INTRODUCTION There are many different types of public address system available on the market for various requirements.

The term PA or public address means different things to different people, for example a PA rig is a common term for a high power amplification system used for performances by artists and bands. Public address and sound reinforcement are terms used for amplification systems used for other purposes. From earliest history, man has felt the need to communicate with others, both singly and in groups. It has been group communication however, that has presented the difficulties as it has been limited by the limited number of people that can be conveniently assembled within earshot of the speaker. Whenever a very large crowd had to be addressed in earlier times, the only way to do it was with relay speakers, who stood within earshot of the speaker and repeated what he had said at suitable intervals. Others within hearing distance transmitted the speech even further until it reached the edge of the crowd. These speakers were usually arranged in concentric circles, with the original speaker at the centre. There were

frequent pauses and delays, and errors would be passed on and added to. Today amplification systems enable gatherings of all sizes to be addressed with ease under varied circumstances. There is increasing demand for such facilities, from clubs and small local halls, to large sports stadia, from single speaker meetings to multi speaker conferences, for stage plays and musical events, for indoor assemblies and outdoor. The main requirements of any public address or PA system where speech is being amplified are that the programme material must be heard comfortably by all the audience or public present, and that the speech is easily intelligible. Naturalness is a desired quality and in instances where speech reinforcement is required, if everyone present can hear clearly without being aware that amplification is in use then the installation can be claimed as successful. It may not always be possible to achieve complete success in this direction especially in large gatherings; however the system should be as unobtrusive as possible but always second to providing intelligible reproduction. It may be worth discussing the difference between public address and sound reinforcement systems. The terms are somewhat self-explanatory, sound reinforcement is used to enhance the natural voices of speakers where they would otherwise be

insufficient to reach the whole audience. Examples of this type of system would include a church or place of worship and a lecture theatre. Public address however is where the public is to be addressed, in other words the audience is large and spread over a wide area, which prevents being reached with the natural voice. Often the speaker will be remote from the listener as in the case of a station announcer or factory public address system. Audio Levels:Voltage,Gain and the Decibel A basic characteristic of any audio signal is its amplitude, measured electrically in terms of voltage or acoustically in terms of sound pressure. When assessing the loudness of a signal, the amplitude or pressure is converted to a decibel value. The decibel scale gives a relative number referenced to a certain voltage or pressure. For example, 0 dBV is a popular standard reference for audio levels, and represents one volt. Note that amplitude is expressed as a voltage, while level (or loudness) is expressed using a dB scale. When working with audio electronics, levels are commonly divided into three ranges: mic level, line level, and speaker level. Mic level is the smallest signal. Microphones and other passive transducers

(devices that convert energy from one form, such as sound, to another, such as electricity) produce signals ranging from a few microvolts to a few millivolts. A typical nominal operating level for a microphone output would be -55 dBV. Line level is hundreds of times greater in voltage terms typically ranging from several millivolts up to around 1 volt, with a nominal level of 0 dBV. Speaker level is the strongest, ranging from a fraction of a volt (during quiet periods) to several dozen volts depending on the output rating of the amplifier. Of course, sound is very dynamic in nature, so whatever the nominal operating level of your signal is, if you read it with a meter during operation, you are likely to see large fluctuations from moment to moment within that range. An important function of amplifiers is providing the gain needed to raise signals from mic or line level up to speaker level. Gain is another word for amplification, and simply means an increase of the voltage or power. The opposite of gain is attenuation. Both gain and attenuation are commonly measured in decibels. The dBV scale is not the only one used for audio levels. Another popular reference scale is the dBu, where 0 dBu represents 0.775 volts. The historical predecessor to these two scales is the original dBm

scale, where 0 dBm represents one milliwatt, or 0.001 watts. Other scales you might encounter include dBW (referenced to one watt) and dBV (referenced to one microvolt). These scales are seen mostly in the radio broadcast industry. Sound Systems Working with audio means working with sound systems. Naturally, the range of sys tems available for different applications is enormous. However, all electronic audio systems are ba sed around one very simple concept: To take sound waves, convert them into an electric current and m anipulate them as desired, then convert them back into sound waves. A very simple sound system is shown in the diagram below. It is made up of two t ypes of component: .Transducer -A device which converts energy from one form into another. The two types of transducers we will deal with are microphones (which convert acoustical energy i nto electrical energy) and speakers (which convert electrical energy into acoustical energy). .Amplifier -A device which takes a signal and increases it's power (i.e. it incr eases the amplitude). 1. The process begins with a sound source (such as a human voice), which creates wa ves of sound (acoustical energy). 2. These waves are detected by a transducer (microphone), which converts them to el ectrical energy. 3. The electrical signal from the microphone is very weak, and must be fed to an am plifier before anything serious can be done with it. 4. The loudspeaker converts the electrical signal back into sound waves, which are heard by human ears. The next diagram shows a slightly more elaborate system, which includes: .Signal processors -devices and software which allow the manipulation of the sig nal in various ways. The most common processors are tonal adjusters such as bass and treble con trols. .Record and playback section -devices which convert a signal to a storage format for later reproduction. Recorders are available in many different forms, including magneti c tape, optical CD, computer hard drive, etc.

1. The audio signal from the transducer (microphone) is passed through one or more processing units, which prepare it for recording (or directly for amplification). 2. The signal is fed to a recording device for storage. 3. The stored signal is played back and fed to more processors. 4. The signal is amplified and fed to a loudspeaker. The 3-part audio model One simple way of visualising any audio system is by dividing it up into three s ections: the source(s), processor(s) and output(s). .The source is where the electronic audio signal is generated. This could be a " live" source such as a microphone or electric musical instrument, or a "playback" source such as a tape deck, CD, etc. .The processing section is where the signal is manipulated. For our purposes, we will include the amplifiers in this section. .The output section is where the signal is converted into sound waves (by loudsp eakers), so that it can be heard by humans. This portable stereo is a good example of a simple system. Sources: There are three sources -two tape machines and one radio aerial (techni cally the radio source is actually at the radio station). Processors: Includes a graphic equaliser, left/right stereo balance, and amplifi ers. Outputs: There are two speaker cabinets (one at each end), each containing two s peakers. Note that there are also two alternative outputs: A headphone socket (which drives the sma ll speakers inside a headphone set) and twin "line out" sockets (which supply a feed for an external audio system).

Now imagine a multi-kilowatt sound system used for stadium concerts. Although th is is a complex system, at it's heart are the same three sections: Sources (microphones, instruments, et c), processors and speakers. IMPEDANCE Impedance refers to the way a device reacts to the application of electric curre nt. The device will exhibit varying amounts of resistance and either capacitance or inductance. For our purposes, the resistance is most important. In keeping with common practice, when we say impeda nce we will mean resistance. Impedance, in this sense, refers to how much resistance the device presents to t he free flow of electricity through it. At a given drive voltage, the lower the impedance of the receiving d evice, the higher will be the current flow through it. This is important to know when working with amplifiers, because if the load impedance presented by the speakers is too low, it may draw so much current that the ampli fier will overwork itself and deliver distorted sound, overheat perhaps even burn out. Impedance is measured in ohms, named for Georg Ohm, who first described the set of electrical relationships now known as Ohm s Law (see fig. below). Every device will have both an input impe dance (also called the load impedance) and an output impedance (also called the source impedance). The input impedance of an amplifier could range from 600 ohms to 10,000 ohms, or even higher. A typical sp eaker impedance may range from 4 to 16 ohms.

The Ohm's Law chart remains one of the most useful formulas to use in the audio world, especially on 100v line speaker systems when calculating speaker and line loads in conjunction with an impedance meter like the ZM-104. The main parts of a PA system There are five main parts to any PA system: 1. The Ohm's Law chart remains one of the most useful formulas to use in the audio world, especially on 100v line speaker systems when calculating speaker and line loads in conjunction with an impedance meter like the ZM-104. The main parts of a PA system There are five main parts to any PA system: 1. Microphones, DI boxes and other sources 2. A mixer 3. 4. Power amplifiers Speakers 5. An arrangement of cables to interconnect the equipment The place of each of these parts in a simple system is illustrated below. (More complex systems may include additional components such asradio microphone systems, graphic equal isers, active crossoversdynamics processors outboard , and effects, as illustrated late r on this page.)

With the exception of the With the exception of the cabling, these five main par ts each have their own descriptions elsewhere on this website (follow the links in the list above), so on this page we will concentrate on how the parts are interconnected to create a complete system. Once connected, the audiosignals will follow the path indicated by the white arrows in the illustrat ion; this path is called the 'signal chain'. In some systems, some or all of the power amplifiers are internal to the mixer o r are internal to the speakers: . A mixer that includes power amplification facilities is called a powered mixer or a mixeramplifier. This arrangement restricts the distance that is possible between the mixer and the speakers, because of the need to keep speaker cables as short as possibl e. Additional separate power amplifiers are often needed, except in the very smalle st systems. . Speakers that include power amplification facilities are called active speaker s. Some systems use a mixture of active andpassive speakers, e.g. monitors active and pa ssive front-of-house vice versa). speakers (or For further details on amplifiers and speakers see the Amplifiers and Speakers p age. If you have not already obtained your equipment, or you are looking to replace i t or extend it, you might find these links useful: . PAforMusic microphone selector . Notes on choosing a mixer . Amplifer and speaker selection for required sound level Connection of microphones and instruments to the multicore Systems in which the mixer is located remotely from the stage usually employ a m ulti-core cable to provide for signal interconnections between the stage and the mixer location. This is usually referred to as the 'multicore' (or 'snake'). At one end, the multicore has a box called a'stagebox'; this end is located (surprise!) on the stage. At its other end, the multicore ha s a suitable means (described in the next section) for its connection to the mixer.

