Chapter 4

Loudspeakers

A loudspeaker is a transducer which converts electrical energy into acoustical energy. A loudspeaker must therefore have a diaphragm of some sort which is capable of being energised in such a way that it vibrates to produce sound waves which are recognisably similar to the original sound from which the energising signal was derived. To ask a vibrating plastic loudspeaker cone to reproduce the sound of, say, a violin is to ask a great deal, and it is easy to take for granted how successful the best examples have become. Continuing development and refinement of the loudspeaker has brought about a more or less steady improvement in its general performance, but it is a sobering thought that one very rarely mistakes a sound coming from a speaker for the real sound itself, and that one nevertheless has to use these relatively imperfect devices to assess the results of one’s work. Additionally, it is easy to hear significant differences between one model and another. Which is right? It is important not to tailor a sound to suit a particular favourite model. There are several principles by which loudspeakers can function, and the commonly employed ones will be briefly discussed.

A word or two must be said about the loudspeaker enclosure. The box can have as big an influence on the final sound of a speaker system as can the drivers themselves. At first sight surprising, this fact can be more readily appreciated when one remembers that a speaker cone radiates virtually the same amount of sound into the cabinet as out into the room. The same amount of acoustical energy that is radiated is therefore also being concentrated in the cabinet, and the sound escaping through the walls and also back out through the speaker cone has a considerable influence upon the final sound of the system.

The moving-coil loudspeaker

The moving-coil principle is by far the most widely used, as it can be implemented in very cheap transistor radio speakers, PA (Public Address) systems, and also top-quality studio monitors, plus all performance levels and applications in between. Figure 4.1 illustrates a cutaway view of a typical moving-coil loudspeaker. Such a device is also known as a drive unit or driver, as it is the component of a complete speaker system which actually produces the sound or ‘drives’ the air. Basically, the speaker consists of a powerful permanent magnet which has an annular gap to accommodate a coil of wire wound around a cylindrical former. This former is attached to the cone or diaphragm which is held in its rest position by a suspension system which usually consists of a compliant, corrugated, doped (impregnated) cloth material and a compliant surround around the edge of the cone which can be made of a type of rubber, doped fabric, or it can even be an extension of the cone itself, suitably treated to allow the required amount of movement of the cone.

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Figure 4.1   Cross-section through a typical moving-coil loudspeaker

The chassis usually consists either of pressed steel or a casting, the latter being particularly desirable where large heavy magnets are employed, since the very small clearance between the coil and the magnet gap demands a rigid structure to maintain the alignment, and a pressed steel chassis can sometimes be distorted if the loudspeaker is subject to rough handling as is inevitably the case with portable PA systems and the like. (A properly designed pressed steel chassis should not be overlooked though.) The cone itself can in principle be made of almost any material, common choices being paper pulp (as used in many PA speaker cones for its light weight, giving good efficiency), plastics of various types (as used in many hi-fi speaker cones due to the greater consistency achievable than with paper pulp, and the potentially lower coloration of the sound, usually at the expense of increased weight and therefore lower efficiency which is not crucially important in a domestic loudspeaker), and sometimes metal foil.

The principle of operation is based on the principle of electromagnetic transducers described in Fact File 3.1, and is the exact reverse of the process involved in the moving-coil microphone (see Fact File 3.2). The cone vibration sets up sound waves in the air which are an acoustic analogue of the electrical input signal. Thus in principle the moving-coil speaker is a very crude and simple device, but the results obtained today are incomparably superior to the original 1920s Kellog and Rice design. It is, however, a great tribute to those pioneers that the principle of operation of what is still today’s most widely used type of speaker is still theirs.

Other loudspeaker types

The electrostatic loudspeaker first became commercially viable in the 1950s, and is described in Fact File 4.1. The electrostatic principle is far less commonly employed than is the moving coil, since it is difficult and expensive to manufacture and will not produce the sound levels available from moving-coil speakers. The sound quality of the best examples, such as the Quad ESL 63 pictured in Figure 4.2, is, however, rarely equalled by other types of speaker.

Another technique in producing a panel-type speaker membrane has been to employ a light film on which is attached a series of conductive strips which serve as the equivalent of the coil of a moving-coil cone speaker. The panel is housed within a system of strong permanent magnets, and the drive signal is applied to the conductive strips. Gaps in the magnets allow the sound to radiate. Such systems tend to be large and expensive like the electrostatic models, but again very high-quality results are possible. In order to get adequate bass response and output level from such panel speakers the diaphragm needs to be of considerable area.

Fact file 4.1   Electrostatic loudspeaker – principles

The electrostatic loudspeaker’s drive unit consists of a large, flat diaphragm of extremely light weight, placed between two rigid plates. The diagram shows a side view. There are parallels between this loudspeaker and the capacitor microphone described in Chapter 3. The diaphragm has a very high resistance, and a DC polarising voltage in the kilovolt (kV) range is applied to the centre tap of the secondary of the input transformer, and charges the capacitor formed by the narrow gap between the diaphragm and the plates. The input signal appears (via the transformer) across the two rigid plates and thus modulates the electrostatic field. The diaphragm, being the other plate of the capacitor, thus experiences a force which alters according to the input signal. Being free to move within certain limits with respect to the two rigid plates, it thus vibrates to produce the sound.

