Pro Engineer School Vol. 1


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Pro Engineer School Vol. 1

  1. 1. Volume 1
  2. 2. Contents 3. Microphone Technology 19. The Use of Microphones 35. Loudspeaker Drive Units 42. Loudspeaker Systems 51. Analog Recording 64. Digital Audio 75. Digital Audio Tape Recording 86. Appendix 1 – Sound System Parameters Copyright Notice This work is copyright © You are licensed to make as many copies as you reasonably require for your own personal use.
  3. 3. Chapter 1: Microphone Technology The microphone is the front-end of almost all sound engineering activities and, as the interface between real acoustic sound travelling in air and the sound engineering medium of electronics, receives an immense amount of attention. Sometimes one could think that the status of the microphone has been raised to almost mythological proportions. It is useful therefore to put things in their proper perspective: there are a great many microphones available that are of professional quality. Almost any of them can be used in a wide variety of situations to record or broadcast sound to a professional standard. Of course different makes and types of microphones sound different to each other, but the differences don't make or break the end product, at least as far as the listener is concerned. Now, if you want to talk about something that really will make or break the end product, that is how microphones are used. Two sound engineers using the same microphones will instinctively position and direct them differently and there can be a massive difference in sound quality. Give these two engineers other mics, whose characteristics they are familiar with, and the two sounds achieved will be identifiable according to engineer, and not so much to according to microphone type. There are two ways we can consider microphones, by construction and by directional properties. Let's look at the different ways a microphone can be made, to start off with. Microphone Construction There are basically three types of microphone in common use: piezoelectric, dynamic and capacitor. The piezoelectric mic, it has to be said, has evolved into a very specialized animal, but it is still commonly found under the bridge of an electro-acoustic guitar so it is worth knowing about. Piezoelectric The piezoelectric effect is where certain crystalline and ceramic materials have the property of generating an electric current when pressure or a bending force is applied. This makes them sensitive to acoustic vibrations and they can produce a voltage in response to sound. Piezo mics (or
  4. 4. transducers as they may be called - a transducer is any device that converts one form of energy to another) are high impedance. This means that they can produce voltage but very little current. To compensate for this, a preamplifier has to be placed very close to the transducer. This will usually be inside the body of the electro-acoustic guitar. The preamp will run for ages on a 9 volt alkaline battery, but it is worth remembering that if an electro-acoustic guitar, or other instrument with a piezo transducer, sounds distorted, it is almost certainly the battery that needs replacing, perhaps after a year or more of service. Dynamic This is ‘dynamic’ as in ‘dynamo’. The dynamo is a device for converting rotational motion into an electric current and consists of a coil of wire that rotates inside the field of a magnet. Re-configure these components and you have a coil of wire attached to a thin, lightweight diaphragm that vibrates in response to sound. The coil in turn vibrates within the field of the magnet and a signal is generated in proportion to the acoustic vibration the mic receives. The dynamic mic is also sometimes known as the moving coil mic, since it is always the coil that moves, not the magnet - even though that would be possible. The dynamic mic produces a signal that is healthy in both voltage and current. Remember that it is possible to exchange voltage for current, and vice versa, using a transformer. All professional dynamic mics incorporate a transformer that gives them an output impedance of somewhere around 200 ohms. This is a fairly low output impedance that can drive a cable of 100 meters or perhaps even more with little loss of high frequency signal (the resistance of a cable attenuates all frequencies equally, the capacitance of a cable provides a path between signal conductor and earth conductor through which high frequencies can ‘leak’). It is not necessary therefore to have a preamplifier close to the microphone, neither does the mic need any power to operate. Examples of dynamic mics are the famous Shure SM58 and the Electrovoice RE20. The characteristics of the dynamic mic are primarily determined by the weight of the coil slowing down the response of the diaphragm. The sound can be good, particularly on drums, but it is not as crisp and clear as it would have to be to capture delicate sounds with complete accuracy. Dynamic microphones have always been noted for providing good value
  5. 5. for money, but other types are now starting to challenge them on these grounds. Ribbon Mic There is a variation of the dynamic mic known as the ribbon microphone. In place of the diaphragm and coil there is a thin corrugated metal ribbon. The ribbon is located in the field of a magnet. <img src="/graphics/coles4038.jpeg" border=0 width=69 height=114 align=RIGHT hspace=5 vspace=5 alt="">When the ribbon vibrates in response to sound it acts as a coil, albeit a coil with only one turn. Since the ribbon is very light, it has a much clearer sound than the conventional dynamic, and it is reasonable to say that many engineers could identify the sound of a ribbon mic without hesitation. If the ribbon has a problem, it is that the output of the single-turn ‘coil’ is very low. The ribbon does however also have a low impedance and provides a current which the integral transformer can step up so that the voltage output of a modern ribbon mic can be comparable with a conventional dynamic. Examples of ribbon mics are the Coles 4038 and Beyerdynamic M130. Capacitor The capacitor mic, formerly known as the ‘condenser mic’, works in a completely different way to the dynamic. Here, the diaphragm is paralleled by a ‘backplate’. Together they form the plates of a capacitor. A capacitor, of any type, works by storing electrical charge. Electrical charge can be thought of as quantity of electrons (or the quantity of electrons that normally would be present, but aren't). The greater the disparity in number of electrons present – i.e. the amount of charge – the higher will be the voltage across the terminals of the capacitor. There is the equation:
  6. 6. Q = C x V or: charge = capacitance x voltage Note that charge is abbreviated as ‘Q’, because ‘C’ is already taken by capacitance. Putting this another way round: V = Q/C or: voltage = charge / capacitance Now the tricky part: capacitance varies according to the distance between the plates of the capacitor. The charge, as long as it is either continuously topped up or not allowed to leak away, stays constant. Therefore as the distance between the plates is changed by the action of acoustic vibration, the capacitance will change and so must the voltage between the plates. Tap off this voltage and you have a signal that represents the sound hitting the diaphragm of the mic. Sennheiser MKH 40 The great advantage of the capacitor mic is that the diaphragm is unburdened by a coil of any sort. It is light and very responsive to the most delicate sound. The capacitor mic is therefore much more accurate and faithful to the original sound than the dynamic. Of course there is a