Smaller arrangements in which the mixer is located on-stage generally do not use a multicore, so for such arrangements this section can be taken to describe the connection of microphones and instruments direct to the mixer. Assuming that you have a mixer having Smaller arrangements in which the mixer is located on-stage generally do not use a multicore, so for such arrangements this section can be taken to describe the connection of microphones and instruments direct to the mixer. Assuming that you have a mixer having balanced low impedance microphone inputs ( which is the normal case) and have microphones to suit, the interconnections between the microphones and stagebox are made using balanced microphone cables. These are cables specifi cally designed to handle microphone-level female signals, and usually have a XLR conne ctor at the end which connects to the microphone and amale XLR at the other end (see the diagram below). These are screened speaker cables cables, and must not be confused with (which are also sometimes equipped with XLRs, but use unscreened cable). These microphone c ables may be extended in length simply by plugging two or more of them together.

Note that it is important that these cables are wired correctly. The electrical signal from the microphone is carried by the two cable conductorsconnected to pins 2 and 3 of th e XLRs. Although the cable would appear to work correctly if the connections to pins 2 a nd 3 were reversed (so that pin 2 at one end connects to pin 3 at the other end, and vice versa), the effect would be a reversal of phase, which is usually undesirable. The cable con ductor that is connected to pin 1 does not carry any signal - this conductor is the 'screen' (o r 'shield') of the cable, a metallic mesh or foil whose purpose is to shield the two signal-carryin g conductors from interference. For unbalanced signal sources such as direct feeds from keyboards and guitars (t hat is, where a microphone is not used), a DI box should always be used. A jack-to-jack cable is usually used to connect the unbalanced signal from the instrument (or its backline amplifier) to the DI box, which converts the signal into a balanced one suitable for travelling the distan ce to the mixer. Microphone cables as described above are then used to make the balanced signal c onnections from the DI boxes to the stagebox. Regarding which source to connect to which channel of the multicore, see the sec tion below. Connection of the multicore to the mixer The mixer end of the multicore is usually equipped with 'tails' having XLR conne ctors which may plug directly into the mixer, or which may connect to the mixer via a patch bay. An alternative arrangement, often used with patch bays, is for the mixer-end of the cable syste m to be equipped with a multi-way connector. Where a patch bay is used, short cables are used to connect from the patch bay to the mixer inputs. In systems without patch bays, it is usual for line channel 1 of the multicore t o connect to 1 of the mixer, and so on for the required number of channels. In this case, sound so urces local to the mixer (tape, CD, etc.) are connected to its higher-numbered channels. It is common practice for the drum kit microphones to be allocated to the channe ls in a 'standard' order, often starting at channel 1 of mixer. These are typically fo llowed by the other instrument channels, then the vocal mics. To enable channels to be rapidly located for adjustment during performances, it is useful to allocate the vocal channels (and sometimes instrument channels) in the same order as the band layout, left to right across the stage. If the multicore channel numbers are to correspond with the mixer channel numbers, then the sources must be connected to the stagebox in the same order that you want them t o appear on the mixer. An example channel allocation chart is shown below, for you to ada pt to your

own requirements. (For example, if your mixer has spare stereo channels then an electronic keyboard instrument could be connected to one of these.) Connection of the mixer to the power amplifiers In the case of powered speakers, their power amplification facilities are provid ed within the speaker enclosures the powered speakers. In the case of a powered mixer or a mixer-amplifierpower amplification , facilit ies are provided within the mixer unit. If no additional amplification facilities are required, t hen you can ignore this section as the connections between the mixing and power amplification parts of t hat equipment are internal to it. (In such cases, the speakers connect direct to the speaker o utputs of the mixer - see'Connection of the Power Amplifiers to the Speakers' for how to do this). When the power amplifiers are separate from the mixer, however, line-level inter connections must be made to carry the mixed signals from mixer to the power amplifiers. When the mixer is located at a good listening point, some distance in front of t he speakers, it is usual to locate the power amplifiers close to the speakers (rather than wi th the mixer),

in order to keep the speaker cables short and so avoid significant loss of power and sound quality. In such cases the signal paths from the mixer to the power amplifiers a re known as the 'returns'. The returns are often balanced circuits that are carried through the same multicore and stagebox that carries the microphone and DI box signals to the mixer, but in larger systems a dedicated returns multicore and stagebox is used. XLR connectors are generally used for the returns, but they are male rather than female, on the stagebox. Balanced XLR -XLR cables (just the same as microphone cables) are used to connect the returns from the st agebox to the power amplifier inputs. In cases where the power amplifiers are located adjacent to the mixer (such as w hen the mixer is on-stage), unbalanced connections (typically using jack connectors) are somet imes used. Between the mixer and the cables to the power amplifiers is the point at which a graphic equaliser is usually connected, to enable the tonal balance of the final mix to be tailored to the speakers and the acoustics of the room. For a stereo set-up, a graphic equaliser capable of handling both the left and right channels is required. Additional graphic equali sation may be needed for the monitor returns. In larger systems, the graphic equalisers may be connected into the returns via a patch bay. Large systems usually have separate speakers for the various audio frequency ran ges (sub-bassbassmidhigh, , , ), and each of these speakers then has its own dedicat ed power amplifier(s). This arrangement is called multi-ampingactive , and requires a uni t called ancrossover at the amplifier-end of the return cables from the mixer. T his unit splits the full-range signal from the mixer into the correct ranges of frequency to feed each amplifie r. To get the best overall result, and to avoid damaging the speakers, is very important that thecrossover frequencies of the unit are correctly set to match the range of each type of spe aker, and that the crossover output-level controls and amplifier input-level controls are also corr ectly set. Likewise, it is essential to ensure that the various outputs of the crossover are each con nected to the correct amplifier. Connection of the power amplifiers to the speakers In the case of powered speakerspower amplification , their facilities are provid ed within the

speaker enclosures. If all the speakers are of this type then you can ignore thi s section as the connections between the power amplification and sound-producing parts of the spe akers are internal to them. However, if you have slave speakers connected to the powered s peakers then what follows is equally applicable to those interconnections. These connections require the use of cables having a suitably heavy gauge of con ductors and fitted with the kind of connectors needed to suit the equipment - usually jack, XLR or Speakon connectors. Do notbe tempted to use instrument cables or microphone cables for t his purpose - the conductors in these types of cable are often quite thin and may overheat, damaging power amplifiers or even starting a fire, if subjected to the heavy currents drawn by speakers. It is always recommended to keep the cables between power amplifiers and speaker s as short as possible, to avoid unnecessary loss ofpower and to maintain a high damping fa ctor. In a multi-amped system (see the previous section), to avoid damage to the speak ers it is essential to ensure that each amplifier is connected to the correct type of spea ker (sub-bass, bass, mid etc). Taking into account all the necessary factors when deciding how amplifiers and s peakers are to be selected and interconnected can be quite a complex matter, especially in l arge systems. What's more, if you get it wrong there's a danger of damage to the amplifiers or speakers - or both. See the Amplifiers and Speakers page for further guidance on this subject. Mains power connections With the exception of microphones, passive speakers and DI boxes, most PA equipm ent requires a source of mains power. Due to the highvoltage risk involved, and the consequent of electric shock, it is essential that mains supply arrangements are made in a safe manner. Particular safety precautions are required for systems located outdoors or in ot her hazardous areas (see, for example, RCD). Additionally, it is essential to ensure that the mains power arrangements are ad equate to supply the required amount of current - particularly to the power amplifiers (and to li ghting systems, if used). For further information on safety see the Safety page. Care must be taken in the cabling arrangements to ensure that mains interference does not enter the audio signal chain. This will usually include: . Keeping the mains cabling (whether for PA or not) spaced as far as possible fr om signal cabling, especially on long cable runs. . Powering the PA equipment from a separate mains outlet(s) (preferably a separa te mains distribution circuit) to those used by interfering equipment (such as ligh ting).

. Using the same phase for all PA equipment (including any on-stage equipment), and running interfering equipment from a different phase (if the supply capacity is adequate for this arrangement). . Using balanced audio lines wherever possible. . Keeping devices that contain audio transformers e.g.passive DI boxes microphon e (and splitters) away from mains-powered equipment, especially amplifiers and video mo nitors. System Arrangements From a PA perspective, there are two distinct arrangements of systems used - tho ugh the real difference lies not in the nature of the system itself, but in by whom it i s operated. The two arrangements are: . When the person operating the system is alsoa performer in the band. In this arrangement, typically adopted by bands of up to 4 members when performing in small venues, the mixer mixer-amplifer(or ) is located on-stage and the responsi ble

person will set up the system prior to the rehearsal, probably making small adju stments during the rehearsal and possibly during the performance. This is something of an "economy" arrangement, as although it has the advantage of avoiding the costs associated with another person, it suffers from the problem that even if a reaso nable mix can be achieved initially, the responsible person cannot properly manage the sound throughout the performance to take account of dynamic factors such as a growing or increasingly noisy audience. (Even the initial entry of the audience can signifi cantly affect room acoustics from those which existed when the system was set up.) This is not necessarily a matter of skill, it's simply that a person who is located behi ndthe frontof-house speakers and is primarily focussed on delivering a good performance is at a significant disadvantage in comparison with someone in a good listening position who is concentrating onlyon the sound. Nevertheless, how much this really matters depen ds to a great deal on the acoustic expectations of the audience.