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There is no cabinet as such to house the speaker, and sound radiates through the holes of both plates. Sound therefore emerges equally from the rear and the front of the speaker, but not from the sides. Its polar response is therefore a figure-eight, similar to a figure-eight microphone with the rear lobe being out of phase with the front lobe.

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Figure 4.2   The Quad ESL63 electrostatic loudspeaker. (Courtesy of Quad Electroacoustics Ltd)

The ribbon loudspeaker principle has sometimes been employed in high-frequency applications (‘tweeters’) and has recently also been employed in large full-range models. Figure 4.3 illustrates the principle. A light corrugated aluminium ribbon, clamped at each end, is placed between two magnetic poles, one north, one south. The input signal is applied, via a step-down transformer, to each end of the ribbon. The alternating nature of the signal causes an alternating magnetic field around the ribbon, which behaves like a single turn of a coil in a moving-coil speaker. The magnets each side thus cause the ribbon to vibrate, producing sound waves. The impedance of the ribbon is often extremely low, and an amplifier cannot drive it directly. A transformer is therefore used which steps up the impedance of the ribbon. The ribbon itself produces a very low acoustic output and often has a horn in front of it to improve its acoustical matching with the air, giving a higher output for a given electrical input. Some ribbons are, however, very long – half a metre or more – and drive the air directly.

A recent panel-type of speaker is the so-called ‘distributed mode loudspeaker’ (DML), developed by the NXT company following the UK’s Defence Evaluation and Research Agency’s discovery that certain lightweight composite panels used in military aircraft could act as efficient sound radiators (Figure 4.4). Its operating principle is the antithesis of conventional wisdom: whereas it is normal practice to strive for ‘pistonic’ motion of a cone driver or panel, the complete area of the radiating surface moving backwards and forwards as a whole with progressively smaller areas of the surface moving as frequency increases, the DML panel is deliberately made very flexible so that a multiplicity of bending modes or resonances, equally distributed in frequency, are set up across its surface. This creates a large number of small radiating areas which are virtually independent of each other, giving an uncorrelated set of signals but summing to give a resultant output. The panel is driven not across its whole area but usually at a strategically placed point by a moving-coil transducer. Because of the essentially random-phase nature of the radiating areas, the panel is claimed not to suffer from the higher-frequency beaming effects of conventional panels, and also there is not the global 180° out-of-phase radiation from the rear.

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Figure 4.3   A ribbon loudspeaker mechanism

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Figure 4.4   DML loudspeaker. (Courtesy of New Transducers Ltd)

Further research into DML materials has brought the promise of integrated audio-visual panels, a single screen radiating both sound and vision simultaneously.

There are a few other types of speaker in use, but these are sufficiently uncommon for descriptions not to be merited in this brief outline of basic principles.

Mounting and loading drive units

‘Infinite baffle’ systems

The moving-coil speaker radiates sound equally in front of and to the rear of the diaphragm or cone. As the cone moves forwards it produces a compression of the air in front of it but a rarefaction behind it, and vice versa. The acoustical waveforms are therefore 180° out of phase with each other and when they meet in the surrounding air they tend to cancel out, particularly at lower frequencies where diffraction around the cone occurs. A cabinet is therefore employed in which the drive unit sits, which has the job of preventing the sound radiated from the rear of the cone from reaching the open air. The simplest form of cabinet is the sealed box (commonly, but wrongly, known as the ‘infinite baffle’) which will usually have some sound-absorbing material inside it such as plastic foam or fibre wadding. A true ‘infinite baffle’ would be a very large flat piece of sheet material with a circular hole cut in the middle into which the drive unit would be mounted. Diffraction around the baffle would then only occur at frequencies below that where the wavelength approached the size of the baffle, and thus cancellation of the two mutually out-of-phase signals would not occur over most of the range, but for this to be effective at the lowest frequencies the baffle would have to measure at least 3 or 4 metres square. The only practical means of employing this type of loading is to mount the speaker in the dividing wall between two rooms, but this is rarely encountered for obvious reasons.

Bass reflex systems

Another form of loading is the bass reflex system, as shown in Figure 4.5. A tunnel, or port, is mounted in one of the walls of the cabinet, and the various parameters of cabinet internal volume, speaker cone weight, speaker cone suspension compliance, port dimensions, and thus mass of air inside the port are chosen so that at a specified low frequency the air inside the port will resonate, which reduces the movement of the speaker cone at that frequency. The port thus produces low-frequency output of its own, acting in combination with the driver. In this manner increased low-frequency output, increased efficiency, or a combination of the two can be achieved. However, it is worth remembering that at frequencies lower than the resonant frequency the driver is acoustically unloaded because the port now behaves simply as an open window. If extremely low frequencies from, say, mishandled microphones or record player arms reach the speaker they will cause considerable excursion of the speaker cone which can cause damage. The air inside a closed box system, however, provides a mechanical supporting ‘spring’ right down to the lowest frequencies.

