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Guide for Goobers--PA Basics and Glossary for Newbies


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CraigV here. Setting expectations for this thread: Let's please keep this to information, and no discussion. Post any discussion in a new forum thread. We can always update and/or correct posts here as needed. I'd just like to keep this thread as On-Topic as possible and easy to read. Thanks!

 

 

Yorkville has an excellent PA Guide. To find it, click on the URL below:

 

http://www.yorkville.com/default.asp?p1=23&p2=0&p_id=17

 

They also have a bunch of other useful info in PDF format in the "Resources" section:

 

http://www.yorkville.com/default.asp?p1=39&p2=0&p_id=70

 

 

See also www.jblpro.com/pages/tech_lib.htm A wealth of information in PDF form is available there.

 

Rane's reference library has tons 'o good stuff in PDF, too. See

www.rane.com/digi-dic.html

 

The above article is in English. Most of the rest of the site is in Spanish. If you read the language, there's some good stuff there, too.

 

The Study Hall at www.live-audio.com is an excellent resource, too. Look for the link on the left side of the page.

 

Phil

 

Glossary of Tech Terms (A-D

 

 

 

AFL (After Fader Listen): also called SIP (Solo-In-Place) See PFL.

 

Amp (Amplifier) Emulator: A device, either digital (Line 6 Pod, Behringer Vamp, etc) or analog (Tech 21 Sansamp, etc), that either models or mimics the characteristic sound of a particular amplifier or amp/speaker combo, allowing the stage amplifier to be used solely as a stage monitor. See also Cabinet Emulator

 

Aux or Aux Send: An auxilliary output that allow you to send some of the output from specific mixer channels to monitors or effects devices. There are two types--post (post fader) where the level of the signal sent to the Aux changes with the level of the main channel fader--most commonly used to send audio to effects devices, and pre--where the signal level remains constant, whether the fader is up or down--this type is typically used to send signal to monitors, or occasionally to feed a chamber-type echo.

 

Aux Return: See RETURN

 

BBE (Barcus-Berry Enhancer): A device that claims to phase-correct your signal for greater transparency and articulation, and also offers adjustable frequency boosts at 50Hz and 5KHz.

 

Balanced: A wiring technique which results in the reduction of noise generated in cable runs. Two identical audio signals feferenced to each other are carried 180 degrees out-of-phase (in opposite polarity) on two leads. At the front end of the preamplifier, the signals are turned in phase and summed, turning any noise propagated in the cable 180 degrees out of phase and cancelling it.

 

Beaming: The higher the frequency of a signal, the more directional it will become, and the narrower its dispersion will be over a given distance.

 

Bridge Mode, Bridged Operation:A method of gaining amplifier output power for a single channel. Each channel of a power amplifier operating in bridge mode is responsible for half of the output cycle: The two channels operate in series, and out of polarity with each other. The minimum safe impedance for bridged operation will be twice the minimum impedance for standard two-channel operation. Note: Not all power amplifiersare designed to operate safely in bridge mode. Some will immediately self destruct. See your power amp manual for instructions on bridge mode operation.

 

Cabinet Emulator: A device, such as the Tech 21 Sansamp, H & K Red Box, DOD Active DI, etc, that either models or mimics the sound of a speaker cabinet, thus allowing the signal to the FOH mixer to be taken direct from the emulator box or from the instrument preamp via the emulator box, yet sound like a mic'd instrument. See also Amp Emulator

 

Cabling: The wiring used to connect or interconnect pieces of equipment (sometimes referred to as "cords"). The main types of audio cabling are mic or instrument cabling, speaker cabling and power cabling.

  • Mic cabling is light gauge (22-26 ga), is generally balanced, and may contain one or two pairs of conductors inside a foil or braid shield, plus a ground or drain wire.

  • Instrument cabling consists of a single lead inside3 a foil or braid shield.

  • Speaker cabling may be single pair or multipair. It is generally much heavier gauge (16ga or heavier)than mic or instrument cabling. It is virtually always unshielded.

  • Power cabling (or "AC") carries electricity from house panels, outlets or a generator to the equipment
Cardioid (unidirectional): A mic pickup pattern that when viewed from the side, is roughly heart-shaped, with a null directly behind the mic. The mic is essentially unidirectional, rejecting sound from the rear. Cardioid patterns, in increasing order of tightness, are cardioid, supercardioid and hypercardioid. The latter two patterns, while tighter than the original cardioid pattern, actually have small on-axis pickup lobes directly to the rear of the mic, meaning that a monitor wedge set up in company with the mic must be placed somewhat off to the side.