. . . When the person operating the system is nota performer in the band. In this arrangement, typically adopted by larger bands and in large venues, the mixer is located at a suitable listening point (preferably central to the audience) and i s set up and operated throughout the performance by one or more people dedicated to this task. Whilst this arrangement has the potential to provide a technically superior soun d (which means, at the end of the day, more pleasing to the audience), it has its own pro blems in that it inevitably creates something of a divide between the performers and t he sound engineer(s). To minimise this kind of problem it is essential that the performer s have full confidence in the engineers, and see them as their essential allies. This w ill often mean that the performers will need to explain to the engineers what kind of soun d they are hoping for, both on-stage and front-of-house; this will usually be an intera ctive trial-and-error process, requiring patience and respect by both parties. In some cases the performers may need to accept that their ideal sound is not achievable, give n the limitations of the available PA equipment and of the venue. Microphone Technique Using a sound system is something of an art form in itself, and excellence can o nly be achieved through experience. Nevertheless, it is hoped that the following brief notes wil l provide some useful hints for performers, without getting too technical. You might also find it helpful to read theIntroduction for Mixing Engineers, to give you some idea of what theyare up a gainst! . You can help to avoid feedback by the correct use of microphones. Do not point them at the monitors, and avoid wandering in front of the front-of-house speakers if at all possible. Maintain a reasonably constant distance from the microphone (except wh en deliberately varying the distance for effect), as advised by the sound engineer( s). Do not wrap your fingers around the basket of the microphone, as this is likely to decr ease the microphone's in-built immunity to feedback. . . . If feedback does occur, the natural reaction is to increase your distance from the microphone. But whether this is the right thing to do depends upon how close you were to it at the time. If you were more than about 4 inches (10 cm) away, by moving further away you are actually increasingthe chance of feedback, because the engineer mus

t provide even more amplification to achieve the desired sound level for the audie nce, and this would make the feedback worse. So decreasethe distance, so that the amplifi cation required can be reduced. If however you were very close (less than 1.5 inches (4 cm) ), moving slightlyfurther away may be helpful because, with some types of microp hone, being very close can decrease the microphone's in-built immunity to feedback. . . . . The following illustrations show some possible positions for a vocal microphone, with associated comments, listed according to the angle of the mic. They assume the t ype of mic most commonly used for live performances (as opposed to studio recording), i .e. an end-firingtype with a cardioid or super-cardioid pick-up pattern.

. . 1. A rarely-used position, generally employed only for situations such as a stand-mounted mic used by a drummer. (In such a situation, the mouthtomic distance must be kept fairly small, to avoid excessive pick-up of the drum kit.) . . 2. A hand-held position sometimes briefly used during a performance, for visual effect. Good pick-up, with a considerable amount of bass boost (due to the proximity effect). . . 3. A position commonly used by some rock and pop artists. As this position gives a very large amount of bass boost, it requires a type of microphone that is intended for such close-up use (such as the Shure SM58). Even quite small changes in the distance to the mic will have a marked effect on the amount of bass boost, and also on the amount of pick-up (unless heavy compression is in use). Not suitable for applications where facial expression is important, or when close-zoom camera work is in use, as the performer's mouth is completely obscured (except for profile camera angles). . . 4. Similar to 3, but with less bass boost and less sensitivity to changes in distance. Slightly better visibility of the performer's mouth, giving an improvement for semi-profile and profile viewing angles. . . 5. Even less bass boost and less sensitivity to changes in distance. However, reduced pick-up of the performer (especially with cardioid mics) versus other sound sources may be a problem when stage sound levels are high, or when the performer is in close proximity to loud instruments. (This is more of a problem when the performer has a quiet voice.) Better visibility of the performer's mouth for profile and semi-profile camera angles. . . 6. Often a good mic position, but the performer's mouth is still largely obscured for frontal views. . . 7. Often an ideal mic position when a certain amount of bass boost is desirable. Much improved mouth visibility from all angles. . . 8. An ideal mic position for strong voices, when bass boost is not required and stage sound levels are not too high. Quiet voices may be difficult to amplify sufficiently, without feedback and/or pick-up of other sounds. The low sensitivity to changes in distance make this position very suitable for stand-mounted mics. Excellent mouth visibility from most angles.

. . 9. & 10. Useful positions when some bass boost is not a problem and good immunity to other stage sounds is required. They also provide reasonably good visibility of the mouth (especially 10.). . . . . 11. An incorrect technique, giving rather poor pick-up of the vocals (particularly with super-cardioid andhyper-cardioid mics). The sound engineer may be able to compensate to some degree. . . 12. The performer seems to think that this is a side-addressed microphone. As it isn't, the pick-up of the vocals will be extremely poor. The sound engineer will not be able to compensate for this; attempts to do so are likely to result in feedback and/or excessive pick-up of other sound sources and room reverberation. . For a slightly more technical discussion on the effect of microphone technique on the pick-up behaviour of microphones, see Use of Microphones on the Microphones page. . . . If you are using a hand-held radio microphone with an aerial that sticks out o f the bottom end of the mic, take care not to hold the microphone around that part, as this c an cause problems with the pick-up of the radio signal by the receiver. . . . . . If you are using a hand-held microphone that has been marked with one or more identifying coloured bands of tape, it would be of assistance to the sound engin eer if you hold the microphone in a manner that keeps at least one of these bands clearly v isible. Musicians . Guitarists: To avoid loud crackles and bangs that are very unpleasant to your audience (and potentially damaging to the PA equipment), then unless your guitar cable ha s a self-shorting jack plug, or you have first used a pedal (or other means) to turn down or cut off your guitar signal, never unplug your guitar cable at the guitar end wit hout being

sure that the engineer has muted your channel at the mixing desk. If you are rel ying on the engineer to do this, then they will have to know when you are about to unplu g; this could, for example, be communicated either by pre-arrangement (e.g. after a spec ific song), or through use of agreed hand-signals. Note, however, that muting at the mixing desk will not usually avoid such sounds from your backline speakers. MICROPHONE Introduction A microphone ('mic' for short, pronounced "mike") is a device for converting aud ible sound into a signal. To accomplish this task with the optimum efficiency and quality of resul t requires a type of mic that is appropriate to the particular situation, so there are many differ ent types of mics some designed for very specific applications and others that are more general pu rpose. "When you have exhausted all possibilities, remember this: you haven't." - Thoma s Edison. Types of Microphone Mics can be categorised in several different ways. The most important of these c ategories are described below. Dynamic or Condenser All types of microphone incorporate some form of diaphragm - this is a small thi n surface which vibrates in sympathy with the sound pressure waves reaching the microphone . However, dynamic and condenser mics vary in how these vibrations are used to produce an e lectrical signal. . In a dynamic mic, sound is converted to an electrical signal by the vibrations of the diaphragm causing the vibration of a coil in a magnetic field - effectively an e lectrical generator on a very small scale. (This is the exact opposite of the operation of a speaker driver.) As this produces sufficient signal level for direct connection to a PA system, no amplification of the signal is required within the mic. Dynamic mics are most us eful for close-proximity applications (i.e. 0 to 15 cm) such as lead vocals, guitar ampli fiers, etc. The sensitivityof low impedance dynamic mics is typically in the region of 1 to 3 mV/Pa. (Impedance is explained later on this page.) . In a condenser mic (also called a capacitor mic), sound is converted to an ele ctrical

signal by the vibrations of the diaphragm causing changes in the capacitance of a charged capacitor. This is achieved by the diaphragm itself being one of the pla tes of the capacitor. As this produces a very small signal level, some initial ampli fication of the signal is required within the mic itself. This internal amplifier may be powered either by an internal battery or by power supplied from the mixer (usually at 48 volts d.c.). The latter arrangement is calledphantom powering and is only possible whe n using a balanced connection between the mic and the PA system. (For mic connecti on types see Low-impedance or High-impedance.) Unfortunately the amplifier inevitab ly

. Sub-cardioid mics have a very gradually reducing sensitivity from the front to the back, maintaining some sensitivity at the back. introduces some noise into the mic signal - see theMicrophone Noise Levels secti on below. Condenser mics are most useful for larger distances between the sound sou rce and the mic (i.e.15 cm upwards), such are encountered with lecterns and with ove rhead miking of drum kits, choirs, theatre stages etc. They can be more prone than dyn amic mics to making a "popping" sound when used close-up with a "breathy" sound sourc e such as a voice or a wind instrument, though this problem can be reduced with a windshield. They are generally more fragile than dynamic mics, so are rarely emp loyed for rough stage use or in very high SPL applications. They are however capable o f a higher quality sound than dynamic mics, and the best versions are therefore exte nsively used in studio recording work. The sensitivity of low impedance condenser mics i s typically in the region of 3 to 20 mV/Pa. Omni-directional or Uni-directional . Omni-directional mics pick up sound with equal sensitivity from all directions . This is not normally useful for PA work, because in PA work each mic is targetted at a singl e sound source (so that the amplification given to that sound can be controlled separate ly from others, and so that pick-up of unwanted sounds can be minimised). Their applicat ion is generally limited to recording work (particularly of ambient sounds) and to soun d-level measurement. . Uni-directional mics pick up sound with greater sensitivity from the front tha n from other directions. There are several variations on this theme. Each of the follow ing types is illustrated with a polar response diagram, in which increased sensitivi ty in a particular direction is indicated by the line on the diagram being closer to the outer circle. Imagine the microphone diaphragm being located at the centre of the circle, with the most sensitive end (or side) of the microphone facing towards the top of the cir cle. So, the upper-most point of the line on each diagram indicates the sensitivity at th e front of the microphone, or at '0 degrees' - i.e.on-axis the response, and the lower-m ost point indicates the sensitivity at the back, or at '180 degrees'. (The diagrams below are simplified to illustrate typical mid-frequency responses; in practice the polar responses vary with frequency, so check the manufacturer's specifications.) . .