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Figure 4.5   A ported bass reflex cabinet construction

A device known as an auxiliary bass radiator (ABR) is occasionally used as an alternative to a reflex port, and takes the form of a further bass unit without its own magnet and coil. It is thus undriven electrically. Its cone mass acts in the same manner as the air plug in a reflex port, but has the advantage that mid-range frequencies are not emitted, resulting in lower coloration.

A further form of bass loading is described in Fact File 4.2.

Fact file 4.2   Transmission line system

A form of bass loading is the acoustic labyrinth or ‘transmission line’, as shown in the diagram. A large cabinet houses a folded tunnel the length of which is chosen so that resonance occurs at a specified low frequency. Above that frequency, the tunnel, which is filled or partially filled with acoustically absorbent material, gradually absorbs the rear-radiated sound energy along its length. At resonance, the opening, together with the air inside the tunnel, behaves like the port of a bass reflex design. An advantage of this type of loading is the very good bass extension achievable, but a large cabinet is required for its proper functioning.

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Fact file 4.3   Horn loudspeaker – principles

A horn is an acoustic transformer, that is it helps to match the air impedance at the throat of the horn (the throat is where the speaker drive unit is) with the air impedance at the mouth. Improved acoustic efficiency is therefore achieved, and for a given electrical input a horn can increase the acoustical output of a driver by 10 dB or more compared with that driver mounted in a conventional cabinet. A horn functions over a relatively limited frequency range, and therefore relatively small horns are used for the high frequencies, larger ones for upper mid frequencies, and so on. This is very worthwhile where high sound levels need to be generated in large halls, rock concerts and open-air events.

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Each design of horn has a natural lower cut-off frequency which is the frequency below which it ceases to load the driver acoustically. Very large horns indeed are needed to reproduce low frequencies, and one technique has been to fold the horn up by building it into a more conventional-looking cabinet. The horn principle is rarely employed at bass frequencies due to the necessarily large size. It is, however, frequently employed at mid and high frequencies, but the higher coloration of the sound it produces tends to rule it out for hi-fi and studio monitoring use other than at high frequencies if high sound levels are required. Horns tend to be more directional than conventional speakers, and this has further advantages in PA applications.

Horn loading

Horn loading is a technique commonly employed in large PA loudspeaker systems, as described in Fact File 4.3. Here, a horn is placed in front of the speaker diaphragm.

The so-called ‘long-throw’ horn tends to beam the sound over an included angle of perhaps 90° horizontally and 40° vertically. The acoustical energy is therefore concentrated principally in the forward direction, and this is one reason for the horn’s high efficiency. The sound is beamed forwards towards the rear of the hall with relatively little sound reaching the side walls. The ‘constant directivity’ horn aims to achieve a consistent spread of sound throughout the whole of its working frequency range, and this is usually achieved at the expense of an uneven frequency response. Special equalisation is therefore often applied to compensate for this.

The long-throw horn does not do much for those members of an audience who are close to the stage between the speaker stacks, and an acoustic lens is often employed, which, as its name suggests, diffracts the sound, such that the higher frequencies are spread out over a wider angle to give good coverage at the front. Figure 4.6 shows a typical acoustic lens. It consists of a number of metal plates which are shaped and positioned with respect to each other in such a manner as to cause outward diffraction of the high frequencies. The downward slope of the plates is incidental to the design requirements and it is not incorporated to project the sound downwards. Because the available acoustic output is spread out over a wider area than is the case with the long-throw horn, the on-axis sensitivity tends to be lower.

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Figure 4.6   An example of an acoustic lens

The high efficiency of the horn has also been much exploited in those PA applications which do not require high sound quality, and their use for outdoor events such as fêtes, football matches and the like, as well as on railway station platforms, will have been noticed. Often, a contrivance known as a re-entrant horn is used, as shown in Figure 4.7. It can be seen that the horn has been effectively cut in half, and the half which carries the driver is turned around and placed inside the bell of the other. Quite a long horn is therefore accommodated in a compact structure, and this method of construction is particularly applicable to hand-held loudhailers.

The high-frequency horn is driven not by a cone speaker but by a ‘compression driver’ which consists of a dome-shaped diaphragm usually with a diameter of 1 or 2 inches (2.5 or 5 cm). It resembles a hi-fi dome tweeter but with a flange or thread in front of the dome for fixing on to the horn. The compression driver can easily be damaged if it is driven by frequencies below the cut-off frequency of the horn it is looking into.

Complete loudspeaker systems

Two-way systems

It is a fact of life that no single drive unit can adequately reproduce the complete frequency spectrum from, say, 30 Hz to 20 kHz. Bass frequencies require large drivers with relatively high cone excursions so that adequate areas of air can be set in motion. Conversely, the same cone could not be expected to vibrate at 15 kHz–15 000 times a second to reproduce very high frequencies. A double bass is much larger than a flute, and the strings of a piano which produce the low notes are much fatter and longer than those for the high notes.

