicated than a line on the ground.

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‍‍Even greater control can be achieved using cardioid programming, (as shown in image 4, below, also viewed from above) but it is expected that the majority of the systems in use will not have the capabilities or expertise to benefit from this technology.

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For events that provide the sound systems, and for the audio production companies that serve them, it's worth considering deploying cardioid system configurations to minimize the spill of noise to the surrounding area. Most cardioid configurations are deployed for the benefit of keeping bass levels lower on stage. Even when festival producers specify the need for the control of sound bleeding off site, very few companies are able to utilize these technologies effectively for that purpose. I recommend that anyone interested in growing their production business invest time in learning how to deploy these principles because, as you can see from the information provided in this brief article, even without the application of complex cardioid system setups, it's possible to realize benefits from the naturally occurring interactions between ordinary speakers.

Q: I have another event coming up with an odd setup request... Inside of an abandoned "Olympic size" swimming pool. The depth isn't quite the same though.

The venue mentioned that they set up sound and dance inside of the pool out of courtesy for the residents about 1/2 mile away. When provoked, apparently when setup outside of the pool, they occasionally complain. Permit laws and such are different here or wed just get one. Cops are unpredictable here as well and basically do whatever they feel like. For example, they raided an acquaintance's event yesterday that was at a legitimate music venue for just being excessively loud apparently!

I'm skeptical it'll make -that- much of a difference but I'd really like to know if it will because, as you know, the bass on your systems is a little more intense and travels more than venues are used to. Will it actually direct a large amount of the bass upwards? I imagine a bit but not what they think.

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As far as putting subwoofers in a pool, it’s hard to predict. The sound will bounce out of the pool, and the hope is it will go upwards instead of sideways. It might. A lot also depends on the orientation of the system relative to the neighbors, the other barriers around the pool, the direction of the wind and all sorts of unknown factors, like what the cops had for dinner, whether the sensitive neighbors are home, etc.

If you know what direction to avoid, setting up a cardioid system or setting the system up at a null-lobe angle relative to the potentially offended neighbors might be more predictable. (The null-lobe angle is usually about 45 degrees from directly behind the system, and may be enhanced by putting all the subs in a row, horizontally on the floor, VS21s inside followed by ZV28s outside.) ZV-VS-VS-ZV (See diagram) You might try doing one of those options inside the pool.

Got a bit of rashing from a hard gig? You can bring your gear back to new with this paint:

Touch Up Paint: Speaker Cabinet Coating – DuraTex® Roller Grade Trial Size Pint Kit – Black

Just curious after hearing your subs recently. Is this accurate? I read it online:

Q: “Reproducing content much below about 20Hz makes no sense for a number of reasons and is a completely unnecessary and gigantic burden for both drivers and their amps so it actually harms real world sound reproduction. Of course there are zillions of people who don't understand this so there is a great reason to include correction capability for infrasonic content below 20Hz:”

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My counter-quote to that is, "Reality doesn't have a high-pass filter." Beyond that, it's well documented that human perception isn't limited to the conscious. People can learn to "see" with their tongues or their backs if an interface device is made to provide consistent, corresponding inputs. It's estimated our brains, aka our unconscious, is aware of 40 million elements in our environment at any given time, and that our consciousness is only aware of up to about 30 specific elements. To ignore the possibility, in fact the reality, that we can perceive things in our environment beyond consciousness, and that those perceptions influence our experience of our reality, is miserably inadequate.

It's not easy to produce those very low frequencies, and the ZV28 goest the lowest of any of the BASSBOSS subs. Going below 20 does require a lot of power, a lot of excursion, and can, if done with the same transducers, or insufficient quantities of transducers, compromise the clarity of higher frequencies, but a blanket statement like the one you quoted doesn't take those factors into account. If you don't do it right, it doesn't turn out right. Shocker!

To put that another way, it is a burden to reproduce frequencies below 20Hz, but it's also a burden to put up giant video walls and huge, decorative stages. People do those things to enhance the experience of the audience, ie to give them an experience they can't have at home or anywhere else. Essentially to create a sense of awe. None of those elements will create the sense of awe that a 10Hz sine wave would cause, especially because the sine wave would be congruous and in context with the music, not separate from it and involving a different pathway to the brain.

Q: I see a lot of subs marketing below 20Hz but there is very little musical or movie information there correct? Your subs roll off at 27Hz or so. Is that due to driver selection or DSP limitation ? What dB are your VS21 or DJ118 outputting below 20Hz?  

Is Dubstep EDM etc mostly 30Hz and above ?

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Musical genres aren’t limited to frequency ranges. Producers make those choices. The vast majority of all music, including those genres, rarely include low frequencies. The lower the frequency, the less often it ’s present. The lower the note, the more scarce it is. There’s more 60 than 50, more 50 than 40, more 40 than 30, more 30 than 20 and there’s so little 20 that going to 10 results in almost none. For now. Some genres include deeper notes more often than others. Extremely fast-paced beats such as in psy-trance, don’t allow for low-frequency content in the drums although it uses the drums for the energy and impact. Thus the lowest frequencies common in psi-trace are about 55 to 60Hz. Equally fast-paced drum-n-bass uses sustained low tones and creates the rhythms at higher frequencies. Those tones can extend into the low 40s, but not very often into the low 30s. 30s are rare and I find them in glitch-hop and hip-hop/rap. 20s are so rare that I can’t even find a pattern.

There isn’t much content in recorded music or movies below wherever the producer chose to filter it. It used to be filtered at higher frequencies than it is now. The reasons for the filtering are technological/equipment limitations. If you can’t record it or reproduce it, you don’t even know it's missing. If you can record it but not reproduce it, you filter it out so as not to put the reproduction mechanisms under excess stress.

The other side of the equation is, we can’t buy mass-produced transducers that operate efficiently in very low frequencies because the lack of demand precludes them being mass-produced. It takes a massive vision and a vast investment to break new ground, or go into the unexplored. The moon landings are a great example of that, as are Elon Musk’s rockets. You need motivation, commitment and capital. Making subwoofers of this sort is a very long and interesting path, but too long for this context in this moment. Suffice it to say that the effects were used to influence people back in the 12th century…

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The net effect of increased rail voltages is the ability to produce higher peak SPL. Watts are the product of voltage and current. Amplifiers are voltage amplifiers. The driver effectively determines the demand for current via its impedance.

If you were to compare two amplifiers that were able to deliver 2400W into 4 ohms, with one of them having the ability to deliver higher voltage but unable to deliver the current to exceed 2400W (ie limited to 15A of current delivery, but able to deliver 160V) and the other one having the ability to deliver more current but not the ability to deliver enough voltage to exceed 2400W, (ie limited to 97V but able to deliver 24A) the one with the higher voltage capacity would be able to deliver more power, and therefor more SPL, everywhere above 4 ohms. The one with the higher current capacity would be able to deliver more power below 4 ohms, but excess capacity in those ranges isn’t always useful because that’s where it needs to be limited to protect voice coils.

Thus more voltage results in higher SPL capability at all frequencies that are outside the range limited by current demand.

Bottom line, higher rail voltages are generally better for music.  Higher rail voltage provides increased dynamic headroom, enabling the amplifier to produce louder sounds without clipping, allowing for better handling of dynamic peaks in music, providing clearer and more detailed audio reproduction.