Cardioid mics have a gradually reducing sensitivity from the front to the back, with very little sensitivity at the back. .

. Super-cardioid mics reduce their sensitivity from the front to the sides at a faster rate than cardioid types, reaching a minimum sensitivity at an angle of around 120-140, measured from the front. The sensitivity then increases again towards the back, but the sensitivity at the back is still very much less than at the front. . Hyper-cardioid mics provide even less sensitivity at the sides than do super-cardioid types, at the expense of a little more sensitivity at the back. Therefore, a . Hyper-cardioid mics provide even less sensitivity at the sides than do super-cardioid types, at the expense of a little more sensitivity at the back. Therefore, a monitorspeaker should never be placed directly behind this type of mic. Their minimum sensitivity is at an angle of around 100-120, measured from the front. . 'Rifle' or 'shotgun' mics are the most directional type, so-called because of their long rifle-like barrels. They are generally used only for long-distance miking (more than 2 metres from the source), e.g.for theatrical work, and should be located such that the back of the mic is not exposed to unwanted sounds. . Bi-directional types . Although not featured in the title of this sub-section (as they are rarely use d in live PA work), bi-directional mics get a mention here for completeness. They pick up sound with equal sensitivity from two opposite directions, shown in the diagram as the front and back; in practice however, as these are usuallyside-addressed type s, the sensitive directions are most commonly on two of its sides. Low-impedance or High-impedance . Low-impedance mics have an impedance ohms of from around 50 to 600 and may onl y be connected to low-impedance mic inputs. They come in two varieties, each of wh ich should only be connected to the corresponding variety of low-impedance mic input : . Unbalanced, where one of the two signal-carrying conductors of the interconnection between the mic and the system is also thesignal earth conductor of the interconnection, usually provided by the screen cable(s)of the . With this arrangement, the cables are prone to pick-up of interference from stra y magnetic fields, earth loops and radio signals, and so these types of mic are suitable only for use with moderate lengths of cable (up to around 10 metres). They are relatively uncommon. . . . .

. Balanced, where the two signal-carrying conductors of the interconnection are separate conductors from the signal earth (cable screen) of the interconnection. This arrangement is highly immune to pick-up of interference, and so may be used with very long lengths of cable (up to 200 metres), provided it is of good quality. Also, balanced connections allow the use of phantom powering. Nearly all professional and semi-professional mics are of this type. See the diagram fo r connection arrangements.

. High-impedance mics have an impedance very much greater than 600 ohms - usuall y in the range of 5,000 to 15,000 ohms (5 kilohms to 15 kilohms). They may only be connected to high-impedance mic inputs. Such inputs are rare in PA systems, as t hey may only be used with short cables (less than 5 metres) if the signal is not to suffer from a reduction in high audio frequencies (treble), resulting in a loss of clarity. To enable a high-impedance mic to be connected to a low-impedance input, or vice versa, a microphone matching transformer can be used. To minimise loss of signal quality, it is important to use a good quality transformer and to locate it so as to minimise the length of the high impedance cable run. To avoid pick-up of hum by the transformer, do not locate i t close to mains-powered equipment. Boundary or Conventional . A boundary mic is a special type which when placed on a surface utilises the s ound energy collected at that surface to provide a greatersensitivity (and therefore, potentially, a better signal-to-noise ratio). Many such mics are generally equally sensitive to sounds in all directions above the surface (a so-called 'half-omni' response pattern) a nd most are condenser types. Typically used for speech, where a convenient surface such as a desk or lectern is available, though some types have a 'built-in' plate to act as the surface. Also known as a 'pressure-zone microphone' (PZM), which is a trademarke d name. . By "conventional" here, we just mean "not a boundary mic". Wired or Radio . Wired mics connect to the PA system by means of a cable. The cable usually att aches to the mic by means of a 3-pole XLR connector. . Radio (or 'wireless') mics contain a battery-powered radio transmitter. The ra dio signal from this transmitter is picked up by a receiver which is connected to the PA sy stem. The mic and the receiver are purchased as a pair and are referred to as a "radio mic system". . Most radio mic systems use a frequency-modulated VHF UHF (FM) radio signal at or frequencies, or in the 2.4 GHz bandfrequency . (Strictly, each system does not o perate at a single frequency, but rather uses a narrow range of frequencies called a 'c hannel'. The frequency that is quoted is the carrier frequency, the frequency at the cent re of the channel.) The frequencies used are either "licensed" or "de-regulated". Use of a licensed frequency requires payment of an annual license fee. Use of the de-regu lated frequencies is free, but as their use is uncontrolled by licensing it is more li

kely that interference will be experienced from other users. Warning:Some UHF systems will allow you set the operating frequency to a value outside the legal ranges, but t his is clearly very inadvisable and may incur severe penalties. All systems, regardl ess of whether licensed or de-regulated frequencies are used, must comply with the appropriate standards. In particular, these standards put limits on the maximum power output of the transmitters and on the maximum levels of spurious frequencies tha t may be radiated. . For VHF systems in the UK there are: 5 de-regulated frequencies, 6 frequencies that are licenced for single-site use (in this case it is the site that is licensed, not the equipment) and 15 frequencies that are licenced for any-site use. . The frequencies are listed below. . . D . 173.8, 174.1, 174.5, 174.8 and 175.0 MHz

e r e g u l a t e d .( M P T 1 3 4 5 / 1 3 1 1 .e q u i p m e n t ) .S i n gl e s i t e l i . 176.4, 177.0, 192.3, 200.1, 207.7 and 208.1 MHz

. . . . 175.25, 175.525, 176.6, 191.9, 192.8, 193.0, 199.7, 200.3, 200.6, 208.3, 208.6, 209.0, 216.1, 216.6 and 216.8 MHz . c e n c e d ( M P T 1 3 5 0 e q u i p m e n t ) A n y s i t e l i c e n c e d ( M P

T 1 3

5 0 . e q u i p m e n t ) . . Many UHF systems are tunable, i.e.can be adjusted to operate on one of several frequencies. UHF systems are generally better than VHF ones, as there is less radio-frequency interference around at UHF. . . Most UHF systems in the UK currently operate at frequencies just above the highest-frequency UHF terrestrial TV broadcast channel (channel 68, which ends at 854.0 MHz). As the UHF TV channels each occupy a frequency slot 8 MHz wide, the same numbering convention is continued (i.e.69, 70, etc.) for each further 8 MHz band above the range used for TV, (i.e.above 854.0 MHz). . Channel 69 is currentlyallocated as a regulated band, within which there 14 specified licenced frequencies for UHF PMSEradio systems. However, in the UK PMSE usage of channel 69 is in the process of being migrated to channel 38 (606 to 614 MHz) as part of the move to clear the socalled '800 MHz band' (790 to 862 MHz) for other services; access to channel 69 is currently intended to cease on 1st July 2012. Channel 38 is expected to accommodate at least 8 simultaneous system frequencies without mutual interference (depending on the equipment specifications). The lower part of channel 70 is allocated as a de-regulated band that is wide enough to accommodate 8 or 9 UHF radio system channels. This band is sometimes referred to as '863 to 865' (its approximate frequency range), or as the ISM ETSor band. Its use for PMSE applications remains unaffected by the move to clear the so-called '800 MHz band' (790 to 862 MHz) for other services throughout much of Europe. Single-site licences may also be granted for use on frequencies withinthe range used for TV broadcasting, i.e.470.0 to 854.0 MHz;in the UK these are mostly within channels 67 and 68 (838 to 854 MHz). This is possible because within the area of each broadcast TV transmitter there are channels that have to remain unused for TV broadcasting, in order to avoid interference with TV transmissions in adjoining areas and for other technical reasons - whereas controlled use of radio systems on specific frequencies within some of these channels may be judged not to be detrimental to TV broadcasts, because of the (relatively) very low transmitted power level of radio systems and the narrow bandwidth they employ. After the changes currently in progress in the UK, TV broadcasts

will occupy only channels 30 and 39-60, and such 'overlap' licensing is expected to be available within channels 39 and 40 (614 to 630 MHz). . The channel 69 and 70 frequencies are listed below. (The frequencies and . . . . . . 863.1 to 864.9 MHz. . . Frequencies commonly used are 863.1, 863.5, 863.7, 864.1, 864.3 and 864.9 MHz precise licensing arrangements for channel 38 (and 39-40) are not yet finalised. ) Licencedc h a n n e l 6 9 ( M P T 1 3 5 0 e q u i p m e n t ) . 854.900, 855.275, 855.900, 856.175, 856.575, 857.625, 857.950, 858.200, 858.650, 860.400, 860.900, 861.200, 861.550 and 861.750 MHz Allow at least 0.2 MHz between systems (see the note below on simultaneously operated systems). Deregul

at ed . . . c h a n n e l 7 0 ( p a r t o f ) ( E N 3 0 0 2 2 0 e q u i p m e n t ) . . Systems operating on the 2.4 GHz band are now popular where a large number of systems (e.g.more than 20) are to be used simultaneously. Furthermore, in some countries certain UHF frequencies are being withdrawn for radio

microphone use and so this band may be the only practicable option. However, as it is a de-regulated band, care must be taken to select frequencies that are not

subject to interference from other types of equipment operating on this band in the vicinity. Some systems utilise digitally-coded transmission, which can assis t in the avoidance of interference from other equipment.