 

Channel conditioning: The use of dynamics processing and/or effects processing to modify the audio signal present on a particular mixer channel.

 

Clipping: A form of distortion usually caused by overdriving an amplifier. It can occur at a number of points on the audio chain. What's most commonly addressed here is power amplifier clipping, which occurs when a power amplifier is overdriven. The amplitude of the input signal becomes greater than the amp can accurately reproduce. It "clips" off the positive and negative peaks of the waveform going through it, leaving the middle of the signal. Minus the peaks, the signal resembles square waves, which resemble DC. This results in increased heat transfer to the speaker voice coils, which eventually overheat and burn. The cause is generally the overdriving of an amplifier to compensate for insufficient amplifier power to a given speaker/group of speakers.

 

Compressor: A device which compresses the dynamic range of an audio signal. It makes the quiet stuff louder, and the loud stuff quieter. Carefully used, you can boost the average signal level of a channel or of your mix, for an increase in perceived loudness. Badly used, you can turn the output of a great sound system into mush.

 

Condensor Microphone: A microphone which picks up sound waves by sensing the current changes in an electrified diaphragm. Condensor mics can offer superior sound quality, but are generally more expensive and more fragile than dynamic mics. They are most commonly seen in live applications as high hat mics and drum overhead mics, and are often used on snare drums, too. Condensor mics must be powered--either by battery, or by phantom power.

 

Crossover: A device for dividing an audio signal into two or more passbands. Passive crossovers (most commonly found in mid-high speakers and monitor wedges) use capacitors and coils to split the audio signal and direct it to specific drivers. Active crossovers are used ahead of the amplifiers to divide the line-level signal electronically into different passbands and direct them to specific drivers or cabinets.

 

DI (Direct Injection, DI Box): A device which allows the user to take an audio signal directly from a source (usually from instrument output or preamp output), rather than micing the speaker.

 

Discriminate Processor: A multifunction device which splits a signal into several frequency bands, and then processes (compresses, enhances, limits, etc) those bands separately. Sometimes called a mastering processor. Examples: TC Finalizer, Behringer Ultradyne 9024, Orban Optimod. These devices are most commonly digital.

 

DDL (Digital Delay Line): An effects device which allows you to delay signal beyond the short times required for reverb or echo.

 

Distro: See Power Distro

 

Drive Rack: The segment of the live reinforcement audio chain that actually drives the power amplifiers. A drive rack would typically contain FOH EQ, compression and limiting, and in larger systems, delays and crossovers as well. These may be discrete pieces of equipment, or the unit may consist of a do-all digital box such as DBX Driverack, Ashly Protea or Brooke-Siren Systems (BSS) Minidrive or Omnidrive. The systemss named use computer modelling to predict performance, align and adjust the system, and actively monitor and control output to maintain balance in the mix.

 

Driver: A loudspeaker. In live sound applications, drivers are generally divided into two types: cone drivers ("speakers") and compression drivers ("horn drivers"). For some reason, piezo tweeters are usually not referred to as "drivers."

 

Dynamic Microphone: The most common type of microphone used in sound reinforcement. It is basically a speaker in reverse. Sound waves vibrate a diaphragm, which moves a voice coil, which generates an electrical current which is amplified. Dynamic mics are usually cheaper and more rugged than condensor mics, but in absolute terms, are generally sonically inferior to condensor mics.

 

Dry (Dry Signal): An uneffected, unprocessed signal.

 

Dynamics Processor: A device which affects the dynamic range of an audio signal--Compressors, limiters, expanders.

 

 

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Effects Processors/Devices: Devices such as reverb, delay, chorus, flange, ring modulators, etc, which are used to add special effects to an audio signal. Some of the newest also have dynamics processing features and/or EQ as well as effects.

 

Enhancer or Exciter (other than BBE): Most commonly Aphex or Behringer brand. This type of enhancer/exciter adds high frequency harmonics to your signal to fatten it up and add "sparkle." Most enhancers have a bass boost and/or bass phase correction function, too.

