Speaker Alignment

Ultra 2013

ULTRA 2013 ZV28 and Z5/D

The physical path length is far from the only factor involved in correcting the time alignment. Crossover frequencies, filter types and filter slopes along with the phase response of the other boxes in the system will have a major effect on the time alignment. One of my customers found this free software which will allow you to do an impulse response measurement. Using the impulse response you can align the various components of your system.

Holm Impulse


Using the impulse response measurements you can make adjustments in the crossover filters and delay times to achieve the best possible alignment.  To suggest a delay time based on the path length of the horn without considering the crossover frequencies and slopes would be essentially misleading. Further, if you’re using certain kinds of amplifiers the net time alignment could be further offset by up to 2ms.  There is a super-simple method for aligning speakers that doesn’t involve analyzing graphs. It’s not absolute but it works a lot better than guessing.  And it’s also free.

Put the speakers next to each other facing the same way with the sound sources as close to each other as possible. Using a tone, which can be sourced from an app for a phone or a track downloaded from a website, drive both speakers to the same level. (HIgher level makes it easier but don’t blow your speakers!)  If both speakers operate in the same passband, the frequency of the tone should be near or below the center frequency of the operating band .(If your subs run from 30 to 100 Hz then you would use a tone of 45 Hz or 50 Hz for example.) If the two speakers operate in different pass-bands then the tone used should be the crossover frequency.

Hold a thin piece of paper, such as normal printer paper, between the two sound sources and adjust the delay until the paper doesn’t move at all. (You may also need to polarity invert one of the boxes to get a better alignment.) If the pressure on one side of the paper matches the pressure on the other side of the paper then the sound sources are aligned in time.  It’s an analog concept in a digital world…

Once you have them aligned in that position, if you then move one speaker backwards relative to the other you can measure that distance and apply delay of that distance to the forward-most speaker.

This should allow you to align practically any two woofer-type boxes together, if you have a method for delaying them….

If you need any help please let me know all the components of your system and all the settings you’re currently using and  I’ll see if I can make any suggestions.

Have fun and let me know how it works out…

The BASSBOSS DV8 Monitor

Testing the DUAL8

Testing the DV8 at the BASSBOSS warehouse on Austin International Airport

The Dual Vented 8, or DV8, was inspired by visits to several smaller venues and my attendance at several weddings. Most often I found the systems in use to have insufficient horizontal dispersion and  they were also horribly ugly. Those experiences gave me the inspiration to design a loudspeaker that was sexy-looking, that had very wide horizontal dispersion, could cover a large room, evenly, and delivered frequency response that was smooth and warm.

It seems that the rooms in which people host wedding receptions tend to be set up with the stage on a long wall, a dance floor in front of it and a lot of audience to the far left and far right. A big stack of speakers on either side of the stage would block the sightlines and, large or small, end up putting hideously ugly speakers on display in an otherwise very carefully decorated room. . (Unfortunately, those hideously ugly speakers end up in the photographs .)

For professional system providers and quality-conscious professional musicians, the DV8 offers amazing sound quality and serious output from an ultra-compact, lightweight, attractive, US made baltic birch constructed self-powered loudspeaker. This is the speaker-on-a -stick for your high-end clients. Outstanding speech intelligibility combined with line-array coverage characteristics make it perfect for hotel ballrooms and convention centers.

If you do weddings as a band or DJ, the DV8 was designed to be small, which will make brides and wedding planners happy, elegant, which will make grooms and photographers (and wedding planners) happy, loud, which will make DJs and bands happy and clear, which will make audiences happy.

So far it has also impressed many traveling and local sound engineers. We have received universal praise for the sound quality of the DV8. And not just the sound quality, its appearance has been very well received as well. It’s a wonderful example of form following function. True to its name, the DV8 stands apart from traditional loudspeakers. To accomplish the widest possible horizontal dispersion, I incorporated a line array waveguide for the high-frequency section. This waveguide provides 120° horizontal dispersion and has the additional benefit of projecting the higher frequencies to greater distances than a conventional horn. To maintain the widest possible horizontal dispersion through the midrange it was necessary to use a narrow midrange driver.


Eight inch drivers were chosen because they offered a combination of wide horizontal dispersion and acceptable low-frequency response using minimal cabinet volume. Smaller drivers would have provided better horizontal dispersion but would not have been able to reproduce sufficient low-frequency output. The 8 inch drivers have octagonal cast aluminum baskets so they can be mounted directly adjacent to each other, minimizing cabinet frontal area. The tweeter was mounted forward of the loudspeaker baffle to minimize edge diffraction and maximize horizontal dispersion.