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Figure 4.7   A re-entrant horn

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Figure 4.8   Cross-section through a typical dome tweeter

The most widely used technique for reproducing virtually the whole frequency spectrum is the so-called two-way speaker system, which is employed at many quality levels from fairly cheap audio packages to very high-quality studio monitors. It consists of a bass/mid driver which handles frequencies up to around 3 kHz, and a high-frequency unit or ‘tweeter’ which reproduces frequencies from 3 kHz to 20 kHz or more. Figure 4.8 shows a cutaway view of a tweeter. Typically of around 1 inch (2.5 cm) in diameter, the dome is attached to a coil in the same way that a cone is in a bass/mid driver. The dome can be made of various materials, ‘soft’ or ‘hard’, and metal domes are also frequently employed. A bass/mid driver cannot adequately reproduce high frequencies as has been said. Similarly, such a small dome tweeter would actually be damaged if bass frequencies were fed to it; thus a crossover network is required to feed each drive unit with frequencies in the correct range, as described in Fact File 4.4.

In a basic system the woofer would typically be of around 8 inches (20 cm) in diameter for a medium-sized domestic speaker, mounted in a cabinet having several cubic feet internal volume. Tweeters are usually sealed at the rear, and therefore they are simply mounted in an appropriate hole cut in the front baffle of the enclosure. This type of speaker is commonly encountered at the cheaper end of the price range, but its simplicity makes it well worth study since it nevertheless incorporates the basic features of many much more costly designs. The latter differ in that they make use of more advanced and sophisticated drive units, higher-quality cabinet materials and constructional techniques, and a rather more sophisticated crossover which usually incorporates both inductors and capacitors in the treble and bass sections as well as resistors which together give much steeper filter slopes than our 6 dB/octave example. Also, the overall frequency response can be adjusted by the crossover to take account of, say, a woofer which gives more acoustic output in the mid range than in the bass: some attenuation of the mid range can give a flatter and better-balanced frequency response.

Fact file 4.4   A basic crossover network

A frequency-dividing network or ‘crossover’ is fitted into the speaker enclosure which divides the incoming signal into high frequencies (above about 3 kHz) and lower frequencies, sending the latter to the bass/mid unit or ‘woofer’ and the former to the tweeter. A simple example of the principle involved is illustrated in the diagram. In practical designs additional account should be taken of the fact that speaker drive units are not pure resistances.

The tweeter is fed by a capacitor. A capacitor has an impedance which is inversely proportional to frequency, that is at high frequencies its impedance is very low and at low frequencies its impedance is relatively high. The typical impedance of a tweeter is 8 ohms, and so for signals below the example of 3 kHz (the ‘crossover frequency’) a value of capacitor is chosen which exhibits an impedance of 8 ohms also at 3 kHz, and due to the nature of the voltage/current phase relationship of the signal across a capacitor the power delivered to the tweeter is attenuated by 3 dB at that frequency. It then falls at a rate of 6 dB per octave thereafter (i.e.: the tweeter’s output is 9 dB down at 1.5 kHz, 15 dB down at 750 Hz and so on) thus protecting the tweeter from lower frequencies. The formula which contains the value of the capacitor for the chosen 3 kHz frequency is:

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f = 1/(2πRC)

where R is the resistance of the tweeter, and C is the value of the capacitor in farads.

The capacitor value will more conveniently be expressed in microfarads (millionths of a farad) and so the final formula becomes:

C = 159 155 ÷ (8 ohms × 3000 Hz) = 6.7μF

Turning now to the woofer, it will be seen that an inductor is placed in series with it. An inductor has an impedance which rises with frequency; therefore, a value is chosen that gives an impedance value similar to that of the woofer at the chosen crossover frequency. Again, the typical impedance of a woofer is 8 ohms. The formula which contains the value of the inductor is:

f = R/(2πL)

where L = inductance in henrys, R = speaker resistance, f = crossover frequency. The millihenry (one-thousandth of a henry, mH) is more appropriate, so this gives:

L = 8000 ÷ (2π × 3000) = 0.42 mH

Three-way systems

Numerous three-way loudspeaker systems have also appeared where a separate mid-range driver is incorporated along with additional crossover components to restrict the frequencies feeding it to the mid range, for example between 400 Hz and 4 kHz. It is an attractive technique due to the fact that the important mid frequencies where much of the detail of music and speech resides are reproduced by a dedicated driver designed specially for that job. But the increased cost and complexity does not always bring about a proportional advance in sound quality.

Active loudspeakers

So far, only ‘passive’ loudspeakers have been discussed, so named because simple passive components – resistors, capacitors and inductors – are used to divide the frequency range between the various drivers. ‘Active’ loudspeakers are also encountered, in which the frequency range is divided by active electronic circuitry at line level, after which each frequency band is sent to a separate power amplifier and thence to the appropriate speaker drive unit. The expense and complexity of active systems has tended to restrict the active technique to high-powered professional PA applications where four-, five- and even six-way systems are employed, and to professional studio monitoring speakers, such as the Rogers LS5/8 system pictured in Figure 4.9. Active speakers are still comparatively rare in domestic audio.

