Remember: The sooner you get us involved with the planning of your event, the sooner we can help guide you with budget and logistical details.

Step 1. Determine Date, Artists.

The more notice we have, the better the chance we will be available for your event, and that we can guarantee the best pricing and make sure you stay on budget.  Please provide artist technical riders and stage plots when requesting quotes so we can ensure that we have included any essential tools the artist will need to perform their best.

Step 2. Estimate Attendance/Listening Area, Event Duration & Available Power

The proposed listening area and anticipated attendance for the event is a key part in determining an appropriate sound system to meet your needs.   We have a scalable system that can handle a variety of event types and sizes, but each component takes time to set up and transport, so the size of the system needed will affect the labor and rental costs we estimate.  Our largest system can fit through standard doorways and roll into place on wheels.  If the venue is wheelchair accessible (whether with ramps, a loading bay or a freight elevator), we can easily load in our system with one person.  If the only way access to the performance area is to navigate stairs, we'll either need to downsize the rig or hire extra hands to help move the equipment, both of which affect the budget.  

If the venue dimensions and load-in/load-out conditions are unknown when you request the estimate, we may require a site visit to review the layout, parking and electrical infrastructure.  Estimates and site visits are free of charge at this time.

Our smallest rig runs on a single 20A 120v power circuit, but our largest rig requires a Nema14-50 230v plug (like you would use to plug in a dryer or stove).  If you are planning to rent a generator to provide power for the sound system, make sure to let us know what connectors it will have, and have a qualified electrician properly ground the chassis and monitor fuel levels throughout the event.  Our largest rig at full output would require a generator with a minimum of 6,000 running watts and 30,000 starting watts.  When possible, Inverter generators are greatly preferred over AVR type generators for this type of task, and they should always be electrically bonded to the stage frame and connected to earth with a dedicated ground rod.

For the event duration, there are a few factors to consider.  First, it takes us anywhere from 2-6 hours to prepare a sound system for sound check, depending on the load-in proximity and scale of the event.  Second, the more acts that need to sound check, the longer that process will take, especially if the artists are travelling from out of town and may or may not be exactly on schedule.  Sound checks can be 15 minutes to 3 hours, depending on the size of the group, complexity of the instrumentation, number of monitor mixes involved, and number of acts.  A seasoned touring band can complete a sound check in half the time it takes a band that is just starting out.  If the audience area is a multi-purpose area (like a dinner or awards ceremony) we need to know the latest we can be performing our sound-check before silence is required.  Finally, estimate the performance duration, and we add the tear-down/load-out estimate to determine the total labour costs.

Don't forget to allow space for the sound system in the venue design.  It's easy to overlook, but after the dance floor and stage are in place, remember that there will still be stands, speakers, cables, racks, and a mixing position that take up space in the venue, and interconnect cables from the mixing position to the stage that require vehicle-free paths as directly as possible..

Step 3. Sign Contract, Pay Deposit (Typically 50%, 30 days before show).

We only ask for a signed contract and a 10% deposit to book a show and make sure the equipment is prepped and ready for your event. 50% Deposit is then due 30 days in advance of your show date (The 10% deposit is non-refundable for any reason. The 50% deposit is non-refundable but is transferrable to alternate dates, pending availability), with the remaining 40% due via check or cash at the conclusion of Sound Check on your show date.

Step 4. Stay In Touch!

This is one that's often overlooked-- as we get closer to show date, make sure you keep us apprised of any changes to the event, the schedule or the attendance. If you're expecting a sellout crowd, we will want to make sure the ordered sound system remains adequate for your needs.


What is a watt anyway?  It's a unit of measure of electrical power consumption.  It is NOT a volume level.  So why does this question come up so much?  Judging a speaker's maximum sound level by it's wattage rating is akin to estimating a car's top speed by it's fuel economy.  A speaker's function is fairly simple. A speaker converts electrical signals into acoustical energy: sound. By moving back and forth, the speaker increases and decreases the air pressure in front of it thus creating sound waves. A speaker is simply a motor (cone and coil) actracted or repelled to a permanent magnet, in relation to a fluctuating electrical signal input.  In live sound, this signal starts as sound waves, converted to an electrical signal by a very similar motor (a microphone capsule). Once the sound waves are turned into electrical signals, they are amplified, equalized, manipulated in various ways, and then combined into a signal that drives the speaker.  This electrical signal is not constant in amplitude, but rather is a series of short bursts of energy.  