The cabinet is constructed with a forward tilt to take advantage of the coverage pattern of the line array waveguide. When it is mounted on a standard speaker pole, the cabinet is automatically at the appropriate downward angle to cover a large room while maintaining outstanding near field sound quality. The DV8 is a self powered loudspeaker. The amplifier for the dual 8 is mounted in the rear of the cabinet. The handles Incorporated in the sides, top and bottom of the cabinet extend past the amplifier to protect it in transit.  In keeping with the tradition of packing an immense amount of punch in a very small package, the DV8 comes equipped with a 1500 W amplifier. (Big things come in small packages.)

I’m not exactly sure where the DV8 falls in the pantheon of loudspeakers but I’m tempted to suggest that it is well up in the hierarchy of sound quality, qualifies as extraordinarily compact for a professional loudspeaker, is extremely loud, particularly for its size, and dare I say is one of the most attractive portable loudspeakers on the market.

I would dare to claim that using any combination of the metrics below, the DV8 has a very good chance of winning any contest.

Sound Quality

Where Do I Put My Subwoofers?

Amped Austin

Amped Austin

“Where do I put my subs in this room?” I had someone ask me a while ago, and not for the first time… Here is part of my answer:

Having subs apart from each other will cause interference patterns.  Because of energy loss over distance, the further the stacks are apart, the less noticeable the nulls will be, but there will be nulls.  The nulls happen wherever you are 1.5X further from one stack than the other.  In other words, in a straight line, you would have a peak half way between the two stacks and a null half way between that center point and each stack.  If you’re standing 20′ directly in front of one stack and the other stack is 30′ away on a diagonal, you will be in a null. In certain places, delaying of the arrays can be used to manipulate where the hot spots and nulls fall in the room. It can be useful to place nulls at exterior doorways or in DJ booths for instance. Some of the nulling may also be affected by wall and ceiling reflections.(*see note below)  Whenever you have multiple subwoofer locations, there will places in the room where the energy from the farther array will be nulling a good portion of the energy from the nearer array.  In short, to get the most even coverage its most often better to drop one big rock in the pond and make one big wave than it is to drop more small ones around the pond making more random ripples.

Putting all the bass on one end of a room will give you multiple advantages.  The first is that you’ll create one big wave rather than a bunch of smaller ones interfering with each other.

The second is coupling efficiency.  You get effectively twice as much output from two speakers coupled together than you do from two that are spaced apart.

Third, you get improved directional control of lower and lower frequencies the larger the array of cabinets is.  This effectively increases the system’s low frequency efficiency in front of the array while also reducing the energy being lost off-axis, where you might not want it to go.

Fourth, you also create a condition of 1/4 space loading for the cabinets, extending their low frequency response.  If the room isn’t too big, this effect can create a condition where the room dimensions are smaller than the wavelengths and you get rising response (room gain) at the extremely low end of the spectrum.

Finally, you increase the near-field listening distance with a large array, which means you lose less energy over the initial distance from the array with a bigger array.  This all means that the sound level on the dance floor would be quite consistent even if the subs were all on one end.  (Provided they aren’t all facing at a wall on the opposite edge of the dance floor.)(*see note below.)

I imagine that by now you will have guessed that I would recommend putting all your sub boxes at one end of the dance floor.  Under most circumstances, setting up the subs that way and using the right processor settings should give you massive extended low frequency response throughout the room. Adding appropriately capable top boxes with the right alignments will complete the high impact punch and deliver the best possible coverage for the space.

* Notes on reflections:
A null will be created half-way between a subwoofer and any solid, bass-reflecting surface. The reflected wave returning from the wall (or floor or ceiling) will be phase reversed from the direct signal at the mid-point between the subwoofer and the reflecting surface so, in short, put subwoofers against a wall rather than facing a wall.

If you must put subs in the ceiling, make sure they are well over 12′ up, preferably minimum 16′ up if your audience will be directly below them. That way the nulling occurs above their heads.

In a low-ceiling space with a hard roof such as a basement, if subs are on the floor, the ceiling reflection may reduce the apparent bass level directly in front of the subwoofers. In such a case it’s better to stack the subs vertically than lay them on the floor.