between 60 and 90 degrees to each other. Some receivers provide an audio output intended for connection to a line input o f the PA system, whilst others have outputs intended for connection to a mic input. Some types may provide both kinds of output, or a single output of adjustable level. When several radio mics need to be operated simultaneously, each system must be set to a different frequency. Furthermore, in order to avoid intermodulation int erference between the systems, the frequencies selected must be chosen from a compatible set for the particular make and type of system being used. The maximum number of frequencies in a compatible set will depend upon the quality of the system. For example, in the case of Sennheiser's UHF systems: . The eW100 G1 and G2 systems allow a maximum of four simultaneous frequencies - a compatible set in the UK de-regulated band is 863.1, 863.5, 864.3 and 864.9 MHz. . The eW100 G3 system allows a maximum of six simultaneous frequencies - a compatible set in the UK de-regulated band is 863.1, 863.4, 863.75, 864.225, 864.550 and 864.975 MHz. . The eW300 range allows up to eight simultaneous frequencies and the eW500 range up to 20. . The only four frequencies available on the freePORT system (frequency range E) are 863.1, 863.7, 864.1 and 864.9 MHz; these can be used simultaneously. Consult the appropriate manufacturer's information for the details relevant to y our specific system(s). It is very inadvisable to simultaneously operate systems fro m different manufacturers, or even different product ranges from the same manufact urer - except that there is unlikely to be any interference between good quality syst ems operating in entirely different bands - i.e. between VHF, UHF and 2.4 GHz system s. Hand-held or Hands-free . Hand-held mics are generally about 6 to 7 inches (15 to 18 cm) long and 1.25 i nches (3 cm) in diameter. They may be held in the hand or placed in a clip on a mic stand . Note that many radio mics have a slightly larger diameter than wired types, and there fore will not fit into 'standard' sized mic clips. Mic clips fix to the stand by mean s of a screw thread, of which there are three common types: . 5/8 inch 27 turns per inch (a large diameter fine thread) - sometimes referred to

. . as an 'American thread'. 1/2 inch (a medium diameter fine thread) - less commonly encountered. 3/8 inch Whitworth (a small diameter coarse thread) - sometimes referred to as a 'Euro thread'.

. In case your mic clip doesn't fit your stand, thread adaptors are available. . Hands-free mics are generally much smaller and are either body-worn (e.g.clipp ed to a lapel or tie, or attached to a head-set) or are suspended by their cable (e.g. above a choir). For theatrical applications, they are often hidden in the hair (or wig ). Bodyworn mics are usually of the radio type, and are used in conjunction with a body pack. A body-worn mic worn on the chest is also known as a lavalier mic. Many types ca n be purchased with either an omni-directional or uni-directional pick-up pattern. An omnidirectional pattern can often be a good choice for chest or lapel-worn mics, pro vided that acoustic feedback is not likely to be a problem (e.g.for recording or broad cast applications, or where the mic is always placed high on the chest and the user h as a strong voice). This is because these types are less susceptible to changes in pi ck-up level due to head movements, and are less likely to pick up unwanted sounds due to friction with clothing. However, in live PA situations where feedback may be a p roblem, or where pick-up of ambient sound needs to be minimised, the uni-directional typ es can be appropriate provided that they are worn at the correct angle and that hea d movements relative to the body are fairly small; these types typically have a ca rdioid pick-up pattern. Use of Microphones To get the best results, it is important to choose an appropriate type of mic fo r the job, and to use it correctly. For guidance on choosing a suitable microphone, see the Mic rophone Selector. The 'correct' use of microphones is a huge subject in itself, and engi neers have their own differing opinions on which techniques give the 'best' results under v arious different circumstances. Performers (especially vocalists) may also have their own preferr ed microphone technique, sometimes without realising the effect that this has on the amplified (or recorded) sound (seeMicrophone Technique Getting Started - for Performers page on the and also the paragraphs below). One thing that everyone agrees on, though, is that the distance between a microp hone and the sound source that it is meant to pick-up is a hugely important factor. This is d ue to at least three major reasons: 1. Proximity Effect Most PA mics are uni-directional types, and all uni-directional mics exhibit wha t is known as the "proximity effect". The result of this effect is that sounds which are made very close to the mic are picked up with a greater bass response than sounds which are made furthe

r away. This is most important for presenters and vocalists to understand, because the d ifference that a change in working distance makes to the sound of their voice can be quite dram atic. It is especially significant for deep-voiced vocalists (usually male), because a great er proportion of their voice is in the frequency range which is subject to the proximity effec t. At a working distance of greater than about 4 to 6 inches (10 to 15 cm), the proximity effect can be ignored. As the distance decreases from this down to zero, the amount of bass emphasis in creases. 2. Unwanted Pick-up of Ambience, Leakage and Feedback Just like an ear, a microphone will pick up sounds that originate close to it mo re readily than sounds that originate further away (simply because of the dispersion Inverse of sound - see square law). Therefore, if a microphone is placed a large distance from the soun d source that it is intended to pick up, its electrical output level (resulting from that sour ce) is likely to be very low, and so a large amount of amplification (gain)will have to be applied to the electrical signal that it produces. This same amount of amplification will also be applied to soun ds that it was not

intended to pick up, such as unwanted room ambience, sounds from other instrumen ts and/or vocals ('leakage'), and sound from the PA speakers (both front-of-house monitors and ) which may result in an over-resonant amplified sound or in acoustic feedback. This problem can be partially addressed by the use of a uni-directional micropho ne, placed and directed so that its direction of maximum pick-up is towards the wanted soun d(s) and its direction(s) of minimum pick-up towards the most troublesome unwanted sounds (se e thepolar response patterns). Sometimes it can also be partially addressed by the use of e qualisation on the picked-up signal, to provide some discrimination in favour of the frequency spectrum of the wanted sound and against that of the unwanted sound(s). However, the most effective method of controlling the problem of unwanted sound pick-up is usually to place the microphone as close as reasonably possible to the wanted so und source and directed towards it - bearing in mind the proximity effect (see above) and t he 'variable distance' factor (see below) - so reducing the amount of amplification that is n ecessary. Also, where practicable, unwanted sound sources should be kept as far as possibl e from that microphone and should not be directed towards it. 3. Level Changes With Variable Working Distance When the distance between a microphone and the sound source that it is intended to pick up is variable, as in the case of most lead vocals microphones, there is another fa ctor to take into account besides the changing proximity effect. This is that the effect on the mi crophone's ouput level of changing the working distance by a given amount (say, 2 cm) depends on what the distance was to start with. To explain this, we need to consider that the output level increases by 6 dB for every halving in working distance (the inverse square law). For example, consider a stand microphone that is 4 cm from a vocalist's mouth. I f he/she then moves 2 cm closer to the mic then the distance will have been halved so the output level will increase by 6 dB, which is very significant. (In addition, there will of co urse usually be a considerable change in the proximity effect.) Now compare this with a starting d istance of 10 cm, and again reduce that distance by 2 cm. In this case, the working distanc e will only have been reduced by a factor of 0.8, resulting in a level increase of only abou t 2 dB from the microphone. So, it can be seen that a microphone that is very close to a sound source is ver y sensitive to changes in working distance, while one further away is much less sensitive to su ch changes.

This partly explains why compression is so often used on close-miked lead vocals . Application When considering how best to apply this information, it is important to take int o account the microphone technique of the vocalist (seeMicrophone Technique on the Getting Sta rted - for Performers page). Unless there is close supervision, or a physical barrier such as a separatelymounted pop screen (both of which are only likely to apply in a studio setting), the vocalist may at any time choose to vary the working distance between several 10's of cm and z ero, as well as varying the loudness of their voice. These variations may be made deliberatel y, or to some extent unintentionally; in any case the result will be changes in the picked-up vocal level. Provided that, in the overall sound mix, the combined effect of such changes (ta king into account any compression applied) is what the vocalist intended, and provided tha t the mic preamplifier gain is set so as to avoid distortion at the maximum output level that will be obtained from the mic as the changes occur, then all is well.