  7. 7. downside too. This is that the impedance of the capsule (the part of any mic that collects the sound) is very high. Not just high - very high. It also requires continually topping up with charge to replace that which naturally leaks away to the atmosphere. A capacitor mic therefore needs power for these two reasons: firstly to power an integral amplifier, and secondly to charge the diaphragm and backplate. Old capacitor mics used to have bulky and inconvenient power supplies. These mics are still in widespread use so you would expect to come across them from time to time. Modern capacitor mics use phantom power. Phantom power places +48 V on both of the signal carrying conductors of the microphone cable actually within the mixing console or remote preamplifier, and 0 V on the earth conductor. So, simply by connecting a normal mic cable, phantom power is connected automatically. That's why it is called ‘phantom’ – because you don't see it! In practice this is no inconvenience at all. You have to remember to switch in on at the mixing console but that's pretty much all there is to it. Dynamic mics of professional quality are not bothered by the presence of phantom power in any way, One operational point that is important however is that the fader must be all the way down when a mic is connected to an input providing phantom power, or when phantom power is switched on. Otherwise a sharp crack of speaker-blowing proportions is produced. A capacitor microphone often incorporates a switched -10 dB or -20 dB pad, which is an attenuator placed between the capsule and the amplifier to prevent clipping on loud signals. Electret The electret mic is a form of capacitor microphone. However the charge is permanently locked into the diaphragm and backplate, just as magnetic energy is locked into a magnet. Not all materials are suited to forming electrets, so it is usually considered that the compromises involved in manufacture compromise sound quality. However, it has to be said that there are some very good electret mics available, most of which are back- electrets, meaning that only the backplate of the capacitor is an electret therefore the diaphragm can be made of any suitable material. Electret mics do still need power for the internal amplifier. However, this can take the form of a small internal battery, which is sometimes convenient.
  8. 8. Electret mics that have the facility for battery power can also usually be phantom powered, in case the battery runs down or isn’t fitted.
  9. 9. Directional Characteristics The directional characteristics of microphones can be described in terms of a family of polar patterns. The polar pattern is a graph showing the sensitivity in a full 360 degree circle around the mic. I say a family of polar patterns but it really is a spectrum with omnidirectional at one extreme and figure-of-eight at the other. Cardioid and hypercardioid are simply convenient way points. To explain these patterns further, fairly obviously an omnidirectional mic is equally sensitive all round. A cardioid is slightly less obvious. The cardioid is most sensitive at the front, but is only 6 dB down in response at an angle of 90 degrees. In fact it is only insensitive right at the back. It is not at all correct, as commonly happens, to call this a unidirectional microphone. The hypercardioid is a more tightly focussed pattern than the cardioid, at the expense of a slight rear sensitivity, known as a lobe in the response. The figure-of-eight is equally sensitive at front and back,
  10. 10. the only difference being that the rear produces an inverted signal, 180 degrees out of phase with the signal from the front. All of this is nice in theory, but is almost never borne out in practice. Take a nominally cardioid mic for example. It may be an almost perfect cardioid at mid frequencies, but at low frequencies the pattern will spread out into omni. At high frequencies the pattern will tighten into hypercardioid. The significant knock-on effect of this is that the frequency response off-axis – in other words any direction but head on – is never flat. In fact the off-axis response of most microphones is nothing short of terrible and the best you can hope for is a smooth roll-off of response from LF to HF. Often though it is very lumpy indeed. We will see how this affects the use of microphones at another time. Omnidirectional Looking at directional characteristics from a more academic standpoint, the omnidirectional microphone is sensitive to the pressure of the sound wave. The diaphragm is completely enclosed, apart from a tiny slow- acting air-pressure equalizing vent, and the mic effectively compares the changing pressure of the outside air under the influence of the sound signal with the constant pressure within. Pressure acts equally in all directions, therefore the mic is equally sensitive in all directions, in theory as we said. In practice, at higher frequencies where the size of the mic starts to become significant in comparison with the wavelength, the diaphragm will be shielded from sound approaching from the rear and rearward HF response will drop. Figure-of-Eight At the other end of the spectrum of polar patterns the figure-of-eight microphone is sensitive to the pressure gradient of the sound wave. The diaphragm is completely open to the air at both sides. Even though it is very light and thin, there is a difference in pressure at the front and rear of the diaphragm, and the microphone is sensitive to this difference. The pressure gradient is greatest for sound arriving directly from the front or rear, and lessens as the sound source moves round to the side. When the sound source is exactly at the side of the diaphragm it produces equal pressure at front and back, therefore there is no pressure gradient and the microphone produces no output. Therefore the figure-of-eight microphone is not sensitive at the sides. (You could also imagine that a
  11. 11. sound wave would find it hard to push the diaphragm sideways – sometimes the intuitive explanation is as meaningful as the scientific one). All directional microphones exhibit a phenomenon known as the proximity effect or bass tip-up. The explanation for this is sufficiently complicated to fall outside of the required knowledge of the working sound engineer. The practical consequences are that close miking results in enhanced low frequency. This produces a signal that is not accurate, but it is often thought of as being ‘warmer’ than the more objectively accurate sound of an omnidirectional microphone. Cardioid and Hypercardioid To produce the in-between polar patterns one could consider the omnidirectional microphone where the diaphragm is open on one side only, and the figure-of-eight microphone where the diaphragm is completely open on both sides. Allowing partial access only to one side of the diaphragm would therefore seem to be a viable means of producing the in-between patterns, and indeed it is. A cardioid or hypercardioid mic therefore provides access to the rear of the diaphragm through a carefully designed acoustic labyrinth. Unfortunately the effect of the acoustic labyrinth is difficult to equalize for all frequencies, therefore one would expect the polar response of cardioid and hypercardioid microphones to be inferior to that of omnidirectional and figure-of-eight mics.