For specific information about your space, contact me through BASSMAXX.com.

How To Set a Limiter for Subwoofers

Set Your Limiter or Weep

Set Your Limiter or Weep

How to Set a Limiter for Subwoofers

Limiters are intended to protect your speakers from getting fried by too much amplifier power or by clipping your amplifier.  What follows is a method for setting a limiter that should prevent your speakers from being blown under most circumstances.

First, look up the power rating and impedance of your speakers.  For this example I’ll use 1000 watts at 4 ohms.  Using an Ohms Law calculator (that can be found online or downloaded as an app) you can determine that 1000W at 4 ohms requires 63.25 volts.   This is the power-handling limit of your speakers in volts.

You will need the following equipment:

Signal Generator or audio player with test tones on cd or mp3

Your mixer/pre-amp





The signal generator can be a CD or MP3 player using a prerecorded sine wave tone or it can be a device that generates sine wave.  There are plenty of sources for either on the internet.

The limiter should have a threshold control, an attack time control, a release time control and an output control.

The signal flow should be:

source mixer/pre-amp crossover (LF out) limiter subwoofer amplifier voltmeter

In a DSP the crossover and limiter will be in one unit but the procedure is the same.

Connect the voltmeter to the output terminals of the amplifier. Do not connect the speaker(s).

Turn your amp’s attenuators to maximum output, adjust the threshold of the limiter to maximum and set the output gain to unity or 0dB.  Set the ratio to the highest setting, usually indicated as a high ratio like 20:1 or as much as infinity:1.

Set the attack time to the minimum setting such as 0.1ms.

Set the release time to minimum also, usually about 1ms.  Some may be labeled as a multiple of the attack time so a low number here is also preferred.

Pick a frequency near the center of the operating band of the speaker you’re protecting. For a subwoofer, 50Hz would be a good choice.  Play that frequency on your source device and bring up the gain on the mixer.  You should be able to see the output voltage of the amplifier rise on the voltmeter.  Bring the level on the mixer up until the clipping indicators come on.  At this point the output of the amplifier should be above the voltage you determined to be their maximum.  In this example the amplifier should be able to produce over 63.25 volts.  If the amplifiers clip indicators come on before the voltage reading gets to your maximum, your amplifier is underpowered for your speakers, but you still need the limiter!

Leave the signal level up to where the clipping indicators stay on.  Lower the Threshold control until the clip lights go off.  Check the voltage reading on the voltmeter.  If the reading is below your predetermined limit, you’re almost done.

At this point the amplifier is technically not loaded.  When you connect a speaker load, the amp will most likely clip a little earlier than it will when only connected to a voltmeter.  It may be necessary to lower the output of the limiter by as much as 2 to 3dB to account for the additional current required to run actual loudspeakers.  In general, the better the quality of the amp, the less the level will have to be reduced.

If the reading is still above your limit, lower the threshold until the voltage reading remains at the predetermined limit.  You don’t have to worry as much about clipping if you have more power than you need.  Your limiter is going to be used to prevent the overheating of your coils instead of preventing the clipping of your amplifier.

For subwoofers the attack time should be set between 4ms and 10ms.  4ms attack times should be used for amps that are near or at their limits when producing the required voltage.  Longer attack times can be used when an amplifier can deliver more than the rated power of the speakers.  This allows you to take advantage of the power headroom of the amplifier without heat saturating the woofers and melting the coils.

For peak limiters, a fast release time is preferred.  Usually the release time is set as a multiple of the attack time so a 4ms attack time will call for a release time of 8 to 16ms.  In some cases the release time can simply be set as 2x, 4x the attack time to get the same result.

Some DSPs also give the option of a release rate in dB/s or decibels per second.  Again, for a peak limiter, a faster rate is better.  100dB/s is a faster release rate than 25dB/s.

Some general rules of thumb:

A lower threshold setting with a higher output setting will produce a thicker sound but may lack impact.  This is usually less noticeable and more useful for prerecorded music and helps get the average level of the bass up when there isn’t really enough to keep up with the available tops.

A higher threshold setting and a lower output setting will produce a punchier sound and is generally better for live music where the bass isn’t quite as continuous.