 

Equalizer (EQ): There are two main types--graphic and parametric. They are used to modify the frequency content of an audio signal. Graphic EQs are generally used for feedback suppression on mains and monitors, and sometimes for tone-shaping. They are multiband fixed-frequency constant-Q (as a rule) devices. Parametrics are multiband variable-frequency variable Q devices. They can be used to boost or (more commonly) cut specific frequencies. BTW, in the sense it's used here,"Q" refers to the width of the notch that is being cut or boosted. You can sweep the frequency of a parametric equalizer until the notch is directly over a feedback node, then broaden the Q and/or deepen the notch to take it out.

 

Expander: A dynamics processor which increases the dynamic range of a signal--basically the opposite of a compressor. Limited application in live sound, except when used as a gate.

 

FBX (Feedback exterminator): A digital device which analyzes the audio signal, identifies feedback nodes and notches them out. In essence, it is an automated digital parametric equalizer. The most common are the Sabine FBX series and the Behringer DSP1100p and DSP110p

 

FOH: Common abbreviation for Front of House, the portion of a sound reinforcement system dedicated to producing sound for the audience. When the term is used by itself, as in "FOH," rather than "FOH power," it generally refers to either the FOH system in its entirety, the main mixing console, or that console's location.

 

Filter: A device which affects the bandwidth of a signal. There are three types: High pass, which passes signal above a predetermined frequency, either preset or user selectable, Low pass, which passes a signal below a predetermined frequency, and bandpass, which passes a predetermined frequency band.

 

Gate (Full Function Gate): A device which is used as an electronic "switch." Signals that are lower in level than a user-set point (threshold) do not pass the gate. It opens and passes the signal unaltered when the signal is sufficiently loud. A full function gate has adjustable attack (open) and release (close) times, and often a hold (duration) feature, as well.

 

IBEW: International Brotherhood of Electrical Workers. Electricians' union

 

IATSE: (Pron: eye-YAT-see) International Association of Theatrical and Stage Employees. Stagehands' union.

 

IEM: In-Ear Monitor

 

Impedance (Z): The AC version of DC resistance (Re). Speaker impedance is a nominal value. True impedance actually varies with frequency. Impedance is actually resistance, affected by capacitance and inductance, which vary the phase angle of the actual impedance load. The formula for figuring total impedance (Zt) of multiple impedances in parallel on a single circuit is:

Zt=1/(1/Z1+1/Z2+1/Z3+1/Z4...). Don't forget to find a common denominator. (Example: The total impedance of an 8ohm speaker and a 4ohm speaker in parallel is 2.67 ohms, figured as follows: 1/8+2/8= 8 divided by 3 or 2.67). Impedance in series is simply additive: Zt=Z1+Z2+Z3+Z4... The two methods can be combined to control the impedance of multiple speaker arrays. Example: The Ampeg SVT 8x10 cabinet load consists of four parallel legs of two 8ohm speakers in series each. Each leg=16ohms, 16/4=4ohms.

 

Insert (Insert Point): A jack which allows you to insert processors (generally dynamics processors) in a signal path--either on a channel (channel insert), a subgroup (group insert), or the mixer output (mix insert or main mix insert)

 

LD: Lighting Director

 

Limiter: A device similar to a compressor. It differs in that the threshold is set at the point you do not want the signal to exceed. A compression ratio of infinity to one then stops the signal from exceeding the threshold. Most compressors can also be used as limiters by setting the compression ratio as high as possible, and then setting the threshold high enough. Some compressors also have separate limiter circuits.

 

Line Driver: a preamplifier device in the output chain which is used to boost the level of the output signal. Compressors and line amplifiers are also used as line drivers.

 

Line level: The nominal output level of a mixer or the equivalent level in an outboard device. In pro equipment it is usually around +4dBM or +4dBU. In home stereo equipment, it is -10 or -20dBv.

 

Matrix, Matrices(pl): A feature of more expensive consoles, allowing the derivation of one or more independent mixes from the subgroups without affecting the main mix. Generally used for sidefill monitors, center clusters or delay towers.

 

Microphone Pickup Pattern: See PICKUP PATTERN, CARDIOID, OMNI

 

Mic Pre (Microphone Preamplifier or Mic Preamp): A device which boosts the low microphone level signal up to line level. It may be part of the input section of a powered mixer or mixing console, or it might be a stand-alone device.