  12. 12. Multipattern Microphones There are many microphones available that can produce a selection of polar patterns. This is achieved by mounting two diaphragms back-to- back with a single central backplate. By varying the relative polarization of the diaphragms and backplate, any of the four main polar patterns can be created. It is often thought that the best and most accurate microphones are the true omnidirectional and the true figure-of-eight, and that mimicking these patterns with a multipattern mic is less then optimal. Nevertheless, in practice multipattern mics are so versatile that they are commonly the mic of first choice for many engineers. AKG C414
  13. 13. Special Microphone Types Stereo Microphone Two capsules may be combined into a single housing so that one mic can capture both left and right sides of the sound field. This is much more convenient than setting two mics on a stereo bar, but obviously less flexible. Some stereo mics use the MS principle where one cardioid capsule (M) captures the full width of the sound stage while the other figure-of-eight capsule (S) captures the side-to-side differences. The MS output can be processed to give conventional left and right signals. Neumann stereo microphones Interference Tube Microphone This is usually known as a shotgun or rifle mic because of its similarity in appearance to a gun barrel. The slots in the barrel allow off-axis sound to cancel giving a highly directional response. The longer the mic, the more directional it is. The sound quality of these microphones is inferior to normal mics so they are only used out of necessity. Sennheiser interference tube microphone
  14. 14. A close relation of the interference tube microphone is the parabolic reflector mic. This looks like a satellite dish antenna and is used for recording wildlife noises, and at sports events to capture comments from the pitch. Boundary Effect Microphone The original boundary effect microphone was the Crown PZM (Pressure Zone Microphone) so the boundary effect microphone is often referred to generically as the PZM. In this mic, the capsule is mounted close to a flat metal plate, or inset into a wooden or metal plate. Instead of mounting it on a stand, it is taped to a flat surface. One of the main problems in the use of microphones is reflections from nearby flat surfaces entering the mic. By mounting the capsule within around 7 mm from the surface, these reflections add to the signal in phase rather than interfering with it. The characteristic sound of the boundary effect microphone is therefore very clear (as long as there are no other nearby reflecting surfaces). It can be used for many types of recording, and can also be seen in police interview rooms where obviously a clear sound has to be captured for the interview recording. The polar response is hemispherical. Crown PZM microphone Miniature Microphone This is sometimes known as a ‘tie-clip’ mic, although it is rarely ever clipped to the tie these days. This type of mic is usually of the electret design, which lends itself to very compact dimensions, and is almost always omnidirectional. Miniature microphones are used in television and in theater, where there is a requirement for microphones to be unobtrusive. Since the diaphragm is small and not in contact with many air molecules, the random vibration of the molecules does not cancel out
  15. 15. as effectively as it does in a microphone with a larger diaphragm. Miniature microphones therefore have to be used close to the sound source; otherwise noise will be evident. Beyerdynamic MCE5 Vocal Microphone For popular music vocals it is common to use a large-diaphragm mic, often an old tube model. A large diaphragm mic generally has a less accurate sound than a mic with a diaphragm 10-12 mm or so in diameter. The off-axis response will tend to be poor. Despite this, models such as the Neumann U87 are virtually standard in this application due to their enhanced subjective ‘warmth’ and ‘presence’. Microphone Accessories First in the catalogue of microphone accessories is the mic support. These can range from table stands, short floor stands, normal boom stands, tall stands up to 4 meters for orchestral recording, fishpoles as used by video and film sound recordists, and long booms with cable operated mic positioning used in television studios. Attaching the mic to the stand is a mount that can range from a basic plastic clip, to an elastic suspension or cradle that will isolate the microphone from floor noise. The other major accessory is the windshield or pop-shield. A windshield may be made out of foam and slipped over the mic capsule, or it may look like a miniature airship covered with wind-energy dissipating material. For blizzard conditions windshield covers are available that
  16. 16. look as though they are made out of yeti fur. The pop-shield, on the other hand, is a fine mesh material stretched over a metal or plastic hoop, used to filter out the blast of air cause by a voice artist's or singer's ‘P’ and ‘B’ sounds.
  17. 17. Check Questions • What is the piezoelectric effect? • Where would you find a piezo-electric transducer? • What is attached to the diaphragm of a dynamic microphone? • What passive circuit component is incorporated in the output stage of all professional microphones? (Note that some microphones use an active circuit to imitate the action of this component). • Describe the sound of a dynamic microphone. • How does a ribbon microphone differ from an ordinary dynamic microphone? • What is the old term for 'capacitor microphone'? • Why does the capacitor microphone have a more accurate sound than a dynamic microphone? • Why does a capacitor microphone need to be powered (two reasons)? • What precaution should you take when switching on phantom power? • Can dynamic microphones of professional quality be used with phantom power switched on? • What is a pad? • Why does an electret microphone need to be powered? • Describe the actual polar response of a typical nominally omnidirectional microphone. • Describe the proximity effect. • What is an 'acoustic labyrinth', as applied to microphones?