Each driver has its own power amplifier, which of course immediately increases the cost and complexity of the speaker system, but the advantages include: lower distortion (due to the fact that the signal is now being split at line level, where only a volt or so at negligible current is involved, as compared with the tens of volts and several amps that passive crossovers have to deal with); greater system-design flexibility due to the fact that almost any combination of speaker components can be used because their differing sensitivities, impedances and power requirements can be compensated for by adjusting the gains of the separate power amplifiers or electronic crossover outputs; better control of final frequency response, since it is far easier to incorporate precise compensating circuitry into an electronic crossover design than is the case with a passive crossover; better clarity of sound and firmer bass simply due to the lack of passive components between power amplifiers and drivers; and an improvement in power amplifier performance due to the fact that each amplifier now handles a relatively restricted band of frequencies.

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Figure 4.9   Rogers LS5/8 high-quality active studio loudspeaker. (Courtesy of Swisstone Electronics Ltd)

In active systems amplifiers can be better matched to loudspeakers, and the system can be designed as a whole, without the problems which arise when an unpredictable load is attached to a power amplifier. In passive systems, the designer has little or no control over which type of loudspeaker is connected to which type of amplifier, and thus the design of each is usually a compromise between adaptability and performance. Some active speakers have the electronics built into the speaker cabinet which simplifies installation.

Subwoofers

Good bass response from a loudspeaker requires a large internal cabinet volume so that the resonant frequency of the system can be correspondingly low, the response of a given speaker normally falling away below this resonant point. This implies the use of two large enclosures which are likely to be visually obtrusive in a living room for instance. A way around this problem is to incorporate a so-called ‘subwoofer’ system. A separate speaker cabinet is employed which handles only the deep bass frequencies, and it is usually driven by its own power amplifier. The signal to drive the power amp comes from an electronic crossover which subtracts the low bass frequencies from the feed to the main stereo amplifier and speakers, and sends the mono sum of the deep bass to the subwoofer system.

Freed from the need to reproduce deep bass, the main stereo speakers can now be small high-quality systems; the subwoofer can be positioned anywhere in the room according to the manufacturers of such systems since it only radiates frequencies below around 100 Hz or so, where sources tend to radiate only omnidirectionally anyway. Degradation of the stereo image has sometimes been noted when the subwoofer is a long way from the stereo pair, and a position close to one of these is probably a good idea.

Subwoofers are also employed in concert and theatre sound systems. It is difficult to achieve both high efficiency and a good bass response at the same time from a speaker intended for public address use, and quite large and loud examples often have little output below 70 Hz or so. Subwoofer systems, if properly integrated into the system as a whole, can make a large difference to the weight and scale of live sound.

Loudspeaker performance

Impedance

The great majority of loudspeaker drive units and systems are labelled ‘Impedance = 8 ohms’. This is, however, a nominal figure, the impedance in practice varying widely with frequency (see ‘Sound in electrical form’, Chapter 1). A speaker system may indeed have an 8 ohm impedance at, say, 150 Hz, but at 50 Hz it may well be 30 ohms, and at 10 kHz it could be 4 ohms. Figure 4.10 shows the impedance plot of a typical two-way, sealed box, domestic hi-fi speaker.

The steep rise in impedance at a certain low-frequency is indicative of the low-frequency resonance of the system. Other undulations are indicative of the reactive nature of the speaker due to capacitive and inductive elements in the crossover components and the drive units themselves. Also, the driver/box interface has an effect, the most obvious place being at the already-mentioned LF resonant frequency.

Figure 4.11 shows an impedance plot of a bass reflex design. Here we see the characteristic ‘double hump’ at the bass end. The high peak at about 70 Hz is the bass driver/cabinet resonance point. The trough at about 40 Hz is the resonant frequency of the bass reflex port where maximum LF sound energy is radiated from the port itself and minimum energy is radiated from the bass driver. The low peak at about 20 Hz is virtually equal to the free-air resonance of the bass driver itself because at very low frequencies the driver is acoustically unloaded by the cabinet due to the presence of the port opening. A transmission-line design exhibits a similar impedance characteristic.

The DC resistance of an 8 ohm driver or speaker system tends to lie around 7 ohms, and this simple measurement is a good guide if the impedance of an unlabelled speaker is to be estimated. Other impedances encountered include 15 ohm and 4 ohm models. The 4 ohm speakers are harder to drive because for a given amplifier output voltage they draw twice as much current. The 15 ohm speaker is an easy load, but its higher impedance means that less current is drawn from the amplifier and so the power (volts × amps) driving the speaker will be correspondingly less. So a power amplifier may not be able to deliver its full rated power into this higher impedance. Thus 8 ohms has become virtually standard, and competently designed amplifiers can normally be expected to drive competently designed speakers. Higher-powered professional power amplifiers can also be expected to drive two 8 ohm speakers in parallel, giving a resultant nominal impedance of 4 ohms.