Live music requires more dynamic power because the signals aren’t as compressed as in mastered, prerecorded music.  Prerecorded music doesn’t demand as much dynamic power but it tends to have a higher duty cycle, meaning that the average power going to the speakers is higher through the same period of time.  Live music tends to blow things through clipping amplifiers and massive dynamic transients whereas prerecorded music tends to blow things though heat saturation.  Building a system that can handle either duty requires knowledge of the demands of both as well as the implementation of technologies that can meet the demands of both while protecting the system from the dangers of either.

DJs tend to blow the woofers of live systems.  If you have a live music system that’s got lots of dynamic power, and it isn’t a BASSBOSS system, chances are you have more power than your woofers can take long-term.  For DJ sound or prerecorded music it’s suggested that you adjust your limiters to a lower threshold.  The DJ may end up riding the limiters but your woofers will have a better experience.  You can cut the threshold dramatically and make up a bit with output gain. This will give the illusion of more bass while hopefully keeping the speakers out of trouble. You can also lower the output level of the tops on the crossover by a significant percentage of the cut you made in the subwoofer threshold.  The mastered recordings are far more dense than live music so the perceived level won’t be that different.

Live music tends to blow the highs in DJ systems.  If you plan to use a DJ or club system for live music you will probably find the amplifiers clipping on the peaks because live music is more dynamic.  Besides buying more powerful amplifiers, there’s not much you can do other than to turn down the levels or the limiter thresholds. Lowering the overall level will keep the sound quality up and keep the speakers working but may not keep everyone happy.  Lowering the limiter thresholds will sacrifice the sound quality a bit but that’s a smaller price to pay than blown tweeters or midranges.

In a BASSBOSS system everything is done for you.  Massive power is available, massive power handling is provided, all the protections are in place and all the parts are matched and balanced to work together in harmony.

Production companies are being called upon to provide sound for DJ/producers and prerecorded music acts more every day.  Having a system that can handle the demands of anything you throw at it, effortlessly and flawlessly, is infinitely preferable to having a system that crashes and burns when the latest music genre is thrust upon it without warning!  Delicate is not synonymous with detailed.  A system that is detailed can also be robust and powerful.  A massively loud, effective subwoofer system doesn’t have to be brutish or inaccurate.  It isn’t necessary to sacrifice nuance in order to get reliability.   Or to get output capacity.  Yes, when you push the levels to their maximum there will be some loss of detail, but when you have a system with a maximum that’s well above your needs, it’s not working as hard and is therefore even more likely to be able to reproduce the nuance since it’s not working hard.

The patient did not recover :-/

The patient did not recover :-/

Horns V. Vented Subwoofers

Everywhere I go I hear sound systems of dubious quality, I see people blowing woofers and drivers in large numbers and I see people working hard on audio jobs just to make enough to pay to keep their audio systems functioning rather than profiting from their labor and investment.  This doesn’t have to be the case.  I believe its possible to show people that buying a quality system package is ultimately not only less expensive but is, in fact, much more profitable than buying a pile of parts that sort of fit together.


During all my years in the business, I have been on a quest for the ultimate subwoofer. Horn-loaded subs were my preference and I started BASSBOSS essentially based on the performance of the B-One and B-Zero horns. We developed a lot of horns over the years, and continue to do so, because horns have a number of desirable and beneficial properties compared to vented boxes. There was, however, just one thing I was always missing from the horn-loaded subs: Really DEEP bass!  Other than building individual horns that are absolutely massive, or building a segmented horn comprised of many separate elements that combine to make one massive horn, there was not a practical way to build a portable horn-loaded bass cabinet that would reach a minimum frequency of 30Hz.   I had to look for a more practical way to deliver the deep bass I wanted to incorporate into the horns.

Eventually I designed a hybrid box that incorporated a vented section with a horn-loaded section and it produced massively low bass but even with the horn-loaded section, it lacked the impact of the purely horn-loaded cabinets. The box was also quite large and complicated. It was impractical, from a size and cost point of view, to build a cabinet that would combine the vented and horn-loaded segments in one construction. What I then did was begin to develop a vented box to see what I could achieve with a highly optimized vented box. What I got was a really magnificent, musical and ruler-flat subwoofer with massive deep-bass output capabilities and virtually unlimited power handling, but it still lacked the impact I loved about the horn-loaded boxes.

In order to deliver both impact and depth I started combining the two types of boxes and found it to be problematic to integrate the two types. I could get the super-deep bass I wanted from the vented boxes and I could get the massive impact from the horns but getting them to line up and do it together took a great deal of time, measurement and experimentation.  I finally figured it out and it was magic!  Massive shock impact with body and depth I’d never heard or been able to achieve through any horn subs alone, especially since they had to at least fit through a door.