  18. 18. • Why does a boundary effect microphone give a clear sound? • Why are large-diaphragm microphones used for popular music vocals? • Describe the differences between wind shields and pop shields.
  19. 19. Chapter 2: The Use of Microphones Use of Microphones for Speech In sound engineering, as opposed to communications which will not be considered here, there are commonly considered to be three classes of sound: speech or dialogue, music and effects. Each has its own considerations and requirements regarding the use of microphones. There are a number of scenarios where speech may be recorded, broadcast or amplified: • Audio book • Radio presentation, interview or discussion • Television presentation, interview or discussion • News reporting • Sports commentary • Film and television drama • Theatre • Conference In some of these, the requirement is for speech that is as natural as possible. In an ideal world perhaps it should even sound as though a real person were in the same room. The audio book is in this category, as are many radio programs. There is a qualification however on the term ‘natural’. Sometimes what we regard as a natural sound is the sound that we expect to hear via a loudspeaker, not the real acoustic sound of the human voice. We have all been conditioned to expect a certain quality of sound from our stereos, hifis, radio and television receivers, and when we get it, it sounds natural, even if it isn’t in objective terms. In the recording and most types of broadcasting of speech there are some definite requirements: • No pops on ‘P’ or ‘B’ sounds. • No breath noise or ‘blasting’ • Little room ambience or reverberation • A pleasing tone of voice Popping and blasting can be prevented in two ways. One is to position the microphone so that it points at the mouth, but is out of the direct line of fire of the breath. So often we see microphones used actually in the
  20. 20. line of fire of the breath that it seems as though it is simply the ‘correct’ way to use a microphone. It can be for public address, but it isn’t for broadcasting or recording. The other way is to use a pop shield. Ideally this is an open mesh stocking-type material stretched over a metal or plastic hoop. This can be positioned between the mouth and the microphone and is surprisingly effective in absorbing potential pops and blasts. Sometimes a foam windshield of the type that slips over the end of the microphone is used for this purpose. A windshield is really what it says, and is not 100% effective for pops, although its unobtrusiveness visually has value, for example, for a radio discussion where hoop-type pop shields would mar face-to-face visual communication among the participants. The requirement for little room ambience or reverberation is handled by placing the microphone quite close to the mouth – around 30 to 40 cm. If the studio is acoustically treated, this will work fine. Special acoustic tables are also available which absorb rather than reflect sound from their surface. ‘A pleasing tone of voice’? Well, first choose your voice talent. Second, it is a fact that some microphones flatter the voice. Some work particularly well for speech, and there are some classic models such as the Electrovoice RE20 that are commonly seen in this application. Generally, one would be looking for a large-diaphragm capacitor microphone, or a quality dynamic microphone for natural or pleasing speech for audio books or radio broadcasting. In television broadcasting, one essential requirement is the microphone should be out of shot or unobtrusive. The usual combination for a news anchor, for example, is to have a miniature microphone attached to the clothing in the chest area, backed up by a conventional mic on a desk stand. Often the conventional mic is held on stand-by to be brought on quickly if the miniature mic fails, as they are prone to through constant handling. Oddly enough, the use of microphones on television varies according to geography. In France for example, it is quite common for a television presenter to hand hold a microphone very close to the mouth. Even a discussion can take place with three or four people each holding a microphone. The resultant sound quality is in accordance with French subjective requirements. Radio microphones are commonly used in television to give freedom of movement and also freedom from cables on
  21. 21. the floor, leaving plenty of free space for the cameras to roll around smoothly. News Reporting For news reporting, a robust microphone – perhaps a short shotgun – can be used with a general-purpose foam windshield for both the reporter and interviewee, should there be one. Such a microphone is easily pointable (the reporter isn’t a sound engineer) and brings home good results without any trouble. The sound quality of a news report may not be all that could be imagined, but a little bit of harshness or degradation sometimes, oddly, makes the report more ‘authentic’. Sports Commentary Sports commentary is a very particular requirement. This often takes place in a noisy environment so the microphone must be adapted to cope with this. The result is a mic that has a heavily compromised sound quality, but this has come to be accepted as the sound of sports commentary so it is now a requirement. The Coles 4104 is an example of a 1950s design that is still widely used. It is a noise-cancelling microphone that almost completely suppresses background noise, and the positioning bar on the top of the mic ensures that the commentator always holds it in the correct position (as, indeed it is always held - sports commentators often like to move around in their commentary box as they work). Film and Television Drama For film and television drama, a fishpole (or boom as it is sometimes known) topped by a shotgun or rifle mic with a cylindrical windshield is the norm. The operator can position and angle the mic to get the best quality dialogue (while monitoring on headphones), while keeping the mic – and the shadow of the mic – out of shot. Miniature microphones are also used in this context, often with radio transmitters. Obviously they must not be visible at all. However, concealing the mic in the costume can affect sound quality so care must be taken. Sometimes in the studio a microphone might be mounted on a large floor mounted boom that can extend over several meters (we’re not in fishing
  22. 22. country anymore). In this case the boom operator has winches to point and angle the microphone. Theatre In theatre the choice is between personal miniature microphones with radio transmitters, or area miking from the front and sides of the stage. Personal microphones allow a higher sound level before feedback since they are close to the actor’s mouth. For straight drama, it isn’t necessary to have a high sound level in the auditorium. In fact in most theatres it is perfectly acceptable for the sound of the actors’ voices to be completely unamplified. However if amplification, or reinforcement, is to be used then area miking is usually sufficient. Shotgun or rifle mics are positioned at the front of the stage (an area sometimes known for traditional reason as ‘the floats’, therefore the mics are sometimes called ‘float mics’) to create sensitive spots on stage from which the actors can easily be heard. The drawback is that there will be positions on the stage from which the actors cannot be heard. The movements of the actors have to be planned to take account of this. Conference I use this term loosely to cover everything from company boardrooms to political party conferences. You will see that there can be a vast difference in scale. In the boardroom it has become common to use gooseneck microphones or boundary effect microphones that are specifically designed for that purpose. This lies beyond what we normally consider to be sound engineering and is categorized in the specialist field of sound installation. The party conference is another matter. To achieve reasonably high sound levels the microphone has to be close to the mouth, yet the candidate – for obvious reasons – does not want to look like a microphone-swallowing rock star. Therefore the microphone has to be unobtrusive so that it can be placed fairly close to the mouth without drawing undue attention to itself (the cluster of broadcasters’ microphones in front of the lectern is another matter, but they don’t have to be so close). The AKG C747 is very suitable for this application. You will have noticed that in this context microphones are often used in pairs. There are two schools of thought on this issue. One is that the microphones should point inwards from the front corners of the lectern.