But otherwise, substantial changes in working distance can be problematic for th e sound engineer and so should be avoided. Compression only goes part-way towards addres sing this, as it does not compensate for the resulting changes in proximity effect, nor for the increased pick-up of unwanted sounds that occurs when decreased mic output level causes an increase in the gain applied by the compressor (or indeed manually by the sound engineer). Care of Microphones Mics contain delicate precision-engineered components, and if you want your mics to continue to perform as well as when they were new, you must look after them very carefull y. Even ruggedised stage mics will benefit from careful treatment. Following these simpl e do's and don'ts will help considerably: . DO keep them in padded protective boxes or pouches - preferably individually when not in use (especially during transport). . DO clean the integral windshield from time to time, when this is accessible. F ollow the maker's instructions (especially for expensive mics!), but in the absence of any instructions the wire-meshed end of most types can be unscrewed and then gently washed in warm soapy water - allow to dry thoroughly before re-attaching to the main part of the mic. . DON'T check them by tapping them or by blowing into them - speak (or sing) int o them instead, and educate users to do the same. . DON'T drop them or allow them to be subjected to other sudden shocks. . DON'T store them in damp conditions or expose them to extremes of temperature. . DON'T expose a microphone to sound levels exceeding its specified maximum SPL at the very least this will give a distorted pick-up of sound and may even damag e the microphone. Microphone Selector The purpose of this information is to give you a general guide as to the most po pular mics for a given application. As there are hundreds of mics available, from many different manufacturers, it would not be practical to try and list them all. Therefore, only the most pop ular manufacturers are listed. If purchasing a UHF radio (wireless) microphone system that you wish to use in a regulated (licensed) frequency band in the UK, then be sure to take account of the fact th at the present licensed band will become unavailable during 2012 (precise dates dependent upon location of use). Furthermore, the replacement band is not yet available in some areas, and

may not be until some date in 2012. Most current systems are unable to operate in both the old and new bands. For further details see Wired or Radio. Note that some mics are designed for a very specific use, whilst others are of m ore general application. It doesn't follow that just because a mic is very expensive that it must be either very specific in application, or that it must be very general purpose! Neither does i t follow that just because a mic is very specific in application, or very general purpose, that it must provide very high performance. As circumstances vary from use to use, before buying a particu lar mic it is usually advisable to check with the supplier that it is a good choice for your p articular situation. The listed microphones are arranged in 'price bands' according to the table belo w. These bands are intended to give an approximate guide as to what you might actually pay on-l ine (not the

manufacturer's R.R.P., which is generally considerably higher); the figures incl ude UK VAT at 17.5%. Actual prices may vary significantly from supplier to supplier and can ch ange from week to week - it definitely pays to shop around. Remember though, when buying to a tight budget, that you get what you pay for an d, in general, the price band shown for each mic can be taken as a rough guide to the quality t o be expected - when comparing like with like. However, note that some users may prefer the so und (or other characteristics) of particular mics, as compared to more expensive models. Note also that mics that are listed for the same application may not be directly comparable - e.g. s ome may be large stand-mounted types while others may be miniature clip-on types, and such factors can also influence the price. Sorry, these mics are not on sale from PAforMusic. Reverberation Often known as 'reverb', this is an effect unit which simulates the ability of a room to cause a sound to die away slowly when the source of the sound ceases abruptly. Reverb units are useful in reducing the 'dryness' of a sound. As rooms differ in the manner and degree to which they exhibit this effect, such effect units usually provide some control over the type and extent of the reverb effect which they produce. The more sophisticated digital units now available generally allow selection fro m a number of reverb types, for example types simulating various different sizes of room and t ypes simulating the old analogue reverb effects such as spring-line and plate units. Gated reverb is an effect which has the facility to automatically cut off the re verberation effect when the input levelsignal falls below a specific . Echo This is an effect unit which simulates a natural echoing of the sound, or which provides an artificial effect of a similar nature. Most units have the ability to provide a single echo or multiple echoes. See also Delay. Delay Delay is another name for an echo unit, because an echo is a delayed (and, usual ly, somewhat modified) copy of the original sound. Such a unit may also be used to provide a delayed version of a signal to 'secondary' speakers which are situated some distance in front of the main speakers (in a very large hall, or outdoors); this enables the sound heard from the secondary speakers to be synchronised with the sound heard from the main speakers - the la

tter having been delayed in travelling through the air to reach the location of the secondar y speakers. Approximately 30 milliseconds of delay is required per 10 metres of distance bet ween the main and secondary speakers (seeSpeed of sound). Chorus This is an effect unit which modifies a signal in such as manner as to simulate the presence of several additional sources of the same (or similar) sound, all operating in u nison - as in "a chorus of voices".

Phase This is an effect that is sometimes used with guitars to improve the 'interest' of the sound. It may be adjusted to give a wide range of effects, a common one being a slow 'sweeping ' sound. Flange This is similar to a more extreme version of the phase effect; a rather harsh 's weeping' effect used with electric guitars to to help give a 'heavy metal' type of sound, and fo r other 'special effects' purposes. (It gained its name from the fact that it was originally prod uced by mixing the sound with a tape-recorded version of it that was slowed down by means of fricti on applied to the flanges of the tape spools.) Distortion Generally used only with electric guitars, an effect that simulates a distorting guitar amplifier. Previously known as 'fuzz'. Overdrive Generally used only with electric guitars, a more subtle version of the distorti on effect, less rich in the higher harmonics. It simulates anoverdriven guitar amplifier. Types of Speakers First a point of clarification. On these pages, when we talk about 'speakers' we always mean the complete enclosures (or cabinets) containing the drivers (that is, the units tha t actually make the sound) along with any other associated parts such as crossovers and protection c omponents. We never mean just the drivers. Speakers can be categorised into the following four types, according to their us e. Each of these types may be unpowered (that is, require external amplifier(s) to power it) or p owered (that is, incorporate its own amplifier(s) within the enclosure). . Backline . Intended for reproduction of the sound of a single instrument (usually a guita r, electric bass or keyboard) for the benefit of the musician playing it. When the speaker i s powered then the complete unit is called a combo, because the amplifier and spea ker are combined. When the amplifier is separate from the speaker then the amplifier is called a head, because it is usually placed on top of the speaker. (The amplifie rs used for backline speakers, whether integral or separate, are different from the ampl ifiers used for the other three types of speakers, in that backline amplifiers provide preamplification facilities, to allow the direct connection of the low-level signal

that is usually obtained from an instrument. The exact nature of these pre-amp facilities is dep endent upon the type of instrument that the amplifier is intended for, but will usually include some form of equalisation.) Backline speakers vary according to the type of inst rument that the speaker is intended for, because different instruments produce sounds with different frequency characteristics. Guitar speakers emphasise the mid-rang e frequencies, bass speakers are specially designed to handle bass-dominant freque ncies

and keyboard speakers usually have a frequency range approaching that of a full range speaker.

. Front-of-House - Full Range (Tops) . Usually consists of one or more woofers (bass drivers) and one or more horns ( treble, or HF drivers) in the same enclosure. When powered, (i.e.contains its own amplif ication within the same enclosure), it may contain a single amplifier to drive both the woofer(s) andhorn(s), or it may contain separate amplifiers for the woofer(s) and horn(s) (called a bi-amped speaker). Although described as 'full range', these speakers are usuall y limited in their deep-bass response - typically 3 dB down at some point in the region of 45 to 100 Hz. Therefore bass bins (subs) are often additionally required, to cover fre quencies below that point, and the 'full-range' speakers are then referred to as 'tops' b ecause they are commonly situated on top of the bass bins - either placed directly or on pol es or stands. . Front-of-House - Bass Bins (Subs) . Consists only of one or more woofers, covering the sub-bass frequencies (below the frequencies covered by the tops). These speakers are usually driven by their own separate amplifier(s), which are supplied with the sub-bass frequencies by an ac tive crossover. This crossover also protects the tops from the sub-bass frequencies. As subbass frequencies are essentially non-directional, the bass-bins are usually oper ated in mono mode - even when the tops operate in stereo. . Monitor . Intended for the benefit of the performers, these are generally either of a mo derately powerful floor-standing wedge-shaped design, or are low-powered stand-mounted units. On large stages, much larger side-fill (or cross-fill) speakers are somet imes used. Increasingly,in-ear monitoring (IEM) is being used in place of monitor speakers. Power Ratings The quoted power handling capacity of a speaker, measured in watts (abbreviated "W"), is a figure to be treated with some caution. This is for two main reasons: Firstly, it tells you only how much electrical power the speaker is able to acco mmodate without sustaining physical damage and/or causing undue distortion of the sound. This gi ves only a very rough guide as to the sound level (volume) that it is capable of producing, because the sound level obtained from a given number of watts of power depends upon the sens itivity of the Secondly, because of the fluctuating power level of most sound signals, the prop er specification speaker, which varies significantly between different models. of power handling capacity requires a standardised way of measuring the power le vel that a particular speaker can accommodate. Unfortunately there are several different me

thods of measurement in use, so when looking at the power rating quoted for speakers you need to be certain of which method is being used, and compare only "like with like". The mo st common methods are: . RMS or "continuous" power. This refers to a sustained average level of power, an d is the most useful figure. (The term "RMS" is strictly inappropriate here, thoug h its use is very widespread. The term is used because it refers to the power calculat ed by multiplying the RMS value of the voltage by the RMS value of the current. The co rrect term is "continuous average sine wave power".) Programme power (DIN 45573, or IEC 268-5), sometimes called 'music power'. This figure attempts to take into account the dynamics of a typical sound signal, and for a speaker will typically be around twice the RMS value. . . Peak power (or PMP, peak music power). This refers to the equipment's ability to handle very short duration peaks in the signal, and can be anything from 2 to 20 times the RMS