 

Mixworld: FOH Mix point. Sometimes called Mixer Beach

 

Monitor Beach: Monitor mix point, usually in wings offstage. Some times called Monitor World or Monitor Planet.

 

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Noise Gate: A simple gate found on many compressors. The threshold (opening point) is typically the only adjustment available.

 

Omnidirectional (omni, omni mic): A microphone pickup pattern which is non-directional. An omnidirectional pickup pattern would look like a sphere with the microphone at the center.

 

Output Processor: A (dynamics) processing device used on the mixer output, either inserted before the main mix out, or immediately after the main mix out.

 

PFL (Pre-Fader Listen): The capability of monitoring (usu. by headphones) a signal passing through the mixer before it passes through the channel fader. PFL is roughly similar to having a pre aux send that goes to the headphone monitor section. Contrasts with AFL (After Fader Listen), sometimes referred to as SIP (Solo In Place), which allows you to gauge the relative level of the signal in a group, or the entire group blend by allowing you to hear it post-fader.

 

Parallel Processor: A signal processor, usually an effects device, connected in parallel with a signal, so that a portion of the signal may be routed through the processor, while a portion of the signal bypasses it. This is typical of effects processors, which are generally used in effects loops or on Aux Sends.

 

Phantom Power: The method of providing external power to a condensor mic. DC voltage of identical polarity is applied to the two signal leads of the XLR connector (pins 2 & 3). Voltage of the opposite polarity is applied to the ground (pin 1). This means that there is no voltage differential on the signal leads to generate noise.

 

Phase: The time-domain relationship between two or more signals. Phase issues arise when identical audio signals from two different sources arive at the same point at slightly diffent times. Some frequencies may be cancelled out or attenuated, while others are boosted.

 

Pickup Pattern: The way a mic picks up or rejects audio due to the position of an audio source relative to the mic. See CARDIOID.

 

Piezo Tweeter: A type of high-frequency loudspeaker usually used as a substitute for the more expensive compression driver, or to add high-frequency "sparkle" to a sound system. It consists of a piezoelectric crystal rod coupled (usually glued) to a lightweight metal or plastic diaphragm. Electrical current applied to the rod causes it to change length, thus moving the diaphragm and causing it to emit sound. Piezos have a few advantages over conventional compression drivers: they are inexpensive and they exhibit good power handling. Because they have an inverse impedance/frequency relationship, they do not require a crossover for use, and become more efficient as frequency rises. Disadvantages: the sound of piezos is generally characterized as "notchy" and "harsh." The High frequency response of piezos extends well beyond human hearing into the 30-50kHz range. They may emit ultrasonic artifacts that will cause a PA to feed back.

 

Pin 1 Lift: An adapter plug or short jumper wired with XLR connectors, in which pin 1 (ground) is unconnected. This is used as a ground lift to defeat ground loop hum

 

Pink Noise: A type of audio signal used for measurement and analysis. Pink noise is derived from white noise (all frequencies in the human hearing range at equal amplitude), by filtering it to attenuate the volume of the signal at a rate of 6dB per octave. This compensates for the doubling of the number of frequencies per ascending octave. This produces a signal which has equal energy at every octave. See White Noise

 

Power Distro (Distro or PD): A system to step-down and distribute electrical power to a PA system, lights and backline from a generator and/or house panels. This is analogous to having the electrical wiring for a large house or a business in portable form, ready to be plugged in at a single point on the pole. At its most sophisticated, a distro will generally allow three-phase to be broken out to multiple single phase circuits, filtered and metered. The voltage is stepped down to that which the equipment requires, then distributed throughout the system through breaker panels and cabling.

 

Power Soak: A dummy load that dissipates guitar amplifier output power as heat, rather than as sound, as would a speaker. This allows the amplifier to be driven harder at a lower volume level.

 

Processing: Changing the character of an audio signal through the use of compressors, limiters, EQ, etc.

 

Processor: A device used to change the character of an audio signal. Compressors, limiters, expanders, effects devices.

 

Rat Fur: (Mouse Fur): Thin Ozite-type indoor-outdoor carpet, used as a covering on some MI-grade racks and enclosures.