As far as electrical motors go, speakers are fairly inefficient at transferring electricity to acoustic energy, with the wasted energy being turned into heat. When installed in a cabinet, the efficiency (or sensitivity) is rated as dB/1w/1m.  This means a speaker with a 99dB sensitivity produces 99 decibels of sound pressure, measured at 1 meter from the speaker, with 1 watt electrical input.  For every doubling of wattage, you gain 3dB of volume.  A speaker will also have a power handling rating, which is a measure of how much power can be applied to the coil before it is permanently damaged or the speaker moves too far out of the permanent magnet field (excursion).  A "peak" power handling number would be like moving your hand across a candle flame.  Yes, it's hot, but because you don't hold your hand over the flame for long you aren't burned.  A "continuous" power handling number would be similar to holding your hand over that same flame.  The same speaker might have a 300 watt continuous rating, and a 1,200 watt peak rating.  So what number do we use?

If a speaker has a continuous power rating of 500 watts, and a rated efficiency of 99dB/1w/1m, then applying 300 watts continuous of electrical signal would predictably produce a continuous sound level of 126dB at 1 meter (since 550 watts is 27dBW, added to the sensitivity).  Now what if we had a speaker that was rated at 104dB/1w/1m with a continuous power handling of 200 watts? 200 watts is 23dbW, added to the sensitivity of 104dB/1w/1m, and we would predictably produce a continuous sound level of 127dB at 1 meter.  In this example, the 200w speaker is actually louder than the 500w speaker, while consuming less electricity.  Wouldn't this be a favourable characteristic, especially if the entire system needs to run on a single 20A power circuit?  The other speaker would make a better space heater, but that's not why we brought it to the event in the first place.  Adding drivers (more speakers) increases efficiency, but only if they can be positioned close enough to each other for their sound waves to sum coherently.

Another important part of this equation is the loss of sound over distance.  I think I can confidently say that most people would agree that things are quieter when they are further away.  A perfect point-source of sound, outdoors with nothing for the sound to bounce off of,  loses sound pressure at a rate of 6dB for every doubling of distance.  So if the speaker can produce 140dB at 1m, then at 2m it would be 134dB; at 4m it would be 128dB; and at 8m it would be 122dB.  There are other factors at play here, such as the effects of boundaries (think the walls and ceiling of a room), and high frequency losses through air depending on temperature and humidity, but those are the general rules that we have to deal with in nature.  

To have a minimal difference in sound level between the front and back of the room, the speakers need to be high in the air.  The difference in sound level between a speaker 6' off the ground or 12' off the ground, as heard from 100' feet away, is negligible since the distance hasn't changed much in ratio.  However, the difference as heard from 6' feet away would be VERY different, since the speaker distance has doubled.  By using high trim heights, we can reduce the difference in level between the closest listening position and the furthest listening position, for a more consistent listening experience throughout the area.

If you want the simple answer, the "peak" wattage capacity of our entire sound rig is 26,257 watts including sound processors and mixers, but I still don't understand how "useful" that number is without knowing where it came from.  A much preferred unit of measure would be decibels, at a specific distance, with a specific weighting applied, and a specific meter response time.  For example, most outdoor music events with mixed age audiences prefer sound levels below 90dB(A), slow, at mixing position.  The SLOW part of that measurement does not include the fast peak bursts that make live music so dynamic and lifelike.

Something else we need to determine is the average power level. Music using shorter bursts of high energy with quieter parts in between may have a low average power level (typically referred to as RMS or root mean square).  Music with high energy from start to finish (think some genres or dance music, metal and punk music) have a much higher ratio of low to high volume, and thus the continuous power handling required by the loudspeaker system in relation to its peak capacity will have to be higher.  When a voice coil in a speaker motor assembly is given short bursts of energy with times or lower energy between (like a conventional kick drum hit in rock and roll) allows time for the voice coil to cool off between uses.  When this average power is increased, the voice coil temperature increases.  Increased voice coil temperatures can lead to mechanical failure, but first they will increase in resistance within the electrical circuit that is the amplifier and speaker, thereby decreasing the power applied to the voice coil.  This is known as power compression.  What this means is that as a loudspeaker reaches the upper end of its power handling capabilities, you will reach a balance point where adding more power will actually reduce the overall volume while increasing the chance of damage to the driver. The higher the average power level, the sooner this power compression will set in.  This is why careful attention to the TYPE of music intended to be played is important when selecting an adequate loudspeaker system.   

How loud does it need to be?

Music is quite dynamic in nature.  The human voice is capable of a dynamic range (quietest to loudest parts) of over 60 decibels.  The dynamic range of human hearing is about 120 decibels (depending on age and environmental factors).  Recorded music with a dynamic range of 18dB is considered to be very dynamic in nature.  Some modern electronic dance music has a dynamic range of less than 6dB (constant bass heavy thumping).  The magic of live performance lies in its dynamic sound, giving it power and emotion that a recording on the radio simply can't recreate, so it's important we don't forget about leaving headroom in our system sufficient enough to recreate all of those short bursts of energy so the mix doesn't begin to sound stale or congested.