Not only do I understand and agree with the love for the impact of horns, I have improved upon it, to make it even more impactful.  Think of it this way, if an 80 kilogram man hit you in the chest with a solid punch, it would be a hard hit, but if that man was 160 kilograms, that hit would move you much further! Understand that to go an octave lower requires four times (4x) the power, and displaces four times the air for the same sound pressure level. If you can extend the power behind that initial hit by an octave, then it effectively hits four times harder because it literally is four times the mass of air behind the hit. This is effectively what I can do with the combination of the horns and vented boxes. The reason I do it with vented boxes is because to do it with horns would take up more than four times the space and that becomes impractical for both portability and for floor space in a nightclub.

BASSBOSS in Bermuda

Please understand that I am not suggesting using vented subs in the place of horn-loaded subs, I am suggesting using vented subs to extend and improve the impact of the horn-loaded subs.

Regarding the choice of amplifiers, I have heard opinions about all kinds of equipment, some of which are more extreme than others.  I’ve even heard it said that if the signal is ever converted from analog into digital it is un-recoverably compromised. This just proves that one can take anything beyond the realm of practicality.  I do my best to focus on the best practical solution.

The problem with anyone having an opinion about a piece of equipment is that in order for the opinion to be valid as anything more than an opinion, in other words for it to have value as a basis for making a decision, it must be derived from some form of test that eliminates any other variable from influencing the opinion.  Personally I have conducted and participated in double-blind tests between amplifiers where all other parts involved in the test were the same.  From these tests three things were evident:

1) Not everyone agrees on what “better” is.
2) There is very little difference between top-quality amplifiers and most people can’t tell one from another if they don’t know which one is playing.  (People hear with their eyes.)
3) One amplifier may be better suited to a particular speaker than another.

The conclusion to be drawn is that the only way to choose the best amplifier for a particular system is to choose based on its compatibility and perceived superiority for use with the specific loudspeaker components of the system.   In other words, you must somehow choose the loudspeaker system you intend to use and then audition every possible amplifier in order to be assured that there isn’t a better choice somewhere in the world.  Naturally this is impractical unless it is your full-time job to know these things.


So, since most people form their opinions based on limited experience, and based on experiences with too many variables to isolate a causal relationship, it is highly impractical to make a decision based on a number of peoples’ unverified opinions. I can tell you my opinions, which are based on over 20 years experience in the industry and hundreds of system installations and productions, tests and experiments, but they will still be just my opinions, based on my personal preferences, and not necessarily perfectly aligned with yours. In audio there is virtually never an objective opinion. Everyone’s ears are different. Everyone’s preferences are different.

The last time I was involved in a double-blind comparison of amplifiers, the one that was consistently loudest on subs was Powersoft. Which one was loudest was measured and was therefore an objective conclusion.

You can drive yourself crazy trying to figure out the minutia of which components will be the best in every single position in the signal chain.  You can also spend a great deal of money!

If you asked me to design you a system for the ultimate in sound quality, I might make different choice than if you asked me to design you a system for ultimate reliability or ultimate SPL or ultimate simplicity.  Somewhere we must make choices and all choices involve compromises.

Horns have 3 factors that dictate their performance.

Their first limitation is the length of the horn.  The internal path length between the driver and the mouth must be long enough to support one-quarter wavelength of the lowest frequency that horn (or array of horns) needs to reproduce.   If the horn isn’t long enough the pressure wave will exit the mouth while the cone is still pushing forward and the “load” on the driver will fall off, losing efficiency and allowing the driver to hyperextend.  Therefor, a horn must be long enough to support the expansion of the wave to the lowest frequency you need it to play.  Assuming 30 Hz is your low frequency target, the horns you choose must be AT LEAST 2.85m long.

The second limitation is the flare constant.  This is how quickly or slowly the area of the horn increases down the path length. Low frequency horns require a slow flare, making their construction more complicated when folded into reasonably-sized cabinets. When designing a horn with a low frequency flare rate one must choose on a continuum between  efficiency and size.  Most horns that are made in a reasonable size are NOT built with a low flare constant AND a long enough path.  One, the other or both are compromised. This leads to peaky response and “horn honk” and a total lack of depth in their performance. These are compromises I’m not willing to live with.


