Sound Reinforcement Terms Explained for DJs By:Dave Yantz

April 8, 2008 by Mobile Beat Staff Writer

What I will provide here is the technical terms and their technical definitions. Following that will be my attempt to place the information in a context than almost anyone can understand. I will be as basic as I can, and will site examples if necessary. The explanations will be from a DJ’s point of view and I hope useful in that light.I understand that most DJs do not have a degree in electronic engineering. But the more we understand about the sound systems we run, the better we can make them sound in the different venues in which mobile DJs play. Playing in different buildings and even at outdoor shows, we need a basic understanding of sound terms to help us understand how it all works.

I have copied the terms and their technical definitions from the Peavey Electronics’ web site. I have only used the terms that I thought would be the most useful for us, as DJs.

Thank you Peavey. J


20 Hz to 20,000 Hz. (Twenty cycles per second to twenty thousand cycles per second). The frequency response spectrum of human auditory perception.


This is the AVERAGE range of hearing in humans; age and sex of the person determine the true hearing range. Most people do not hear the full range of 20 Hz to 20,000 Hz. DJ sound systems do not cover the full range of the sound we hear. Our sound systems cut out the low end of the spectrum anywhere from 63Hz (with two and three way enclosures) to 40Hz with good subwoofers. The very low sounds are very hard to reproduce with a “portable” sound system.


Separating the audio spectrum into two bands, i.e., high frequencies (high pass) and low frequencies (low pass) by means of an electronic crossover and using two separate amplifiers or channels of an amplifier. One amp or channel is used to amplify and project the high pass signals (high frequencies) from the high frequency component or horn of the speaker system. The other amp or channel amplifies the low pass signals (low frequencies) and projects them from the woofer or low frequency component of the speaker system, resulting in increased headroom and dynamic range.


Bi-Amping your system can and will make your system “cleaner” sounding by increasing the headroom of the system. More power = more headroom. Your gear does not work so hard to make the same volume or sound pressure.


Operating a stereo amplifier in mono via the bridge mode switch, which then makes Channel A output the positive power rail and Channel B output the negative power rail. Since the signal swings between A and B Channels, the output of the amplifier is twice that of single channel operation.


Connecting one electrical circuit in parallel with another. Example: Paralleling power amplifier inputs.


Depending on your Amp/Speaker combination, bridging your Sub woofer amp (or amps) may increase your headroom and dynamic range. Even if your tops in a bi-amp system are in stereo, the subs, being in mono, will have little to no effect in the stereo image. The human ear cannot tell the direction of sub woofer frequencies. Bridging the amp is also called Parallel Mono because of the A and B sides (or right and left channels) of a stereo amp being in parallel with each other when bridged.

How much of an advantage is bridging your Subwoofer amplifier in a BI-amp system?

For this example I will use two 8-ohm subwoofers and a Peavey CS-800S stereo Amplifier. Connected in stereo, the subs would be receiving 240 watts per channel @8 ohms per channel. (480 watts total both channels.)

The same Amplifier Bridged into the same two 8 ohm Speakers (making a 4 ohm load now) would be making 1200 watts!

Also very important is the “Low Cut” feature found on some power amplifiers, electronic crossovers and EQs. The cut is normally at 40Hz. If you have this feature you should have it to the “ON” Position. This is because MOST sound enclosures (speakers) can not reproduce sub frequencies below 40Hz. This has your amp wasting power making frequencies that the speaker can not reproduce. The result of this is both a waste of your power and your headroom.


Amplifier overload causing a squaring off or undesirable change in the wave form resulting in distortion or perceptible mutilation of audio signals.


Music and other audio are in a wave form that is rounded at the top and bottom. When and amplifier is turned up beyond it’s capability to reproduce the wave form correctly it will chop off the top and bottom of the wave form. This is called a “clip”.

Clipping can cause damage to the amplifier and the speakers. The power amplifier is not the only amplifier capable of clipping. The mixer board pre-amp can be “clipped” as well by trying to boost the signal too much.

If a clip is sent from the mixer pre-amp to the amplifier, the amplifier will reproduce the clip just like any other source and can still damage the power amplifier and speakers without the clip lights on the amplifier ever lighting up!