  23. 23. This allows the speaker to turn his or her head and still receive adequate pickup. Unfortunately, as the head moves, both microphones can pick up the sound while the sound source – the mouth – is moving towards one mic and away from the other. The Doppler effect comes into play and two slightly pitch shifted signals are momentarily mixed together. It sounds neither pleasant nor natural. The alternative approach is to mount both microphones centrally and use one as a backup. The speaker will learn, through not hearing their voice coming back through the PA system, that they can only turn so far before useful pickup is lost. It is worth saying that in this situation, the person speaking must be able to hear their amplified voice at the right level. If their voice seems too loud, to them, they will instinctively back away from the mic. If they can’t hear their amplified voice they will assume the system isn’t working. I once saw the chairman of a large and prestigious organisation stand away from his microphone because he thought it wasn’t working. It had been, and at the right level for the audience. But unfortunately, apart from the front few rows, they were unable to hear a single unamplified word he said.
  24. 24. Use of Microphones for Music The way in which microphones are used for music varies much more according to the instrument than it possibly could for speech where the source of sound is of course always the human mouth. First, some scenarios: • Recording • Broadcast • Public address • Recording studio • Location recording • Concert hall • Amplified music venue • Theatre The requirements of recording and broadcasting are very similar, except that broadcasting often works to a more stringent timescale, and in television broadcasting microphones must be invisible or at least unobtrusive. There are two golden rules: Point the microphone at the sound source from the direction of the best natural listening position. The microphone will always be closer than a natural comfortable listening distance. So, wherever you would normally choose to listen from is the right position for the microphone, except that the microphone has to be closer because it can’t discriminate direct sound from reflected sound in the way the human ear/brain can. It is always a good starting point to follow these two rules, but of course it may not always be possible, practical, or a natural sound may not be wanted for whatever reason. Broadcasters, by the way, tend to place the microphone closer than recording engineers. They need to get a quick, reliable result, and a close mic position is simply safer for this purpose. Ultimate sound quality is not of such importance. The recording studio is a very comfortable environment for microphones. The engineer is able to use any microphone he or she desires and has
  25. 25. available. The mic may be old, large and ugly, cumbersome to use perhaps with an external power supply (not phantom) and pattern selector, prone to faults etc., but if it gets the right sound, then it will be used. Location recording is not quite so comfortable and you need to be sure that the microphones are reliable and easy to use, preferably without external power supplies and with a simple stand mount rather than a complicated elastic suspension. As far as comfort goes, the concert hall is a reasonably good place to record in as at least they are used to the requirements of music (the owners of many good recording venues often have higher priorities – religious worship being a prominent example). There are however restrictions on the placement of microphones during a concert. Usually it is against fire regulations to have microphones among the audience, unless the mics are positioned in such a way that they don’t impede egress and cables are very securely fixed. Generally therefore there will be a stereo pair of mics slung from the ceiling, supplemented by a number of mics on stage, which are closer than the engineer would probably prefer them to be under ideal circumstances. For amplified music, the problem is always in getting sufficient level without feedback. This necessitates that microphones are very much closer than the natural listening position, to the point that natural direction has very little meaning. The ultimate example would be a microphone clipped to the bridge or sound hole of a violin. It wouldn’t even be possible to listen from this position. In rock music PA, microphones are used as close to the singer’s lips as possible, right against the grille cloth of a guitarist’s speaker cabinet and within millimetres of the heads of the drums. Primarily this is to achieve level without risk of feedback. However this has also come to be understood as the ‘rock music sound’ because it is what the audience expects. In this context, the most distant mics would be the drum overhead mics, which don’t need much gain anyway. For string and wind instruments there are a variety of clip-on mics available. There are also contact mics that pick up vibrations directly from the body of the instrument, although even these are not entirely immune to feedback. In theatre musicals, the best option for the lead performers is to use miniature microphones with radio transmitters. The placement of the mic is significant. The original ‘lavalier’ placement, named for Mme Lavalier
  26. 26. who reportedly wore a large ruby from her neck, has long gone. The chest position is great for newsreaders but it suffers from the shadow of the chin and boominess caused by chest resonance. The best place for a miniature microphone is on a short boom extending from behind the ear. Mics and booms are available in a variety of flesh colours so they are not visible to the audience beyond the second or third row. If a boom is not considered acceptable, then the mic may protrude a short distance from above the ear, or descending from the hairline. This actually captures a very good vocal sound. It has to be tried to be believed. One of the biggest problems with miniature microphones in the theatre is that they become ‘sweated out’ after a number of performances and have to be replaced. Still, no-one said that it was easy going on stage. For the orchestra in a theatre musical, clip on mics are good for string instruments. Wind instruments are generally loud enough for conventional stand mics, closely placed. So-called ‘booth singers’ can use conventional mics.