The third limitation is that a horn (or array of horns) must have a radiating area whose perimeter is at least one full wavelength of the lowest frequency to be reproduced.  For 30Hz that would require the perimeter of the frontal area of the horn array to be 11.43m.  That calculates to an array approximately 3m high by 3m wide or roughly 6 horn-loaded subs per side to deliver the performance my system design will achieve with just 3 subs per side.

From this information you can gather that in order to install a horn-only subwoofer system with flat frequency response to 30Hz would require a large amount of space, cabinets and money.  BUT low frequency response is NOT what people love about horns.  It’s the impact and immediacy of their delivery.  It’s the tight and punchy-ness and the tremendously dynamic transient response.  Therefor, if you use the horns for THAT function and use direct radiators for the extended low frequency response, you can use smaller horns that are specifically designed and perfectly suited to delivering the impact that people love about them.

The TRICK is combining the right horn with the right box in the right way.  The vented box you choose MUST offer low frequency extension well below the horn and must be capable of output sufficient to match that of the horn.  Horns are naturally more efficient than boxes, so they require less power.  They convert more of the electrical energy into acoustical energy.  Boxes are less efficient than horns, so in order to have a box (or boxes) able to match the output level of a horn an octave lower than the horn, that box must be able to handle a great deal of power.  In essence, the trade-off between horns and boxes is size and complexity versus power demand.  Vented boxes can invariably be built smaller and lighter than horns but they require more amplification to reach equivalent output.  Further, boxes can’t deliver the impact of horns due to the limitation of coil/cone inertia.  In most cases, the vented boxes can’t reach equivalency with horns before they reach either an excursion, thermal or distortion limit.  The answer has been to add more vented boxes until there are enough drivers to match or beat a horn-loaded array.  Unfortunately, all those vented boxes might be able to match the output LEVEL of a horn-loaded array but not the transient dynamic response and impact of the horn-loaded array.

While a single horn can’t achieve low frequency extension in a reasonable size due to the limitations of low frequency horn-loading, a single box can reproduce extremely low frequencies. Our SSP218, with 6000W power handling can also reach output parity with a horn. Using 2 of the SSP218s allows for parity to be reached at half power, taking the strain off the system, lowering distortion and ensuring long service life. The combination allows the horn to deliver the impact they are known for and leaves the low frequencies to the slower cones for which they are better suited.

This solution takes advantage of both technologies’ strengths to create a system that offers the benefits of both with the weaknesses of neither. The combination overcomes the limitations of both the boxes and the horns. Provided the cabinets are properly aligned and integrated with each other, (which I do for you,) the shocking impact of the horn combines with the extended depth of the boxes while keeping the size and the power requirements reasonable and affordable.

If, however, you do insist on an all horn-loaded system, the B-Zero/D is the best single-box solution I’ve ever heard. The only thing you won’t have with 3 B-Zero/D per side is the super-low bass that nobody even knows is there until they hear it on a speaker like the SSP218. (In my opinion, the combination is still better than either one alone!!)

Because I design boxes for a living I know the limitations of almost every loudspeaker design and configuration. The difficulty is in trying to explain the difference between competitor’s systems and what I can offer you.  Are you comparing an assortment of speakers and amps to a complete, integrated system? Do they have the necessary time/phase alignment correction in the processing?  Compared to any system I’ve ever heard, I can deliver deeper, louder, harder hitting, better sounding bass, and that is just the beginning.  Its difficult to understand without the experience.  The problem is explaining what “Oh My God!” feels like.

I know what bass is supposed to sound like.  I know what bass is supposed to feel like.  My subs were used on the main stage of the Ultra Music Festival year after year because nobody else could deliver what I can.  We delivered 120dB flat to 30Hz at 120′ (front of house mix position) for 20,000+ people with only 16 SSP218s.  In over 4 years in production, we have never lost a single voice coil in that box and we’ve had Bassnectar, Freq Nasty, Datsik, MSTRKRFT, Deadmaus, Black-Eyed Peas and so many others try to push our systems to the limit.  They still haven’t found the limit.