This power rating represents the most conservative statement of the capability of an amplifier. It is also called “RMS” power. It denotes the amount of power an amplifier can deliver when amplifying a constant steady tone. It is usually measured at a signal frequency of 1000 Hz for a specific distortion. Continuous power in watts: W = V2/R Power in watts equals the voltage squared divided by the resistance of the load.


This is the only power output measurement of an amp to look at when buying an amp. It is the true reading of the power output of the amp. This is the power reading with a single tone just before the amp starts to “clip”. This works the amp harder than we will with music as a source.


An electronic device that is used to separate an audio signal into two or more bands of frequencies or component signals above and below a certain frequency, said to be the crossover frequency or crossover point. Crossovers can be active or passive.


A passive crossover is built into most speaker cabinets in order to separate bands of frequencies from the full range speaker level signal produced by the power amplifier, and routing those bands of frequencies to the proper speaker or driver. Most commonly found speaker crossovers also use iron in the inductors to decrease their size. This can be a source of distortion due to the nonlinearities in the coil from core saturation. The power going to the high frequency drivers must be attenuated due to the increase in efficiency of a high frequency driver as compared to a bass driver. This power has to go somewhere and it’s usually converted into heat through the use of resistors.


The last two sentences are the important ones here. The passive crossover wastes power from the amp by dumping part of the power to a resistor, converting the power to heat inside the speaker cabinet. Too much amp power can burn up the resistor and ruin the crossover.


Electronic or active crossovers don’t have the problem of excess power because only the power needed by the driver must be generated by the amplifier. An active crossover is employed when bi-amping a system. The active crossover separates the audio spectrum (full range) into bands of frequencies above (high pass) and below (low pass) a certain frequency (x-over point). The low pass is rolled off (attenuated) so many dB per octave above the crossover frequency. The high pass is rolled off (attenuated) below a certain crossover frequency at a rate of so many dB per octave. The high pass and low pass outputs of the electronic (active) crossover are connected to the inputs of two separate power amplifiers whose respective outputs are used to drive the high end (horns) or low end (woofers) of a sound system.


A much more efficient way of sending the correct frequency to the correct drivers (speakers inside the cabinet). The active crossover will not waste power from the amp by the use of resistors. All the power made by the amplifier (or amplifiers) is used to make sound. The electronic crossover is used inline before the amplifier in a bi- or tri-amp system. Some brands of amps have them built right in the amp!


A unit of motion referenced to a time period of one second. The frequency of a vibration or oscillation in units per second. 100 Hertz or 100 c.p.s. (cycles per second) refers to the number of times a second (100) a string is vibrated or an amplifier is swinging between its positive and negative supply voltage.


The easiest way to compare this is with common 115-volt house voltage. (North American) The AC current swings from positive to negative 60 times a second, or 60 Hz (Hertz). With voltages, the term only refers to AC (alternating current).

DC (direct current) does not change frequency or polarity.


Originally the “bel” in honor of Alexander Graham Bell, was the logarithmic term called the “transmission unit.” This was used to express the transmission losses of long telephone lines. The “bel”, being too large for practical use, was later changed to “decibel”. The decibel has no actual numerical value, but is used only to express a ratio between two voltages, currents, powers, or impedance’s. BASIS OF THE DECIBEL SYSTEM MATHEMATICS – The logarithm: The exponent of that power to which a fixed number (called the base) must be raised in order to produce a given number (called the antilogarithm). The decibel uses logarithms to the base of 10 called LOG. This is not to be confused with the so-called natural logarithm to the base “e” called LN used in many electronic formulas. Below are mathematical manipulations of antilogarithms and logarithms. Voltage, current, SPL, distance: 20 Log X1/X2 Power = 10 Log P1/P2.

dB (decibel)-

A unit for describing the ratio of two voltages, currents, or powers. The decibel is based on a logarithmic scale when measuring differences in sound pressure level (SPL). The amount of change in sound pressure level perceivable is directly proportional to the amount of stimulus (the more sound present, the greater the change must be, to be perceived).