  27. 27. Stereo Microphone Techniques Firstly, what is stereo? The word ‘stereophonic’ in its original meaning it suggests a ‘solid’ sound image and does not specify how many microphones, channels or loudspeakers are to be used. However, it has come to mean two channels and two loudspeakers using as few or as many microphones that are necessary to get a good result. When it works, you should be able to sit in an equilateral triangle with the speakers, listen to a recording of an orchestra and pinpoint where every instrument is in the sound image. (By the way, some people complain that ‘stereophonic’, as a word, combines both Greek and Latin roots. Just as well perhaps, because if it had been exclusively Latin it would have been ‘crassophonic’!) When recording a group of instruments or singers, it is possible to use just two or three microphones to pick up the entire ensemble in stereo, and the results can be very satisfying. There are a number of techniques: • Coincident crossed pair • Near-coincident crossed pair • ORTF • Mercury Living Presence • Decca Tree • Spaced omni • MS • Binaural The coincident crossed pair technique traditionally uses two figure-of- eight microphones angled at 90 degrees pointing to the left and right of the sound stage (and, due to the rear pickup of the figure-of-eight mic, to the left and right of the area where the audience would be also). More practically, two cardioid microphones can be used. They would be angled at 120 degrees were it not for the drop off in high frequency response at this angle in most mics. A 110-degree angle of separation is a reasonable compromise. This system was originally proposed in the 1930s and mathematically inclined audio engineers will claim that this gives perfect reproduction of the original sound field from a standard pair of stereo loudspeakers. However perfect the mathematics look on paper, the results do not bear out the theory. The sound can be good, and you can with
  28. 28. effort tell where the instruments are supposed to be in the sound image. The problem is that you just don’t feel like you are in the concert hall, or wherever the recording was made. The fact that human beings do not have coincident ears might have something to do with it. Coincident crossed pair Separating the mics by around 10 cm tears the theory into shreds, but it sounds a whole lot better. Near-coincident crossed pair The ORTF system, named for the Office de Radiodiffusion Television Francaise, uses two cardioid microphones spaced at 17 cm angled outwards at 110 degrees, and is simply an extended near-coincident crossed pair.
  29. 29. The redeeming feature of the coincident crossed pair is that you can mix the left and right signals into mono and it still sounds fine. Mono, but fine. We call this mono compatibility and it is important in many situations – the majority of radio and television listeners still only have one speaker. The further apart the microphones are spaced, the worse the mono compatibility, although near-coincident and ORTF systems are still usable. ORTF Mercury Living Presence was one of the early stereo techniques of the 1950s, used for classical music recordings on the Mercury label. If you imagine trying to figure out how to make a stereo recording when there was no-one around to tell you how to do it, you might work out that one microphone pointing left, another pointing center and a third pointing right might be the way to do it. Record each to its own track on 35mm magnetic film, as used in cinema audio, and there you have it! Nominally omnidirectional microphones were used, but of course the early omni mics did become directional at higher frequencies. Later recordings were made to two-track stereo. These recordings stand up remarkable well today. They may have a little noise and distortion, but the sound is wonderfully clear and alive. The same can be said of the Decca tree, used by the Decca record company. This is not dissimilar from the Mercury Living Presence system but baffles were used between the microphones in some instances to create separation, and additional microphones might be used where necessary, positioned towards the sides of the orchestra.
  30. 30. Decca tree Another obvious means of deploying microphones in the early days of stereo was to place three microphones spaced apart at the front of the orchestra, much more distant from each other than in the above systems. If only two microphones are used spaced apart by perhaps as much as two meters or more, what happens on playback is that the sound seems to cluster around the loudspeakers and there is a hole in the middle of the sound image. To prevent this, a centre microphone can be mixed in at a lower level so that the ‘hole’ is filled. There is no theory on earth to explain why this works - being so dissimilar to the human hearing system - but it can work very well. The main drawback is that a recording made in such a way sounds terrible when played in mono. The MS system, as explained previously, uses a cardioid microphone to pick up an all-round mono signal, and a figure-of-eight mic to pick up the difference between left and right in the sound field. The M and S signals can be combined without too much difficulty to provide conventional left and right signals. This is of practical benefit when it is necessary to record a single performer in stereo. With a coincident crossed pair, one microphone would be pointing to the left of the performer, the other would be pointing to the right. It just seems wrong not to point a microphone directly at the performer, and with the MS system you do, getting the best possible sound quality from the mic. It is sometimes proposed as an advantage of MS than it is possible to control the width of the stereo image by adjusting the level of the S signal. This is exactly the same as adjusting the width by turning the mixing console’s panpots for