What is “Oh My God!”?   Its what happens when you hear a familiar song and realize there are layers in the music you’ve never heard before.  Frequencies below the capability of what you’ve heard before, dynamics above it, textures behind it and subtleties within it.  Its the experience of being immersed in the music, swimming in it, being a part of it.  Its beyond hearing it. It turns the air into a liquid.   Its the transcendence that happens when you fully integrate the senses of hearing and feeling into the experience of music and dancing.  For those who get it, its an instant addiction.   It takes your breath away, modulates your voice, caresses your skin, moves your clothes, blows you hair and gives you warm chills….


Such a system will allow you to listen at very high levels and not be in pain.  It sounds big and effortless and open.   From barely on to dangerously loud, it will be crystal clear at any level and will keep the same character no matter how high you turn it up.  It offers tremendous loudness with no compromise in quality.  Its intelligently designed so that you have enough headroom that you never force it to strain and distort and compromise the experience of the audience.

In what I am offering, there is no need for you to have to choose tight over deep or deep over tight or make any compromise.  I know how to set up subwoofer systems better than anyone.  I’ve been doing this continuously since I left the hi-fi business in 1987.  My companies have  done the sound for over 100 clubs and easily as many shows and festivals.  You will have tight, controlled, kicking bass that goes lower than any song you are ever likely to play on it.

DSP Considerations

Symetrix SSP Layout

I believe that using one good quality DSP is preferable to having all the components as separate analogue pieces.  There are arguments for the superiority of analogue sound.  I believe that given all the losses incurred by the cables and connectors required between the pieces, the potential for hum and noise between those pieces, the potential for errors in gain structure, the sheer amount of equipment required and the higher cost of the equivalent quality analogue equipment necessary to do the work of a good DSP, an all-analogue solution is highly impractical.

A top-quality DSP will provide a better overall solution with less chance for error, lower cost, lower noise floor and far greater flexibility and consistency thanks to its ability to recall settings.  Even if you could have a perfectly set and tuned analogue equipment rack, if one knob is bumped, or one potentiometer starts to fail, or one cable connection works loose, the entire signal chain is compromised and you have a huge task just to trouble-shoot and find the problem.  After that, you almost no chance of getting it back to exactly what it should be without a thorough re-calibration of the entire system.  With a DSP, there is a lot less to fail and the setting file can be saved both on the unit and on a computer drive.

If the DSP fails, the down-side is that the entire system is down, but the up-side is that the identical settings can be loaded on a new unit in minutes.  With a DSP-in-the-amps solution, you get the benefit of DSP consistency with multiple-processing redundancy – so if you lose one amp you still have a partially functional system.  With multiple DSP equipped amplifiers, the amps can be re-allocated and alternate settings loaded in minutes, bringing the critical components of the system back on line and properly protected so the show can go on.

So, if the amps have DSPs in them you have what is, in my opinion, the best overall solution.  I’ll add one caveat:  The amplifier and its DSP must be of high quality. Powersoft is of very high quality, but many people have based their opinions of Digital components based on experience with cheap, inferior equipment, which is why some people have a negative opinion of digital processors.

And as far as analogue to digital converters are concerned, a top-quality DSP has to have a top-quality A to D converter. Supplementing it for a separate A-D converter adds cost which might be put to better use by going to a better DSP that already has better A-D converters. When you’re at the level of splitting hairs over A-D converters, you must have already assembled the finest set of components possible, including the highest reference-grade loudspeakers and amplifiers, (with backups,) in order to be able to recognize and take advantage of the benefits that that type of change might offer.

I would rather have two identical DSPs, one for backup, or spare DSP amps than the best-in-the-world DSP, A-D converters or analogue processing rack.  With the proper settings in a good quality DSP, the problems of blowing speakers can be essentially eliminated.  Loudspeaker transducers are passive components and, barring defects, they will not fail unless damaged by the inappropriate application of power or by some physical abuse.  Amplifiers and DSPs have more potential avenues to failure and yet they are generally very reliable.  Like air crashes and car accidents, most loudspeaker failures are caused by operator error.  Many times the only error the operator makes is in believing the power-handling claims of the loudspeaker manufacturer, which leads to an error in understanding the limitations of the loudspeaker components.

Having a properly set-up DSP solution for your specific loudspeakers is the most important factor to consider to ensure their reliability.  Details about how good the A-D converters are or whether analogue sounds better are valid points to ponder, but for those people whose business is providing sound, reliability, consistency and quality should be considered in that order.  All are equally important, but a failure of the first renders all other considerations moot.