So let's use an example:  If our speaker can produce 100 decibels continuously at the furthest listener location (as calculated by our efficiency, power handling, and distance formulas above), and the noise floor in the room is 70dB (how loud the room is with the sound system turned off due to HVAC, lighting and people) then we have 30dB of audible dynamic range with which to mix our sound before the quietest parts are too quiet, or we reach the upper limit of the reliable operation of the speaker system.  Louder rooms (excited audience, appliances like kitchens or air conditioners) will have a higher noise floor, and therefore would leave us less dynamic range for the music to sound natural and may require an audio system with a higher initial volume.  Music with very little dynamic range will not sound natural, so it's important to make sure that the speaker system specified for the event gives us enough capacity to be heard "above the noise".  


Just like a speaker, sound sources get quieter the further away they are.  The 6dB/doubling of distance rule we talked about above (also known as the inverse square law) becomes very critical in the space between a handheld microphone capsule and the vocalists mouth.  If the microphone is held 8 inches from their mouth (like a news reporter), the sound at the microphone would be 12 decibels quieter than if it were held 2 inches from their mouth (2 inches to 4 inches = -6dB; 4 inches to 8 inches = another -6dB).  Since it's the sound engineer's role to balance the sound levels with the mixer, their job becomes much more challenging if the distance between the mic and the mouth is not kept consistent throughout the performance.  

When someone steps back from a microphone and it gets too quiet for all of the listeners to hear them, the sound engineer (who is paying close attention to these details) will boost the level of that mic signal.  When the same person then steps closer to the microphone, the level is immediately increased and the sound engineer has to reduce that same level for the sound in the house system to stay the same.  For a consistent, professional sounding mix, it's important to be aware of how you are holding the mic and keep it as consistent as possible for the duration of the performance. Also remember that sound can transmit through surfaces, so tapping your finger on the microphone handle will be amplified through the system and should be avoided so that it doesn't distract the audience.

What about the other sounds entering the mic?  Imagine a lead singer in a band who leaves their mic on stage in the mic stand during an instrumental break.  Typically on small stages, the drummer in the band is positioned directly behind the lead vocalist, and now that microphone is picking up the entire drum set and amplifying it.  In microphones, the loudest source at the mic wins, and this goes for foldback monitors as well.  On a typical stage, the vocalist is the quietest instrument acoustically, yet should be the loudest as heard in the mix.  This means the vocal mic usually has more electrical boost than any other mic on the stage, making it very sensitive.  The louder the stage sound (percussion, guitar and bass amps, horns, etc) the more of that background noise gets picked up by the un-muted vocal mic.  For a sound engineer to truly have control over the mix, they want each microphone to pick up as much of the intended source as possible with as little of the background noise as possible.  Imagine boosting a vocal mic in the house system and hearing more cymbals and guitar in the mix as you bring up the vocal channel, and you've just experienced one of the reasons recording studios use isolation booths to separate instruments in a quiet environment.

Now imagine a soft-voiced singer in a band with a loud guitar player.  The singer will probably want their voice quite loud in their monitor so they can hear themselves above the sound of the guitar amp on stage.  The louder the monitor, the more sensitive that microphone becomes, and suddenly the guitar amp is being picked up by the vocal mic and then amplified from the monitor, making it that much tougher to hear the vocals.  In fact, a "hot" vocal mic can actually start to amplify the sound of the monitor itself, which is what leads to that annoying feedback that everyone hates (when the microphone and speaker begin to amplify each other in a cycle, getting louder and louder until something mechanical fails or the microphone level is reduced).  

This is where the term "eating the mic" comes from.  If the vocalist keeps the mic right near their mouth, their voice will be significantly louder (as heard by the mic) than the guitar or the monitor, and the monitor level can be lower to get the same amount of vocal.  Keep in mind that the louder the stage volume, the louder the monitors will need to be, which then creates more "stage wash" noise to be overcome by the house system, taking control away from the sound engineer to create a polished mix.  This is one of the reasons in-ear monitors have become so popular with professional performers.  The in-ear monitors (like an earbud) are too quiet to be heard by the front few rows of the audience or by the microphones themselves, leading to a clearer sound in the house system.  Other techniques to reduce stage volume include drum shields, guitar amp shields, pointing amps away from the audience and vocal mics, and coaching performers to tailor their playing to the size of the venue.  