 

Rat Shack: Radio Shack

 

Return (Aux Return): A line-level input usually tied directly to the main mix buss. Returns may or may not have EQ. If they do, it is generally simpler than input channel EQ. Originally intended for the return of effects signals that were sent out on Aux Sends, now most often used as spare inputs.

 

RTA(Real Time Analyzer): A device used in conjunction with a pink noise generator and calibrated microphone to analyze the acoustic characteristics of a room, allowing you to compensate for hangs, rings, reflections and absorption when setting up a sound system. See PINK NOISE

 

SIP (Solo-In-Place): See PFL, also Solo

 

Send: See Aux

 

Serial Processor: A signal processor, generally a dynamics processor, connected in-line with an audio signal so that the entire signal must pass through the processor.

 

Sidefill(s): A monitor mix or mixes fired cross-stage to allow the musicians to hear what's going on elsewhere on stage.

 

SMPTE (Pron: SIMP-tee): Society of Motion Picture and Television Engineers. Usually used to refer to SMPTE Time Code, which is used to sync audio and video, or multiple machines, or may be used to control console automation.

 

Solo: A monitoring mode which allows the isolation of a channel or group of channels on a console via headphones for monitoring or troubleshooting purposes. See PFL, AFL, SIP

 

Spill: Audio leakage from an undesired source into a mic, For example, drums or guitar amps spilling into vocal mics.

 

SPL (Sound Pressure Level): Audio measurement taking into account the entire audible spectrum. When measured, expressed in dB SPL.

 

Squint: LD or lighting tech. Sometimes referred to as the "filament fairy"

 

Standing wave(s): A phenomenon that occurs when audio is reflected perpendicularly between two parallel hard surfaces. It will "hang" and not decay normally, but will continue to ring audibly. Standing waves of different frequencies may interact to cancel some frequencies and/or reinforce others, and may also create sum and difference tones.

 

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Wet (Wet Signal): An audio signal that has been processed--usually by adding reverb or other effects.

 

White Noise: An audio signal that contains every frequency audible to the human ear--generally 20 Hz to 20 kHz at equal amplitude. White noise is perceived as having more high-frequency content than low, because each successive octave has twice as many frequencies as the one preceding it. For example, from 100 Hz to 200 Hz, there are one hundred discrete frequencies. In the octave from 200 Hz to 400 Hz, there are two hundred frequencies. See Pink Noise

 

Wiring Conventions: Most wiring schemes rely on standard colors, and most interconnects rely on standard wiring schemes for mutual compatibility. In wiring house current (110 volt in the US), black is generally hot and white is neutral, while green is earth ground. In balanced audio wiring, Black is generally low potential (cold), while red or white (depending on the colors available) is the hot (high potential) lead. The shield is the audio ground. In some situations, it may take the form of an aluminum or aluminized mylar shield in contact with a bare, stranded drain wire, in others it may be stranded wire wrap or braid.

 

Some Connector wiring conventions:

    The abbreviation for impedance. The nominal impedance of a speaker is very roughly 1.2-1.5 times its DC resistance (Re or DCR), as measured with an ohmmeter across the speaker terminals.
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What's the +4/-10 button do on some gear?

 

That "+4 -10" button changes the nominal operating level between the two standards. This nominal level is what is referenced to "0dB" on the meters. Since 0dB is an arbitrary point, there needs to be a reference as to what zero actually will be for further measurements or metering.

 

The -10dBV level is often classified as "consumer gear", and the higher level of +4dBu is usually pro gear. +4 dBu audio voltage reference level is equal to 1.23 Vrms. -10 dBV is a standard voltage reference level for consumer and some pro audio use, equal to 0.316 Vrms. RCA connectors are a good indicator of units operating at -10 dBV levels.

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Originally posted by cyber_ops

Not sure if this is the place to ask this, but I'll ask anyways.


What is the proper order in turning on the amp / eq / mixer? Currently the amp stays on (at church) and when I turn the mixer off I get an annoying THUMP. I'm afraid I'm gonna blow the woofer
:(

suggestions?

tanx.

 

Suggest you turn amps "ON" LAST, and "OFF" FIRST.

 

Easiest way to remember--follow your signal flow:

"ON"

1. Mixer

2. Effects

3. EQ/Drive/XO

4. Power amps

 

"OFF"

1. Power Amps

2. EQ/Drive/XO

3. Effects

4. Mixer.

 

Backline should be on and all connections made before amps are turned "on". Board main fader(s) should be down.