Symetrix SSP Layout A

Symetrix SSP Layout A

Loudspeaker Efficiency

ULTRA Miami 2013

ULTRA Miami 2013

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. The only other consideration in this equation is whether the work is being done where and how you want it. 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. Buying or renting either system based on the power rating or efficiency alone would be ill advised.

When you hear a system you like, consider what it is that you like about it, what makes you like it. Consider what you might not like about it, too. When you hear a system you don’t like, consider what its doing to make you dislike it. I would imagine that you won’t be able to tell from listening how many watts its using.

We once had a system out at a festival and the promoter wanted to have a BIIIGG system! We had a big system with about 60,000W of amplification behind it. The noise police had us limited to something like 92dB at the property line, so we were holding way back, but how far back, I wondered? I went to the amps, which display the real-time power output in watts and picked the peak numbers displayed from each frequency band and added up the entire system output we were using to achieve what was inevitably over the SPL limit at 100’. The answer: About 1W.

Sensitivity is a specific measurement of efficiency however without a reference to a frequency range it doesn’t tell you enough to make a choice. Efficiency is the ratio of the work done to the effort expended. In this case, Watts are how we measure the effort expended and dB SPL is how we measure the work done.

One could build that desperately inefficient system that had 60,000W of power amps and would move more air with the fans in the amps than with the speakers and it would be easy to sell to people who only ask how many watts. Alternately, one could build a ruthlessly efficient system that was capable of 115dB at 1 watt. It would be easier to run it off 1 wall outlet but it would be harder to sell to the uninitiated. Given the physics involved, for decent low-range response it would probably also be larger to transport.

To me, the more important consideration in this equation is whether any work is being done where and how you want it. Efficiency isn’t the determining factor for a subjective evaluation of sound quality.

Buying or renting a system based on its power rating alone would be ill advised. Similarly, choosing a system based only on the efficiency rating is also lacking relevant data. Picking a sound system based on how many watts it uses is about as appropriate as picking your girlfriend for how many calories she eats in a day. There are probably more important considerations!

Efficiency can be related to suitability for a particular purpose. For me, the consideration of efficiency isn’t necessarily limited to how many watts a system uses for a given dB SPL. In my view, it could well be expanded to how big a vehicle is required to transport it and how much time it takes to set up and how many people (and sweat) it takes to set it up. In some cases its fair to trade a little electrical energy loss to offset losses elsewhere, especially those losses you feel very personally every time you do a gig!

Republic Live, Austin TX

Travis County Expo Center, Austin TX

Power handling is essentially a measurement of thermal dissipation capacity. Higher power handling only really means that you can shove more power in without melting something. Sensitivity expresses the ratio of electrical power input relative to sound power output. To put this into perspective, a loudspeaker that has a sensitivity of 93dB measured with 1 watt at 1m distance is technically only 1% efficient. 96dB 1W@1M is 2% efficient , 99dB 1W@1M is 4% efficient, 102dB 1W@1M is 8% efficient and so on. For every 3dB more output you get for the same input, the electrical to acoustical conversion efficiency doubles. A system that has a sensitivity of 93dB 1W@1M is converting 1% of the electrical energy into acoustical energy and the rest, 99%, is wasted as heat. For a system that has a sensitivity of 105dB 1W@1M, 16% of the electrical energy is being converted into acoustical energy, which is 16x more than the 93dB sensitive system! That’s 1600% more sound for the same watts!

That’s why how many watts are going in is relatively insignificant and what percentage of it that you get out as sound is so significant. Even in the 105dB sensitive system, 84% of the electrical energy is still being lost as heat, so power handling, the capacity to dissipate heat, is still a major concern. A system that is efficient AND has high power handling is most desirable but in the context of loudspeaker drivers, power handling alone is largely an expression of how much energy can be wasted.

What about peak SPL? Industry practice is to quote peak SPL as a number derived from a calculation of the specified sensitivity of the speaker combined with the available amplifier power, or power handling, which is often quoted as a “peak” number. This is a short-term consideration at best and assumes the specifications are accurate, achievable and representative of broad-spectrum performance. In truth they are mostly cherry-picked and chosen to look good on paper. Very rarely can they be achieved. Choosing a system based on a specification sheet’s peak SPL number is like believing everything you read on an internet dating site: A recipe for disappointment.

Efficiency is the ratio of work done to the effort expended. What do you need from a system in order to get your work done and expend the least amount of energy? That’s the kind of efficiency we work to deliver at BASSBOSS.