value. For professional speakers, a figure of 4 times the RMS value is often quo ted. There is no agreed standard for how this value should be measured, so it is of l ittle real use and is often best ignored. These different measurement methods must also be taken into account when matchin g the power rating of a speaker with the output power rating of the amplifier that is going to drive it. (When looking at amplifier rating plates, be careful not to confuse the main s input power requirement with the audio output power rating.) Assuming that the same power ra ting methods are being used for both the speaker and the amplifier, one possible approach is to ensure that the speaker power handling capacity is amply adequate to produce the highest sou nd level required (taking into account the speaker's sensitivity), and then choose an amp lifier whose output power rating into the overall speaker impedance to be driven (see below) is around 3050% higher than the total power handling capacity of the speakers to be connecte d to it. (This is a very basic approach - a more thorough one is detailed in Amplifier and Speaker Selection for required SPL, later on this page.) For example, if two 100 W RMS rated speakers having a combined impedance of 4 oh ms are to be connected to a mono amplifier, then the amplifier output power rating into 4 ohms should be around 260 to 300 W RMS if the maximum useful sound level is to be safely obt ained from the speakers. The reason for the 30-50% margin is that the short-term power hand ling capacity of a speaker is well in excess of its RMS value, whereas that of a high-quality amplifier is usually only marginally above its RMS value. So to be able to utilise the availa ble capacity of the speaker fully, without risk of speaker damage due to the amplifier being dri ven into overload during short-term peaks in the sound level, it is necessary to use an amplifier with an RMS output power capability that is considerably greater than the total RMS capacity of the speakers that are to be connected to it. The down-side of this is that it is possible to slightly over-drive the speakers , but this should be a rare occurrence because if the speakers are correctly rated then over-drivi ng them would produce a sound level higher than what you required. In any case, most speakers will tolerate a degree of short-term over-driving without difficulty. If this is a concern, you can either take care to keep a close eye on the amplifier power meters, or you can start by choosing speakers with a rating somewhat higher than you expect to need (but this may increase the cost significantly). Alternatively you can buy powered speakers, in which case the manufacturer has m ade these

difficult decisions for you! Speaker Sensitivity The sensitivity of a speaker is a measure of the sound pressure level (SPL) that it will produce for a given amount of electrical input power (or voltage - see below). It is usu ally quoted in dB SPL at an input power of one watt, measured at a distance of one metre directly in front of the speaker, for a given type of programme signal. The type of signal is important b ecause the sensitivity of a speaker varies with frequency. A typical sensitivity figure wou ld be 100 dB SPL @ 1W @ 1m, which corresponds to an efficiency of approximately 10%. The sensitivity figure can be used 'in reverse' to estimate the speaker input po wer required to achieve a particular SPL at a given distance, using the inverse square law. F or example, suppose that we require a maximum SPL of 112 dB at a distance of 8 metres from a speaker having the sensitivity figure quoted above. From the inverse square law, the SPL will be 118 dB at 4 metres, 124 dB at 2 metres and 130 dB at 1 metre. Therefore, the maximum po wer input required to the speaker is 30 dB greater than 1 W, which is 1 kW. (For a more co mprehensive

explanation and example, see Amplifier and Speaker Selection for required SPL la ter on this page.) However, this is only an estimate because many complicating factors such as room acoustics and grazing have not been taken into account. Of course, if the partic ular speaker being used is unable to handle this power level (see Power Ratings above), then several can be used together - but the maximum power input to them will still need to total 1 kW. (Refer to the Impedance section below for information regarding the connection of several speakers to a single amplifier.) In practice, an amplifier supplies a controlled voltage, not a controlled power, and the power taken by the speaker depends upon its impedance, which varies with frequency. Th erefore it is becoming increasingly popular to specify speaker sensitivities in voltage-rel ated terms. For nominally 8 ohm speakers, the value is specified in dB SPL @ 2.83 V @ 1m, 2.83 v olts being the voltage required for a power of 1 W in 8 ohms. (For 4 ohm speakers the volta ge required is 2 V.) So, the sensitivity value specified in this new way is effectively equi valent to a value specified in the old power-related way. Impedance For a general definition of impedance, see its glossary entry. The impedance of a speaker is important for three main reasons: . It affects the amount of power that the speaker will draw from an amplifier. A speaker of half the impedance will draw twice as much power from an amplifier, assuming tha t the amplifier input level and the amplifier settings remain the same. (This also ass umes that the amplifier can cope with the lower impedance and that the amplifier's output power rating is not exceeded). . It governs the number of speakers that may be connected to a single amplifier (assuming the usual simple approach to connecting multiple speakers, which conne cts them in parallel). This is because the total load impedance presented to the amp lifier output is the impedance of a single speaker divided by the number of them (assum ing all the speakers have the same impedance), and amplifiers are designed to operat e into load impedances within specified limits - typically 2 to 15 ohms. (We are here r eferring to a single-channel(mono) amplifier, or to onechannel of a multi-channel amplifi er - regardless of how many speaker sockets it has for the connection of speakers t o that channel.) Exceedingthe specified maximum impedance will not usually harm the amplifier - you will simply not be able to achieve the rated output power of the

amplifier. However, by connecting an impedance lowerthan the specified minimum, you risk permanent damage to the amplifier (with possible consequential damage t o the speakers). Where four or more speakers must be connected to a single high-po wer amplifier, a 'series-parallel' method of connection is sometimes used to avoid t he load impedance becoming too low. . It affects the gauge (thickness) of the speaker cables required, and their max imum length, in order to avoid an unacceptable loss of power in the cables, cable ove rheating and a low damping factor. Lower impedance speakers will require heavier gauge an d/or shorter cables. It should be noted that the impedance value quoted for a speaker is a nominal va lue, and in practice the value varies with frequency. The minimum value is often of particul ar interest; this is likely to be around three-quarters of the nominal value, so an 8 ohm speaker is likely to have a minimum impedance of around 6 ohms. A special case of speaker impedance arises in the case of so-called '100 volt li ne' speakers,

which have a much higher impedance than conventional speakers. This allows many of them, located over a large area (such as throughout a public building), to be connecte d to a common amplifier - which must be one having a 100 volt line output - using moderate gau ge cable. These speakers are generally of low power rating (5 to 50 watts RMS each), and a re usually equipped with a means to adjust the power level drawn from the 100 volt line by each speaker. As the usual application of this arrangement is for speech announcements and/or 'musac', the sound quality of these units is often not high - particularly as regards bass re sponse. Bridging The Basics Bridging is a technique to improve the matching between the impedance of a speak er (or the overall impedance of several interconnected speakers) and the optimum load imped ance of the available power amplifiers, so as to increase the maximum amount of power that t he amplifiers can provide to that speaker(s). It is most useful when it is required to use mor e of the powerhandling capacity of the speaker(s), or more of the power-providing capability o f the amplifiers, than could be achieved with a simple non-bridged connection of the speaker(s) to the amplifier. In a bridged arrangement, two identical amplifier channels are driven by signals of opposite polarity, and the speakers are connected between the "hot" (i.e. non-earthy) ter minals of their output connections. This is properly referred to as a 'bridge-tied load' (BTL) c onnection - or, less formally, as 'bridging the amplifiers' - and effectively doubles the voltag e available to the speakers. Some models of amplifier have a switch to select this mode of oper ation; it is sometimes additionally necessary to use a different speaker output connector of the amplifier, or to 'manually' arrange the appropriate connections to its output terminals. WARNING:The output voltage of bridged high-power amplifiers can be high enough t o cause electric shock. The speaker cable used must be suitable for the voltage and curr ent supplied by the amplifier. Speakers and/or amplifiers can be seriously damaged by inappro priate use of bridging, or by making incorrect connections. The following lists provide some basic guidance regarding when it might be appro priate to use bridging, but they are not exhaustive. Reasons NOTto bridge . The two amplifier channels to be used are not identical. . The amplifier handbook does not explicitly state that its channels may be brid ged, or does not state the maximum output power levels that may be obtained in bridged m ode. . The amplifier handbook does not give clear guidance on how to connect the spea ker(s)

to the amplifier outputs to achieve bridged operation, or does not specify the r esulting drive polarity. . The voltage at exposed amplifier/speaker terminals would result in an unaccept able shock hazard. . The speaker cable or connectors to be used are unsuitable for the voltages and /or currents that would result from bridging. . The power level available from the bridged amplifiers places the connected spe akers at too great a risk of overload damage. . The overall impedance of the single load is below the minimum specified for th e amplifier when used in bridged mode. (Typically that minimum value will be twice the minim um load impedance permissible for the individual channels when used in non-bridged mode.)

. Reliability / system integrity considerations dictate that the two amplifier cha nnels must be used independently of each other. Frequency Range Audio frequencies are generally considered to fall into the following ranges (or "bands"): . Sub-bass - Below 80 Hz . Bass - 80 Hz to 250 Hz . Mid-range - 250 Hz to 2.5 kHz . HF - Above 2.5 kHz The boundary frequencies between the ranges are somewhat arbitary, and you will find other similar figures quoted. Sometimes the mid-range band is divided into upper mid a nd lower mid bands. The range of frequencies that a speaker system must handle depends very much on the application. Speakers intended only for announcements in public buildings, for e xample, may only require a range of 200 Hz to 10 kHz, whereas good quality reproduction of m usic requires at least 80 Hz to 15 kHz. Good response down into the "sub-bass" region will all ow the music to be felt as well as heard, and good response up towards the upper limit of hum an hearing (20 kHz) will add increased clarity and crispness to the sound. A speaker that is capable of handling frequencies across the bass, mid and HF ra nges is known as a "full range" speaker, and these are commonly available up to about 80 0 W RMS power handling. However, where a wide frequency response is required at much hig her powers than this, separate speakers are normally used for each frequency range, each sp eaker being specifically designed for a particular band and being driven from an amplifier t hat is fed from an active crossover unit. In this latter arrangement the crossover frequencies m ust be set appropriately for the speakers being used, and the relative levels of the amplif iers must be properly adjusted. It is important to understand that the distribution of audio power is not unifor m across the frequency spectrum - most of the power falls into the bass and lower-mid ranges. As you look at higher and higher frequencies, less and less power is present. This has important consequences in the design of speaker and amplifier systems - especially when se parate speakers are used for the different frequency bands. For example, a 5 kW 3-band system might be made up as follows: . 3 kW bass . 1.6 kW mid-range . 0.4 kW HF Some caution is needed in interpreting the frequency range figures quoted by spe aker