 

Power amps should be turned "off" before the backline is shut down, or anything is disconnected.

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courtesy of Audiopile

 

Ok, right or wrong, here's what I do:

 

For technical application questions, I suggest contacting Kathy Evans at Neutrik USA at info@neutrikusa.com

 

1) I prefer using the Neutrik NL4/8 MPR series round style panel jacks. The new style has a metal surface on the front. I recommend seeking these out since the metal makes it just about impossible to jam the cord end upside down in the jack.

 

2) I cut the mounting holes to accommodate the NL8MPR jack, even if you are going to use the NL4MPR jack, since they both have the same footprint for the mounting flange and holes, however the NL-8 body is fatter. Cutting the holes for the NL-8 will accommodate the NL-4 jack fine, and if you ever need to upgrade to the NL-8 jacks, the hole is already there. The only thing is, the NL-8 mounting flange and holes are fairly narrow compared to the cut-out for the NL-4, so if you go in for the NL-8 cut-out you need to be fairly exacting about it or you won't have enough meat left in the cabinet around the mounting flange area to get a good bite with the screws.

 

3) I usually load two jacks in the box, wired in parallel. Depending on the box here is where I put them:

A) Monitors: if the cabinet has side handles I try to recess them into the handle holes, one on each side, so you can go in one side and chain out through the other side. I don't like putting them on the back of wedges, because if you have a dual angle wedge, there can be serious interference problems, and the cord is hanging off the front of the stage...bad deal. In and out the sides seems best. If your wedges are equipped with recessed flip handles, Penn Fab sells flip handles that are pre-punched for Speak-on jacks. Part number is H1057K. These Penn handles will accommodate the small square Neutrik NL-4MP Speak-on panel jack.

B) FOH cabinets I mount them in the back, usually above the existing Jack Panel if possible, usually side-by-side.

 

4) If mounting in wood:

A) First I use drill a small pilot hole with a 1/8" drillbit in the center of where I think I want the round panel jack mounted. It helps to have a look inside first and make sure you are not drilling into anything tender.

B) Next I use a GOOD 2" hole saw drill. I prefer the Carbide Tipped ones that interchange on the mandrel. I drill into the wood about 1/4" to 3/8" following the pilot hole. Then I take a sharp wood chisel and remove that slab of wood to the closest plywood layer. This allows the Speak-on jack to be recessed.

C) Next I change the bit on the hole saw drill to either a 15/16" (1" is fine) for the NL-4 or 1 1/4" for NL-8 and drill the rest of the way through. Ease you way through at the last so you don't splinter the wood bad on the back side. It's best to drill part way through from the outside of the cabinet, the finish drilling through the from the inside. The hole saws can catch pretty easy, so I use a low power drill and make sure I've got my feet well planted.

D) Next I take a can of flat black spray paint and paint around the edges of the recessed hole (this way you don't have a dumb looking white ring around the jack).

E) Wire up the jacks. This is the hard part. I use a Weller pistol grip solder gun and go in hot and fast. Tin your wire and solder posts first. I use 3/4" lengths of shrink tube over the solder points. Parallel jacks are a bitch. It's best to practice on the solder bench before sitting on a milk crate with a bunch of wires sticking out of the holes and trying to do it there. I pre-wire the parallel one on the bench first so it's only single wires to deal with once the wires are threaded through the holes. You might find it easier to wire single wires to the terminals on the jacks and strip a 1/2" area in the middle of the wire to jump to your crossover or speakers. Some folks use female spade terminals on the Speak-on terminals. These do work real good if you can find just the right ones. It's not a bad idea to use a set of these on the ends of the wires going to the parallel jack. This way you can do all the solder work on the bench and then load the soldered jack in, and fish the wires through to the other side and just plug them on. Needle nosed pliers work best to push the spade terminals on the tabs, since the right female spade terminals should fit tight enough that it's difficult to push them on by hand. I solder the spade terminals on the wire rather than crimp them if possible.

F) I mount the Speak-on jacks in with just regular 1" drywall screws. The course thread ones work better than the fine thread ones. Of course if you are mounting on metal, then pop rivets (nail rivets) work well. I use #43 pop rivets. The drill hole size for a #43 pop rivet is 9/64".