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Figure 4.10   Impedance plot of a typical two-way sealed-box domestic loudspeaker

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Figure 4.11   Impedance plot of a typical bass reflex design

Sensitivity

A loudspeaker’s sensitivity is a measure of how efficiently it converts electrical sound energy into acoustical sound energy. The principles are described in Fact File 4.5. Loudspeakers are very inefficient devices indeed. A typical high-quality domestic speaker system has an efficiency of less than 1 per cent, and therefore if 20 watts is fed into it the resulting acoustic output will be less than 0.2 acoustical watts. Almost all of the rest of the power is dissipated as heat in the voice coils of the drivers. Horn-loaded systems can achieve a much better efficiency, figures of around 10 per cent being typical. An efficiency figure is not in itself a very helpful thing to know, parameters such as sensitivity and power handling being much more useful. But it is as well to be aware that most of the power fed into a speaker has to be dissipated as heat, and prolonged high-level drive causes high voice-coil temperatures.

It has been suggested that sensitivity is not an indication of quality. In fact, it is often found that lower-sensitivity models tend to produce a better sound. This is because refinements in sound quality usually come at the expense of reduced acoustical output for a given input, and PA speaker designers generally have to sacrifice absolute sound quality in order to achieve the high sensitivity and sound output levels necessary for the intended purpose.

Fact file 4.5   Loudspeaker sensitivity

Sensitivity is defined as the acoustic sound output for a given voltage input. The standard conditions are an input of 2.83 volts (corresponding to 1 watt into 8 ohms) and an acoustic SPL measurement at a distance of 1 metre in front of the speaker. The input signal is pink noise which contains equal sound energy per octave (see ‘Frequency spectra of non-repetitive sounds’, Chapter 1). A single frequency may correspond with a peak or dip in the speaker’s response, leading to an inaccurate overall assessment. For example, a domestic speaker may have a quoted sensitivity of 86 dB W−1, that is 1 watt of input will produce 86 dB output at 1 metre.

Sensitivities of various speakers differ quite widely and this is not an indication of the sound quality. A high-level professional monitor speaker may have a sensitivity of 98 dB W−1 suggesting that it will be very much louder than its domestic cousin, and this will indeed be the case. High-frequency PA horns sometimes achieve a value of 118 dB for just 1 watt input. Sensitivity is thus a useful guide when considering which types of speaker to choose for a given application. A small speaker having a quoted sensitivity of 84 dB W−1 and 40 watts power handling will not fill a large hall with sound. The high sound level capability of large professional models will be wasted in a living room.

Distortion

Distortion in loudspeaker systems is generally an order of magnitude or more higher than in other audio equipment. Much of it tends to be second-harmonic distortion (see ‘Harmonic distortion – technical’, Appendix 1) whereby the loudspeaker will add frequencies an octave above the legitimate input signal. This is especially manifest at low frequencies where speaker diaphragms have to move comparatively large distances to reproduce them. When output levels of greater than 90 dB for domestic systems and 105 dB or so for high-sensitivity systems are being produced, low-frequency distortion of around 10 per cent is quite common, this consisting mainly of second-harmonic and partly of third-harmonic distortion.

At mid and high frequencies distortion is generally below 1 per cent, this being confined to relatively narrow bands of frequencies which correspond to areas such as crossover frequencies or driver resonances. Fortunately, distortion of this magnitude in a speaker does not indicate impending damage, and it is just that these transducers are inherently non-linear to this extent. Much of the distortion is at low frequencies where the ear is comparatively insensitive to it, and also the predominantly second-harmonic character is subjectively innocuous to the ear. Distortion levels of 10–15 per cent are fairly common in the throats of high-frequency horns.

Frequency response

The frequency response of a speaker also indicates how linear it is. Ideally, a speaker would respond equally well to all frequencies, producing a smooth ‘flat’ output response to an input signal sweeping from the lowest to the highest frequencies at a constant amplitude. In practice, only the largest speakers produce a significant output down to 20 Hz or so, but even the smallest speaker systems can respond to 20 kHz. The ‘flatness’ of the response, i.e.: how evenly a speaker responds to all frequencies, is a rather different matter. High-quality systems achieve a response that is within 6 dB of the 1 kHz level from 80 Hz to 20 kHz, and such a frequency response might look like Figure 4.12(a). Figure 4.12(b) is an example of a rather lower-quality speaker which has a considerably more ragged response and an earlier bass roll-off.

The frequency response can be measured using a variety of different methods, some manufacturers taking readings under the most favourable conditions to hide inadequacies. Others simply quote something like ‘±3 dB from 100 Hz to 15 kHz’. This does at least give a fairly good idea of the smoothness of the response. These specifications do not, however, tell you how a system will sound, and they must be used only as a guide. They tell nothing of coloration levels, or the ability to reproduce good stereo depth, or the smoothness of the treble, or the ‘tightness’ of the bass.