manufactures and suppliers. The norm is to quote the frequency at which the sens itivity has reduced to 3 dB below frequencies around the centre of the band for which it is intended (i.e. the power level of the reproduced sound has halved). However, sometimes "u sable" frequencies are quoted - which may mean a reduction of as much as 10 dB (i.e. of one tenth the sound power) at the quoted frequency, compared with the speaker's mid-band s ensitivity. Amplifier Classes To describe the various possible modes of internal operation, reference is somet imes made to the 'class' of a power amplifier. This is a design parameter, and rarely impa cts significantly

on how the amplifier is used within a system, but for completeness and technical interest this section gives a brief explanation of the most common classes encountered. Some o f these are not relevant to PA applications, but are still listed. As a starting point, it is necessary to understand that all power amplifiers hav e at least two 'output devices' (usually transistors of some kind, though valves are someti mes still encountered) - one device supplies the current to the speaker(s) on positive exc ursions of the waveform, and the other device supplies the current on the negative excursions. (High power amplifiers have several devices operating together to perform each of these two functions, but this has no bearing on the amplifier class.) Simply put, the class of the amplifier is a description of how the two power out put devices are utilised to supply the required output current, particularly how they are coordi nated in their operation in order to achieve the crossing over of the output voltage waveform f rom positive to negative and vice versa. The amplifier class has a bearing on the linearity and the efficiency of the amplifier. 'Linearity' relates to distortion levels - good linearity means l ess distortion (and vice versa). 'Efficiency' relates to how much power is wasted as heat in the amp lifier - high efficiency means that little power is wasted, meaning less mains power is requir ed and that the amplifier can be much smaller and lighter, and requires less cooling. Class Brief description Comment A Both devices pass some current continuously, during both positive and negative excursions. Poor efficiency. B At the point where one device starts to pass current, the other ceases to. Good efficiency but poor linearity around the crossover region. AB One device starts to pass current before the other ceases to, i.e. they are both 'on' for a part of the waveform cycle. Moderate efficiency with good linearity. AB+B A hybrid configuration, using one pair of output devices operating in class AB and another pair operating in class B. Improved efficiency with good linearity. C One device ceases to pass to current before the other starts to, i.e. they are both 'off' for a part of the waveform cycle. Generally used only forradio-frequency power amplification.

D The devices operate in switching mode, either fully 'on' or fully 'off'. They are never both 'on' at the same time. In audio applications, this class is now used to refer to what was originally called class S (see below). [Note that D does not stand for 'digital', and Class D amplifiers should not be referred to asdigital amplifiers.] Very high efficiency. E Normally has only a single output device, driven by Not generally

rectangular pulses. relevant to audio applications. F Similar to class C, but can operate at harmonic frequencies. For radio-frequen cy use. G Uses two different internal DC voltages to supply the output devices, the higher voltage being used only when high signal peaks require it. Improved efficiency. H A further development from class G, in which the higher DC voltage is arranged to track the signal being amplified. Further improved efficiency. S A switching amplifier (as per class D) that is arranged to provide a normal audio output by the addition of a filter to remove the switching transients. Now generally called class D, in audio applications Very high efficiency. Speaker Components Enclosures (Cabinets) All speakers require some kind of enclosure to house the drivers. Enclosures are usually either: . Traditional vinyl-covered or 'carpet'-covered wood construction. . Moulded plastic construction, offering a lighter weight and improved weather-r esistance. However, in the case of full-range, bass and mid-range speakers, the enclosure d oesn't simply house the drivers - it has a significant effect on the sound produced. This is p rimarily because of the damping effect of the air trapped within the enclosure, which is often mo dified by carefully designed 'porting' arrangements - one or more holes which may be fitted with int ernal tuning tubes or ducts. Bass Drivers (Woofers) These are large, heavy units, usually with 12", 15" or 18" diameter cones. As a general rule, the larger the diameter the lower the frequencies that can be reproduced, and th e heavier the magnet the greater the power handling capability. It is important to understand that the operation of a bass driver is heaviliy in fluenced by the speaker enclosure - its dimensions, construction and porting arrangements. There fore, when replacing faulty bass drivers, it is essential to fit the correct type - the low -frequency response or power-handling capability of a speaker cannot usually be improved simply by f itting a higherspec driver. HF Drivers (Horns) HF drivers fall into the following categories: . Dynamic . Piezo (or Electrostatic) Piezo drivers are capable of little power output in comparison with dynamic ones , and so are

used only in low power equipment, generally less than 200 W overall system power . Because the amount of power at high frequencies in an audio signal is much less than the

overall power of the signal, HF drivers are used which have a power rating much lower than the bass drivers. Therefore, the HF drivers are more prone to damage by overload than are the lower-frequency drivers in the speaker system. Although some systems provide som e measure of internal overload protection for the HF drivers, in order to prevent the poss ibility of serious damage it is important to avoid: . High frequency feedback - especially for prolonged periods . Driving the HF drivers from an overdriven (i.e. distorting) amplifier . High-level impulsive sounds, e.g. loud crackles due to faulty cables, making alterations to cabling on an in-use channel, etc. Crossovers and Built-in Amplification The crossover is the component that separates the signal into a number of freque ncy ranges, usually two or three. There are two types of crossover: . Passive: A device that is internally connected between the input connector(s) and the drivers of a passive full-range speaker, or between a single internal amplif ier and the drivers of a powered full-range speaker. The crossover is arranged to supply the high frequency part of the signal to the HF driver(s), the low frequency part to the bass driver(s) and (when the crossover separates the signal into three parts) the mid -range part to the mid-range driver(s). It requires no power source of its own. . Active: A device that is connected between the mixer and the input of the amplifiers, there being a separate amplifier for each frequency range into which the signal is split by the crossover. Therefore, this type will be used as an internal component of a s peaker only if the speaker is of the powered full-range type and contains multiple amplifiers. When an active crossover is used external to the speakers, the speakers connecte d to each amplifier must be suitable for the relevant frequency ranges and the cro ssover frequencies must be adjusted to suit them. The amplifier that is connected to th e HF output of the crossover feeds the HF speakers or drivers, the amplifier connecte d to the LF output of the crossover feeds the bass speakers or drivers) and (when the cro ssover separates the signal into three parts) the amplifier connected to the mid-range output of the crossover feeds the mid-range speakers or drivers. An active crossover requi

res its own power source; in the case of powered speakers this will be the same mains su pply that powers the built-in amplifiers. Note that in the case of a powered full-range speaker, either a passive or an active crossover may be included. Amplifier and Speaker Selection for required SPL There are many factors which may influence the choice of speakers and amplifiers for a particular situation - here we are just consideringsound levels and power rating s. Note that, unless stated otherwise, all the sound levels and power levels and ra tings that we refer to here are 'continuous average' values (often referred to as 'RMS' values , although this is not strictly correct). Do not substitute 'programme power', 'music power' or 'peak power' values (see Power Ratings above). For brevity we will just use the term 'continu ous' here rather than 'continuous average'. To illustrate the principles clearly, we will initially consider a single full-r ange speaker connected to a single amplifier. In summary, the procedure is: . Select a speaker that is able to deliver the required sound level (SPL) at the r equired distance. . Calculate the power input to the speaker necessary for it to give the required s ound level.

. Select an amplifier that is able to deliver the power level required by the sp eaker. We will now break this procedure down into more detailed steps: 1. Determine the furthest distance from the speaker at which members of the audi ence will be located. 2. Decide on the maximum continuous sound level that is needed at that furthest distance. 3. Calculate the continuous sound output level that must be produced by the spea ker at 1 metre from it, in order to give the maximum sound level needed at the furthest r equired distance. (This can be done approximately using the inverse square law.) 4. Check that the sound level will not be excessively high for audience members nearest to the speaker - if so, reconsider the location of the speaker or audience and g o back to step 1. 5. Select a speaker of the desired type that is able to provide the required con tinuous sound output level at 1 metre, and look up itssensitivity figure and impedance v alue. 6. Using the speaker's sensitivity figure, calculate the continuous power input that is required to produce the desired continuous sound output level at 1 metre. (Note that sensitivities that are specified in dB SPL @ 2.83 V, for an 8 O speaker, or in d B SPL @ 2 V, for a 4 O speaker, are essentially equivalent to the more traditional dB SP L @ 1 W figures.) 7. Decide on the amount of headroom required, above the maximum continuous value , to handle the peaks and transients programme of thein question (taking into account any limiting that is applied to the programme signal). 8. Using the continuous power value and the headroom figure, calculate the power required during the peaks and transients, and check that this value of peak powe r can be handled by the selected speaker. 9. Select an amplifier that is able provide an output power level, into that spe aker s impedance, that is at least as high as the value required during the peaks and t ransients, but not so high as to risk the speakers being easily overdriven. (The likelihood of overdriving will depend to some degree upon the skill-level of the sound enginee r.) For multi-channel amplifiers, be sure to use the 'all channels driven' figure. 10. Take care that the amplifier headroom intended to cater for peaks and transi ents is not abused by driving the speaker continuously at greater than its continuous power rating.