  31. 31. the left and right signals closer to the centre. Therefore it is in reality no advantage at all. Binaural stereo attempts to mimic the human hearing system with a dummy head (sometimes face, shoulders and chest too) with two omnidirectional microphones placed in artificial ears just like a real human head. It works well, but only on headphones. A binaural recording played on speakers doesn’t work because the two channels mix on their way to the listener, spoiling the effect. There have been a number of systems attempting to make binaural recordings work on loudspeakers but none has become popular. In addition to the stereo miking system, it is common to mic up every section of an orchestra, whether it is a classical orchestra, film music, or the backing for a popular music track. Normally the stereo mic system, crossed pair or whatever, is considered the main source of signal, with the other microphones used to compensate for the distance to the rear of the orchestra, and to add just a little presence to instruments where appropriate. Sectional mics shouldn’t be used to compensate for poor balance due to the conductor or arranger. Sometimes however classical composers don’t get the balance quite right and it is not acceptable to change the orchestration. A little technical help is therefore called for. Instruments We come back to the two golden rules of microphone placement, as above. It is worth looking at some specific examples: Saxophone There are two fairly obvious ways a saxophone can be close miked. One is close to the mouthpiece, another is close to the bell. The difference in sound quality is tremendous. The same applies to all close miking. Small changes in microphone position can affect the sound quality enormously. There are many books and texts that claim to tell you how and where to position microphones for all manner of instruments, but the key is to experiment and find the best position for the instrument – and player – you have in front of you. Experience, not book learning, leads to success. Of the two saxophone close miking positions, neither will capture the natural sound of the instrument, if that’s what you want. Close mic positions almost never do. If you move the mic further away, up to
  32. 32. around a meter, you will be able to capture the sound of the whole of the instrument, mouthpiece, bell, the metal of the instrument, and the holes that are covered and uncovered during the normal course of playing. Also as you move away you will capture more room ambience, and that is a compromise that has to be struck. Natural sound against room ambience. It’s subjective. Piano Specifically the grand piano – it is common to place the microphone (or microphones) pointing directly at the strings. Oddly enough no-one ever listens from this position and it doesn’t really capture a natural sound, but it might be the sound you want. The closer the microphones are to the higher strings, the brighter the sound will be. You can position the microphones all the way at the bass end of the instrument, spaced apart by maybe 30 cm, and a rich full sound will be captured. Move the microphones below the edge of the case and angle them so that they pick up reflected sound from the lid and a more natural sound will be discovered. You can even place a microphone under a grand piano to capture the vibration of the soundboard. It can even sound quite good, but listen out for noise from the foot pedals. Drums The conventional setup is one mic per drum, a mic for the hihat perhaps, and two overhead mics for the cymbals. Recording drums is an art form and experience is by far the best guide. There are some points to bear in mind: You can’t get a good recording of a poor kit, particularly cymbals, or a kit that isn’t well set up. It is often necessary to damp the drums by taping material to the edge of the drum head to get a shorter, more controlled sound. The mics have to be placed where the drummer won’t hit them, or the stands. Dynamic mics generally sound better for drums, capacitor mics for cymbals.
  33. 33. The kick drum should have its front head removed, or there should be a large hole cut out so that a damping blanket can be placed inside. Otherwise it will sound more like a military bass drum than the dull thud that we are used to. The choice of beater – hard or soft - is important, as is the position of the kick drum mic either just outside, or some distance inside the drum. The snares on the underside of the snare drum may rattle when other drums are being played. Careful adjustment of the tension of the snares is necessary, and perhaps even a little damping. Microphones should be spaced as far apart from each other as possible and directed away from other drums. Every little bit helps as the combination of two mics picking up the same drum from different distances leads to cancellation of groups of frequencies. The brute force technique is to use a noise gate on every microphone channel, and this is commonly done. Noise gates will be covered later. Perhaps this is a brief introduction to the use of microphones, but it’s a start. And to round off I’ll give away the secret of getting good sound from your microphones: Listen!
  34. 34. Check Questions • What problem is commonly found in live sports commentary? • What does a fishpole operator concentrate on while working? • In theater, what is 'area miking'? • How is feedback avoided in live sound (the simplest technique)? • Why must the speaker at a conference hear his or her own amplified voice at the right level? • Write down, copy if you wish, the two golden rules for microphone positioning • Why do microphones have to be placed closer than a natural listening position? • Where are personal mics worn in the theater? • What is stereo? • Describe the coincident crossed pair. • What is the benefit of separating the microphones (relate this to the human hearing system)? • What is the value of mono compatibility? • Why is it desirable to mic up every section of an orchestra independently? • Pick an instrument other than those mentioned in the text. Describe the effect of two alternative close miking positions. • When you look at a grand piano, performed solo, on stage, does the pianist sit on the left or the right? Why? • Why do drums often need to be damped?