Speakers and microphones all have "directional patterns".  Microphones used in live sound are typically cardioid or hypercardioid, meaning they are more sensitive to the sounds in front of them than behind them, so we can place the mic at the right angle to pick up what we want it to hear, and reject the sounds we don't want it to hear.  Speakers on the other hand aren't quite as directional as we'd like to believe.  Most speaker systems are comprised of a number of sub-systems, such as subwoofers, mid-range woofers, and high-frequency horns.  The electronics in the speaker system separate the frequencies and send them to the drivers that can most effectively amplify them.  A high frequency horn can very efficiently reproduce high frequencies, but would not be able to reproduce low frequencies very loudly before damaging themselves.  

In order to "point" sound the direction we want to, we need to look at each subsystem individually.  High frequency horns are fairly directional, as the wave lengths of high frequencies are very short (in distance) and therefore can be controlled with a small horn lens.  The lower the frequency, the longer the wavelength, and the larger the device must be to control its direction of propagation.  Therefore a larger horn will maintain its specified pattern control to a lower frequency than that of a smaller horn.  With human hearing picking up sounds as low as 20hz, and as high as 20,000hz, we need to be able to control wavelengths between 56.4 feet long and 0.06 feet long.  

This creates some very unique challenges for the loudspeaker designer, and begs into question any speaker manufacturer who claims their speaker has a consistent horizontal or vertical pattern, regardless of frequency.  Most "pattern angles" that are specified are within a certain range of frequencies, but will not be throughout the entire frequency range of the speaker, so it's important to understand what happens to the sound coming off the back of the speaker.  Also, this pattern specified only means that the sound is 6dB quieter than directly on-axis, not that there's no sound emanating whatsoever.  So if a speaker is listed as being 90 degrees horizontal, it doesn't mean that standing 50 degrees to one side would be completely silent, but rather than it would be quieter by 6dB or more, but that's still not silent.  This affects stage monitors (lows being omnidirectional and being heard behind them in the audience) as well as the main house system (performers hearing the lows of the mix from the stage even with the speakers pointed away).  

Other than horns, we can also use multiple drivers to create a directional pattern through sound wave interference, but you can't get "good" interference without accepting some "bad" or "destructive" interference (known as comb filtering, or large peaks and dips in frequency response in different parts of the coverage area).  The best sounding system will always have the least number of speakers involved.  Adding more speakers to get higher volume levels or better pattern control will always be a trade off for clarity and frequency response.  

For two sources of sound to combine constructively they need to be spaced apart within a 1/2 of the wavelength of interest.  So for a speaker reproducing 500hz, they would need to be less than a foot apart before the benefits of the second source become a disadvantage due to comb filtering.  With subwoofers, their wavelengths are long, and it's fairly easy to add more and more subwoofers to gain sound level without comb filtering.  With high frequencies, the drivers can only be so close to each other physically before it becomes impossible to achieve complete summation without comb filtering.  

Line Array speaker systems have become quite popular in recent years, since it allows the sound provider to stock a number of small cabinets that can be scaled up or down for smaller or larger vertical audience coverage as needed.  The idea is that Line Arrays utilize a number of speaker cabinets arranged in a long vertical array (shaped like a J typically) hung from motorized lifts above the stage.  These systems increase their directional control vertically down to a fairly low frequency (based on it's length).  Directional control is important for live sound, so it's obvious why this would be so intriguing.  Line arrays that are shorter ("dash" arrays as some call them, since it's not a full line) due to site-line concerns, smaller budgets or limited trim height are not able to control the directional pattern of the lower frequencies as well, since the physical size of the array is smaller.  At frequencies below 100 Hz (wavelength of 11.3 ft) the line array which is less than approximately 3 meters long will start to become omnidirectional, so the system will not conform to line array theory across all frequencies.  Above about 400 Hz the driver cones themselves become directional, again violating the theory’s assumptions, and at high frequencies, many practical systems use directional waveguides whose behavior cannot be described using classical line array theory. In short, the geometry of real-world audio line arrays as used in public address systems can only be modeled approximately by line array theory, and only in the 100–400 Hz range. 

Our Synergy Horn point source loudspeakers allow a single loudspeaker to be deployed per side to maximum fidelity and uniform coverage of the audience area without all the interactions of a multi-speaker approach that degrade the sound in different locations in the venue and would be more suscepible to wind and destructive interference over distance.  The SM80 from Danley Sound Labs can easily be heard at the other side of a football field while maintaining its coherence and intelligibility, using less equipment, power, and lifting infrastructure than a "dash" array while maintaining higher fidelity and consistency.  They cannot replace a full-scale line array concert system, but for smaller events they are a much preferred solution to a scaled-down line array.