O dB

In the measurement of SPL or Sound Pressure Level, 0 dB is referenced to the threshold of hearing or auditory perception of a tone of 1000 cycles (hertz) per second (1 kHz). 0 dB must always be referenced to some base of measurement. In gain functions 0 dB is unity gain (1).

3 dB

The amount of SPL gained by doubling the power to a speaker. The amount gained by doubling the number of speakers.

+/- 3 dB

Plus or minus 3 dB is a measurement of frequency response that exhibits no more than +3 dB and no less than -3 dB below a given reference. It is actually a 6-dB window. The Response of 60 Hz to 14 kHz +/-3 dB means that within the bandwidth of sixty cycles per second to fourteen thousand cycles per second, no frequency is +3 dB more nor -3 dB less than a specified reference frequency.

3 dB DOWN (-3 dB)

The point at which a measured power level is 3 dB below the specified level. In an electronic crossover, the point (frequency) at which the high pass signal is -3 dB down in response or power level is considered the crossover point (frequency).

-6 dB

The amount of loss in SPL as you double the distance away from a sound source.


A decibel scale referenced to 0 dBm = 1 milliwatt of power into 600 Ohms or .7746 volts RMS across 600 ohms.


Primarily a British term for gain referenced to 0 dBu = .7746 volts RMS.


A decibel scale referenced to 1 volt RMS; 0 dBV = 1 volt.


A term for power gain referenced to 0 dBW = 1 Watt.


The most confusing term I know of, the decibel is the most misunderstood term in the sound reinforcement world. Even with careful reading above, most people don’t understand.

The decibel is not a fixed number or value. It is used to tell the difference between two values. The way it is figured depends on what you’re working on. Sound pressure is figured differently than electrical values. Voltage, current, SPL (sound pressure level), and distance are all different values.

And all use the term decibel to state their different values. That’s why it’s so confusing!

A decibel is a logarithm formula, if you remember that fact you will be much less confused in the future.


The ratio, usually expressed as a percentage, of the useful power output to the power input of a device. EFFICIENCY RATING OF A TRANSDUCER/ENCLOSURE…is the SPL the unit produces at a 1 W RMS input power level measured 1 meter from the unit. Doubling the input power raises the SPL 3 dB. Doubling the number of enclosures raises the SPL 3 dB. Doubling the input power and the number of enclosures raises the SPL 6 dB. Doubling the distance (near field) lowers the SPL 6 dB.


This is the 1 watt/1meter SPL (sound pressure level) printed on the back of your pro speaker cabinet. The Higher the number the better the speaker enclosure uses the amp power you give it. Or another way to put it is the higher the 1W/1M number; the louder the speaker cabinet is with the same power (watts) given to it. By today’s

standards, a rating of at least 98db 1W/1M should be common for Pro DJ use.


The difference between the average operating power level of an amplifier circuit and the point at which clipping or severe distortion occurs.


Headroom is the volume left in your system between your normal volume level and the maximum output of the system. If you have to run your system wide open all the time to do your shows you have no headroom in the system, and you should consider upgrading your system (amplifier/s and/or speakers). Headroom is good to have because the system will sound better if it is not being strained to its limits. Imagine driving your car everywhere you go with your foot on the floor all the time.


The human hearing system is very well designed. It has a dynamic range of over 120 dB. Contemporary digital recording techniques can only achieve a dynamic range of about 90 dB. The typical threshold of pain is around 140 dB, with discomfort starting around a sound level of 118 dB.

THE NORMAL AUDIBLE FREQUENCY RANGE is considered to be 15 Hz to 20 kHz. The typical hi-fi specification range is 20 Hz to 20 kHz. One has to question the validity of this range since 20 Hz is more “feeling” than “listening”, and most people can’t hear 20 kHz (only the young). Sound reinforcement specifications reflect 50 Hz to 15 kHz (sometimes 40 Hz). Interestingly enough, this just happens to be the FCC limits on FM radio. The typical telephone has a frequency response of 400 Hz to 4 kHz. The human ear does not hear all frequencies at the same


(60 dB, 1 kHz); about 50% of people can hear a 2 dB change; everyone can hear a 3 dB change. Therefore . . . 1 dB frequency response specs are good; 3 dB specs are fair. TWICE AS LOUD TESTING: 50% of people say about 7.5 dB change is twice as loud, some as low as 5 dB, and some as high as 10 dB. This test is very level and frequency sensitive. Higher sound levels produce lower numbers . . . frequencies below 1 kHz and above 5 kHz yield higher numbers. Therefore, since a doubling of power is only 3 dB more, how much more is really required to produce the “twice as loud”; see your power chart! “A CS 800® plays twice as loud as a CS 400®”…”wrong”, only 3 dB louder. With a complex signal, such as program music, a 10-dB change is approximately twice as loud.