G) I usually parallel to the existing 1/4" jack if the wires are decent (which they usually are not). If not, then I go to the crossover with new wires and tie in there leaving the existing wires going to the 1/4" jacks. I have a good sized can full of plastic plugs here for plugging 1/4" jacks that leak air. These plugs are what is used at hydraulic shops to plug orifices during repair proceedures or for sealing hydraulic devices for shipping to keep the dirt out. They are cheap and plentiful at hydraulic supply stores. Take a 1/4" jack with you and you will find plugs that fit nice and tight. The ones you need will be fairly small, but a well stocked supply house should have them.

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Output classes as used in push pull amplifiers:

 

Class A - Each half of the output stage conducts at all times. Not very efficient but does not suffer from issues relating to crossover notch distortion.

 

Class B - Each half of the output stages conducts for exactly one-half of the waveform. This is the most efficient of the linear non-switched rail amp topologies.

 

Class AB - Each half of the output stage conducts for slightly longet than one-half of the waveform. This type of amp can be biased from almost pure class B to almost class A, but is generally configured very close to clas B.

 

Class G/H - Essentially a class AB amp with more than one level of main power supply rail voltage (called multi-tier or multi-step) where the rail voltage is switched instantaniously with respect to the needs of the waveform. This amp is the most efficient of the linear amp topologies.

 

Class D - essentially a digital amplifier, the audio signal input is converted to digital representation of the waveform (PWM) then is level shifted to a higher voltage (still PWM) then recovered via reconstruction filters back into analog audio. This is essentially "power D/A converters". This is the most efficient topology, but is difficult to impliment well into tough loads, faulted loads and at high frequencies and levels. As the technology develops and the semiconductor manufacturers develop highly integrated chipsets to support class D technology, this is the future of larger amps.

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I don't know how accurate the info at this link is, but it seems to jibe with stuff I have read and almost understood here. This web page has a nice overview of stuff like what SPL means and how db and watts and sensitivity relate... geez, almost sounds like I know enough to blow up an expensive rig. :p

 

Anyway, here is the link.

 

link to overview of basics on speakers, spl, db, etc...

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This is a link to an applet that is basically a wavetank. I think it is cool that you can look at some of the things we talk about in this forum a lot. Such as interference between two sources, reflections and standing wave patterns (it does 3D too!). Anyway, just thought i'd pass on a cool link.

-greg-

edit to add the link:

 

http://www.falstad.com/ripple/

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Brand and model aside, how is it that a less-tuned system responds worse to the feedback killer than a well-tuned system?

 

Feedback always happens for the same reason. That reason is you cross the unity threshold point (where the sound of the speaker is as loud into the mic as the sound of your voice into the mic). So maybe the question is how and where did I do that.

 

Let's try a simplified explanation ...

 

Let's consider one mic and one speaker. There is a distance from your mouth to the mic and a different distance from the speaker to the mic. So when the sound from your mouth and the sound from the speaker arrive at two different times it sets up a comb filter. The top of the teeth of the comb are in phase and the gaps between the teeth are out of phase. You can only get feedback on the top of a tooth in the comb and cannot get feedback in the gaps. When you change these distances you will change the number and the width of these teeth.

 

Your sound system is not perfectly flat ... it has peaks and dips ... especially off axis from the speaker and because of the pattern of the mic. Now consider that ragged frequency response. As you turn up the system when one of the peaks in your system crosses the unity threshold right on top of one of the tops of a comb tooth ... presto ... you get feedback.

 

A "more tuned system" simply minimizes the sound level of the speaker from reaching the mic. The first thing here is to aim the speaker away from the mic as best you can while still pointing it in the direction needed. Now there are two tunings you need to make. The first is tuning the speaker so that the sound reaching the audience sounds pleasing. The second tuning is cutting those frequencies that present themselves as peaks to the mic. These may or may not be the same thing. In fact they can be exactly opposite.

 

Feedback (if systems had no distortion harmonic or otherwise) would be a single frequency and would be close to a sine wave. So specific that you could remove that single frequency. The human ear can't identify such a narrow bit of info so your brain just fills it in (integrates it). If fact your ear/brain combination (on average) can't hear the number of points that would be 1/3rd of an octave at frequencies below about 400Hz and about 1/6th of an octave above that. There is of course a first frequency that crosses the threshold and then a second and a third and so on.You need to grab those single frequencies only without grabbing anything else.