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Figure 4.12   Typical loudspeaker frequency response plots. (a) A high-quality unit. (b) A lower-quality unit

Power handling

Power handling is the number of watts a speaker can handle before unacceptable amounts of distortion ensue. It goes hand in hand with sensitivity in determining the maximum sound level a speaker can deliver. For example, a domestic speaker may be rated at 30 watts and have a sensitivity of 86 dB W−1. The decibel increase of 30 watts over 1 watt is given by:

dB increase = 10 log 30 ≈ 15 dB

Therefore, the maximum output level of this speaker is 86 + 15 =101 dB at 1 metre for 30 watts input. This is loud, and quite adequate for domestic use. Consider now a PA speaker with a quoted sensitivity of 99 dB W−1. 30 watts input now produces 99 + 15 = 114 dB, some 13 dB more than with the previous example for the same power input. To get 114 dB out of the 86 dB W−1 speaker one would need to drive it with no less than 500 watts, which would of course be way beyond its capabilities. This dramatically demonstrates the need to be aware of the implications of sensitivity and power handling.

A 30 watt speaker can, however, safely be driven even by a 500 watt amplifier providing that sensible precautions are taken with respect to how hard the amplifier is driven. Occasional peaks of more than 30 watts will be quite happily tolerated; it is sustained high-level drive which will damage a speaker. It is perfectly all right to drive a high power speaker with a low-power amplifier, but care must be taken that the latter is not overdriven otherwise the harsh distortion products can easily damage high-frequency horns and tweeters even though the speaker system may have quoted power handling well in excess of the amplifier. The golden rule is to listen carefully. If the sound is clean and unstressed, all will be well.

Directivity

Directivity, or dispersion, describes the angle of coverage of a loudspeaker’s output. Very low frequencies radiated from a speaker are effectively omnidirectional, because the wavelength of the sound is large compared with the dimensions of the speaker and its enclosure, and efficient diffraction of sound around the latter is the result. As the frequency increases, wavelengths become comparable to the dimensions of the speaker’s front surface, diffraction is curtailed, and the speaker’s output is predominantly in the forwards direction. At still higher frequencies, an even narrower dispersion angle results as a further effect comes into play: off-axis phase cancellation. If one listens, say, 30° off-axis from the front of a speaker, a given upper frequency (with a short wavelength) arrives which has been radiated both from the closest side of the speaker cone to the listener and from the furthest side of the cone, and these two sound sources will not therefore be in phase with each other because of the different distances they are away from one another. Phase cancellation therefore occurs, perceived output level falls, and the effect becomes more severe as frequencies increase. The phenomenon is mitigated by designing for progressively smaller radiating areas of the speaker cone to be utilised as the frequency increases, finally crossing over to a tweeter of very small dimensions. By these means, fairly even dispersion of sound, at least in the mid and lower treble regions, can be maintained.

Various other methods have been used to control directivity (the acoustic lens has been covered) and one or two will be described. Low frequencies, which are normally omnidirectional, have been given a cardioid-like dispersion pattern by mounting large speaker drivers on essentially open baffles which by themselves give a figure-of-eight polar response, the output falling with falling frequency. To these was added a considerable amount of absorbent material to the rear, and together with appropriate bass boost to flatten the frequency response of the speakers, predominantly forwards-radiation of low frequencies was achieved. A more elegant technique has been to mount essentially open-baffle speakers (the rear radiation therefore being 180° out of phase with the front producing a figure-of-eight polar pattern, and with bass boost applied to flatten the frequency response) adjacent to closed-box omnidirectional speakers. Their combined acoustical outputs thereby produce a cardioid dispersion pattern, useful for throwing low frequencies forwards into an auditorium rather than across a stage where low-frequency feedback with microphones can be a problem.

Another fascinating technique, introduced by Philips in 1983, is the Bessel Array. It was developed to counteract the beaming effects of multiple-speaker systems. Essentially it makes use of Bessel coefficients to specify phase relationships and output level requirements from each of a horizontal row of speakers necessary to obtain an overall dispersion pattern from the row which is the same as one speaker on its own. Normally, path-length differences between off-axis listeners and the various speaker drivers result in phase cancellations and consequent loss of level, particularly in the upper frequency range. For a horizontal five-speaker row, labelled A, B, C, D and E, the Bessel function gives:

A:B:C:D:E = 1:2:2: – 2:1

In other words, speakers A and E are required to draw half the current of speakers B, C and D; and speaker D must be connected out of phase. A practical implementation would be to connect speakers A and E in series, with speakers B, C and D each connected straight across the system’s input terminals but with D wired out of phase. The speaker drivers are mounted side by side very close together to give good results across the frequency range.

For a seven-speaker row, the Bessel function gives:

A:B:C:D:E:F:G = 1:2:2:0: – 2:2: – 1

Speaker D can therefore be omitted, but a space in the row must be left in its position so as to preserve the correct distance relationships between the others.