I put these in here just for your information. We need to understand basic human hearing as well as Sound Equipment to do the best job possible in my opinion.


The total opposition to alternating current flow presented by a circuit. The resistance to the flow of alternating current in an electrical circuit, generally categorized as either “high” or “low”, but always expressed in ohms. Commonly used to rate electrical input and output characteristics of components so that proper “match” can be made when interconnecting two or more devices, such as a microphone, loudspeaker or amplifier.


For DJ use this term is common for microphones. For microphones it’s very easy for us to tell the differences, as the connectors are different for high or low impedance microphones (1/4 inch for high and XLR plugs for low).

For speakers and amp matches we use the term “ohms” more than impedance. I will address that explanation in the ohms heading, even though impedance is also a proper term.


Another word for room resonance. When sound energy is restricted by boundaries (such as walls, floor, and ceiling), waves are developed at certain frequencies or wavelengths that are integers of the distance between the room boundaries. Room modes or resonance cause standing waves because once the wave is generated it stands there, i.e., the positive pressure peaks (anti-nodes) and negative pressure troughs (nodes) stay stationary within the boundaries.


After setting up in a Hall and walking around and listing carefully, you sometimes can hear the standing waves in the room at certain points (dead spots in the room as far as hearing the sound system evenly everywhere.) The best way to correct this is not with the EQ, but by changing your speaker placement in the room. Changing the angle from the stage to the center of the dance floor or the space from the back of the speaker to the wall can effect the sound greatly. You also can use standing waves to your advantage by making it easier to talk in the table area and louder on the dance floor


The unit of electrical resistance, equal to the resistance through which a current of one ampere will flow when there is a potential difference of one volt across it. Ohm is the unit of measure used to express opposition to current flow. Every wire or part through which electricity passes has some resistance to that passage.


This is commonly used to match speaker/amplifier loads. The main thing to remember about ohm numbers and speakers is that the lower the number, the harder the amplifier has to work and the lower the ohm load on the amplifier, the more power the amplifier will make.

Amplifier load maximums are printed on the amplifier or are in the manual (sure you still have the manual! J

An example of common speaker loads for a stereo amplifier per channel and available amplifier power are, (our example amplifier is a Peavey CS-800S and can use a 2-ohm load per channel Max):

One 8-ohm cabinet per side (240 watts per channel @8 ohms)


Two 8-ohm cabinets per side giving us a 4-ohm load per channel. Speaker cabinets are connected by using the in and out on the back of your Pro speaker cabinets; out from the amp to the first cabinet and out of the first cabinet to the second (400 watts per channel @4 ohms).


Three 8-ohm cabinets per side giving us a 2-ohm load per channel. These are connected the same as the two cabinets per side example except that the out on the second cabinet goes to the third speaker cabinet (600 watts per channel @2 ohms).


One 4-ohm speaker cabinet per side. The advantage to using one 4-ohm cabinet over one 8-ohm cabinet is that the 4-ohm cabinet is 3Db louder. The amp makes more power into the 4-ohm load than the 8-ohm load and you get more volume

(400 watts per channel @4 ohms).


Two 4-ohm Speaker cabinets per side, still using the ÿÿÿ

Mobile Beat Staff Writer (375 Posts)

This is the general editors account for Mobile Beat Magazine and Website. Who reads Mobile Beat online and in print and attends Mobile Beat events? DJs, VJs and KJs to start with, especially those who own and operate mobile entertainment services. They provide music, video, lighting and a myriad other entertainment choices for corporate events, wedding receptions, dances and innumerable other gatherings.

Filed Under: Sound Engineering for Mobile DJs