 

The standard practice is to throw a 1/3rd oct graphic at this ... but a 1/3rd octave graphic has filters that are about an octave wide ... which you can hear and will miss. They are called 1/3rd oct graphics because the filters are SPACED a third of an octave apart.

 

So what you really want to do is tune your speaker for listening with a 1/3rd oct GEQ for the listening part and use a very narrow parametric EQ for the feedback part.

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Limiters for speaker protection "limit" the amount of voltage an amplifier can deliver to a speaker. This should go a long way to protecting the speaker from thermal burnout (you still need to protect against excursion limits with HP filters). It should be said for the record that it doesn't matter what the waveform looks like ... only the amount of power contained in it ... square wave, sine wave or anything in between. Amp clipping itself does not burn speakers and underpowering does not burn speakers. Too much voltage for too long will burn the voice coil or separate it from the former itself. Protecting your amp from clipping has very little to do with anything except your amp (and most amps have clip limiters built in)

 

Let's start with the basics and add exceptions as we go along

 

Set the limiter for about a 20:1 ratio or higher. In the real world there is almost no practical difference between 20:1 and infinity:1 (because you have almost nothing to drive above threshold. If you set the threshold at "0" you'd have to drive 20 dB more to get the output to rise 1 dB). Anything less than 20:1 doesn't give you much protection. The attack and release will depend of the averaging style of your limiter and the frequencies involved. You probably want the attack on the order of 15ms - 20ms for woofers and maybe 5ms for tweeters (not written in stone). The release is generally set to be 10 times the attack time.

 

Now comes the tricky part ... how do you set the threshold?

 

Now you need to know how much voltage your speaker will handle. It will be called the "continuous", "average", or "rms" rating. If it is called "program" then cut that number in half and if "peak" divide by 4 (generally). Understanding the "real" number is a bit difficult ... so setting a limiter like this depends on how good this number really is. Adjust (down) as necessary for your own comfort.

 

It is helpful to know by which method the manufacturer has used to make this rating. If done by the AES method then it considers the minimum impedance. Speakers have a "nominal" rated impedance e.g. 4 ohms or 8 ohms but the true impedance varies with frequency. It would be typical for an 8 ohm speaker to vary between a low of maybe 6 ohms (at low frequencies) to a high of 20 or 40 ohms at high frequencies. So your amp is delivering varying amounts of "wattage" at different frequencies. You may have to subtract a little bit of power handling ability.

 

Convert the speaker rating to Volts;

 

sq root (speaker rating * impedance)

 

so a 500W 8 ohm speaker looks like this ...

 

square root (500*8) = 63.24 volts

 

(just type "square root (500*8) into google and it will do the math for you)

 

At this point you can simply drive signal through your limiter to your amp and bring down the threshold until you reduce the output of the amp to 63Vac measured on an AC voltmeter (assuming you have proper gain settings)

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If the board and the amps clip at the same time, and your board has the pro standard of +26dBu output capability, but the input and output of the DRPA (and 260 and many other DSP's) clip at +20dBu, then you will be clipping the DSP 6dB before clipping everything else. This is probably one of the most common user errors with DSP in general. Internally, most DSP's operate at around +6dBu before internal clipping, so there's input and output scaling that occurs inside the DSP boxes.

 

For proper system operation, set the DRPA's limiters off, set the DRPA to unity gain (0dB in, and out's to average around 0dB... some may be a few db + and the rest a few dB - ) then do the same thing you did before w/ pink noise but set the output level of your console to +12dBu. Turn your amps up to the point of clipping or limiting and THEN set your limiters on DSP so that the amps just no longer clip. Now, when you drive the system hard, the DSP's limiters and drive electronics will have 8dB of headroom on which to work. The term is called complance, and is the differential on which the limiter's gain reduction algotithems can operate while the rest of the DSP's I/O remains in linear operation. The penalty is a little noise.

 

With the limiters, it's a fine line between adequate steepness and overshoot versus audible artifacts, but if you are using this primarily as protection, it's not a big deal.

 

This is good general DSP information, and understanding the interanal AND exteral gain structure of all devices in the system are a big part of troubleshooting performance issues and maximizing the full potential of any DSP product.

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