Both horizontal and vertical rows of speakers can be combined into a square arrangement so that an array of, for example, 25 speakers, together having potentially very high power handling and output level capability, can, however, give the same dispersion characteristics of one speaker on its own. The amplitude and phase relationships necessary in such an array are given by the numbers in the circles representing the speakers in Figure 4.13.

Images

Figure 4.13   A Bessel Array

It is worth mentioning that the same technique can also be applied to microphones, offering potential for a high output, very low noise array whilst still maintaining a good polar response.

A highly directional speaker incorporating a parabolic reflector of about 1.3 metres in diameter has been developed by the Mayer loudspeaker company as their type SB-1. Designed to work between 500 Hz and 15 kHz, the system comprises an outrigger supporting a small horn and compression driver at the focus of the dish which fires into it, and a small hole at the dish’s centre admits sound from a 12-inch cone driver. Claimed dispersion (−6 dB points) is 10° vertical and 10° horizontal, and maximum peak output at 100 metres distance is 110 dB.

Setting up loudspeakers

Phase

Phase is a very important consideration when wiring up speakers. A positive-going voltage will cause a speaker cone to move in a certain direction, which is usually forwards, although at least two American and two British manufacturers have unfortunately adopted the opposite convention. It is essential that both speakers of a stereo pair, or all of the speakers of a particular type in a complete sound rig, are ‘in phase’, that is all the cones are moving in the same direction at any one time when an identical signal is applied. If two stereo speakers are wired up out of phase, this produces vague ‘swimming’ sound images in stereo, and cancellation of bass frequencies. This can easily be demonstrated by temporarily connecting one speaker in opposite phase and then listening to a mono signal source – speech from the radio is a good test. The voice will seem to come from nowhere in particular, and small movements of the head produce sudden large shifts in apparent sound source location. Now reconnect the speakers in phase and the voice will come from a definite position in between the speakers. It will also be quite stable when you move a few feet to the left or to the right.

Occasionally it is not possible to check the phase of an unknown speaker by listening. An alternative method is to connect a 1.5 V battery across the input terminals and watch which way the cone of the bass driver moves. If it moves forwards, then the positive terminal of the battery corresponds to the positive input terminal of the speaker. If it moves backwards as the battery is connected, then the positive terminal of the battery is touching the negative input terminal of the speaker. The terminals can then be labelled + and −.

Positioning

Loudspeaker positioning has a significant effect upon the performance. In smaller spaces such as control rooms and living rooms the speakers are likely to be positioned close to the walls, and ‘room gain’ comes into effect whereby the low frequencies are reinforced. This happens because at these frequencies the speaker is virtually omnidirectional, i.e.: it radiates sound equally in all directions. The rear- and side-radiated sound is therefore reflected off the walls and back into the room to add more bass power. As we move higher in frequency, a point is reached whereby the wavelength of lower mid frequencies starts to become comparable with the distance between the speaker and a nearby wall. At half wavelengths the reflected sound is out of phase with the original sound from the speaker and some cancellation of sound is caused. Additionally, high-frequency ‘splash’ is often caused by nearby hard surfaces, this often being the case in control rooms where large consoles, tape machines, outboard processing gear, etc. can be in close proximity to the speakers. Phantom stereo images can thus be generated which distort the perspective of the legitimate sound. A loudspeaker which has an encouragingly flat frequency response can therefore often sound far from neutral in a real listening environment. It is therefore essential to give consideration to loudspeaker placement, and a position such that the speakers are at head height when viewed from the listening position (high-frequency dispersion is much narrower than at lower frequencies, and therefore a speaker should be listened to on axis) and also away from room boundaries will give the most tonally accurate sound.

Some speakers, however, are designed to give of their best when mounted directly against a wall, the gain in bass response from such a position being allowed for in the design. A number of professional studio monitors are designed to be let into a wall such that their drivers are then level with the wall’s surface. The manufacturer’s instructions should be heeded, in conjunction with experimentation and listening tests. Speech is a good test signal. Male speech is good for revealing boominess in a speaker, and female speech reveals treble splash from hard-surfaced objects nearby. Electronic music is probably the least helpful since it has no real-life reference by which to assess the reproduced sound. It is worth emphasising that the speaker is the means by which the results of previous endeavour are judged, and that time spent in both choosing and siting is time well spent.

Speakers are of course used in audio–visual work, and one frequently finds that it is desirable to place a speaker next to a video monitor screen. But the magnetic field from the magnets can affect the picture quality by pulling the internal electron beams off course. Some speakers are specially magnetically screened so as to avoid this.

Loudspeaker positioning issues affecting two-channel stereo and surround sound reproduction are covered in greater detail in Chapters 16 and 17.

Recommended further reading

Borwick, J. (2001) ed. Loudspeaker and Headphone Handbook. Focal Press

Colloms, M. (2005) High Performance Loudspeakers, 5th Edition. Wiley

Eargle, J. (2003) Loudspeaker Handbook. Kluwer Academic Publishers

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