  35. 35. Chapter 3: Loudspeaker Drive Units Loudspeakers are without doubt the most inadequate component of the audio signal chain. Everything else, even the microphone, is as close to the capabilities of human hearing as makes hardly any difference at all. However, amplify the signal and convert it back into sound and you will know without any hesitation whatsoever that you are listening to a loudspeaker, not a natural sound source. Loudspeakers can be categorized by method of operation and by function: • Method of operation: • Moving coil • Electrostatic • Direct radiator • Horn • Function: • Domestic • Hi-fi • Studio • PA In this context we will use ‘PA’ to mean concert public address rather than announcement systems that are beyond the scope of this text. The moving coil loudspeaker, or I should say ‘drive unit’ as this is only one component of the complete system, is the original and still most widely used method of converting an electric signal to sound. The components consist of a magnet, a coil of wire (sometimes called the ‘voice coil’) positioned within the field of the magnet and a diaphragm that pushes against the air. When a signal is passed through the coil, it creates a magnetic field that interacts with the field of the permanent magnet causing motion in the coil and in turn the diaphragm. It is probably fair to say that 99.999% of the loudspeakers you will ever come across use moving coil drive units. The electrostatic loudspeaker (and this time it is a loudspeaker rather than just a drive unit) uses electrostatic attraction rather than magnetism. The electrostatic loudspeaker has the most natural sound quality, but is not
  36. 36. capable of high sound levels. Hence it is rarely used in professional audio outside of, occasionally, classical music recording. A moving coil drive unit can be constructed as either a direct radiator or a horn. In a director radiator drive unit, the diaphragm pushes directly against the air. This is not very efficient as the diaphragm and the air have differing acoustic impedance, which creates a barrier for the sound to cross. A horn makes the transition from vibration in the diaphragm to vibration in the open air more gradual, therefore it is more efficient, and for a given input power the horn will be louder. Let's look at these in more detail: Moving Coil Drive Unit Perhaps the best place to start is a 200 mm drive unit intended for low and mid frequency reproduction. This isn't the biggest drive unit available, so why are larger drive units ever necessary? The answer is to achieve a higher sound level. A 200 mm drive unit only pushes against so much air. Increase the diameter to 300 mm or 375 mm and many more air molecules feel the impact. The next question would be, why are 300
  37. 37. mm or 375 mm drive units not used more often, when space is available? The answer to that is in the behavior of the diaphragm: The diaphragm must not bend in operation otherwise it will produce distortion. It is sometimes said that the diaphragm should operate as a ‘rigid piston’. The diaphragm could be flat and still produce sound. However, since the motor is at the center and vibrations are transmitted to the edges, the diaphragm needs to be stiff. The cone shape is the best compromise between stiffness and large diameter. High frequencies will tend to bend the diaphragm more than low frequencies. It takes a certain time for movement of the coil to propagate to the edge of the diaphragm. Fairly obviously, at high frequencies there isn't so much time and at some frequency the diaphragm will start to deviate from the ideal rigid piston. 200 mm is a good compromise. It will produce enough level at low frequency for the average living room, and it will produce reasonably distortion-free sound up to around 4 kHz or so. When the diaphragm bends, it is called break up, due to the vibration ‘breaking up’ into a number of different modes. ‘Break up’, in this context, doesn't mean severe distortion or anything like that. In fact most low frequency drive units are operated well into the break up region. It is up to the designer to ensure that the distortion created doesn't sound too unpleasant. By the way, it is often thought that a larger drive unit will operate down to lower frequencies. This isn't quite the right way to look at it. Any size of drive unit will operate down to as low a frequency as you like, but you need a big drive unit to shift large quantities of air at low frequency. At high frequency, the drive unit vibrates backwards and forwards rapidly, moving air on each vibration. At low frequencies there are fewer opportunities to move air, therefore the area of the drive unit needs to be greater to achieve the desired level. The material of the diaphragm has a significant effect on its stiffness. Early moving coil drive units used paper pulp diaphragms, which were not particularly stiff. Modern drive units use plastic diaphragms, or pulp diaphragms that have been doped to stiffen them adequately. Of course, the ultimate in stiffness would be a metal diaphragm. Unfortunately, it would be heavy and the drive unit would be less efficient. Carbon fiber
  38. 38. diaphragms have also been used with some success. (It is worth noting that in drive units used for electric guitars, the diaphragm is designed to bend and distort. It is part of the sound of the instrument and a distortion- free sound would not meet a guitarist's requirements). Moving up the frequency range: as we have said, the diaphragm will bend and produce distortion. Even if it didn't, there would still be the problem that a large sound source will tend to focus sound over a narrow area, becoming narrower as the frequency increases. In fact, this is the characteristic of direct radiator loudspeakers: that their angle of coverage decreases as the frequency gets higher. This is significant in PA, where a single loudspeaker has to cover a large number of people. (It is perhaps counter-intuitive that a large sound source will focus the sound, but it is certainly so. A good acoustics text will supply the explanation). Because of these two factors, higher frequencies are handled by a smaller drive unit. A smaller diaphragm is more rigid at higher frequencies, and because it is smaller it spreads sound more widely. Often the diaphragm is dome shaped rather than conical. This is part of the designer's art and isn't of direct relevance to the sound engineer, as long as it sounds good. It might be stating the obvious at this stage, but a low frequency drive unit is commonly known as a woofer, and a high frequency drive unit as a tweeter. In loudspeakers where a low frequency drive unit greater than 200 mm is used, it will not be possible to use the woofer up to a sufficiently high frequency to hand over directly to the tweeter. Therefore a mid frequency drive unit has to be used (sometimes known as a squawker!). The function of dividing the frequency band among the various drive units is handled by a crossover, more on which later. Damage There are two ways in which a moving coil drive unit may be damaged. One is to drive it at too high a level for too long. The coil will get hotter and hotter and eventually will melt at one point, breaking the circuit (‘thermal damage’). The drive unit will entirely cease to function. The other is to ‘shock’ the drive unit with a loud impulse. This can happen if a microphone is dropped, or placed too close to a theatrical pyrotechnic effect. The impulse won't contain enough energy to melt the coil, but it
  39. 39. may break apart the turns of the coil, or shift it from its central position with respect to the magnet (‘mechanical damage’). The drive unit will still function, but the coil will scrape against the magnet producing a very harsh distorted sound. Many drive units can be repaired, but of course damage is best avoided in the first place. The trick is to listen to the loudspeaker. It will tell you when it is under stress if you listen carefully enough. One common question regarding damage to loudspeakers is this: What should the power of the amplifier be in relation to the rated power of the loudspeaker? In fact, although the power of an amplifier can be measured very accurately, the capacity of a loudspeaker to soak up this power is only an intelligent guess, at best. During the design process, the manufacturer will test drive units to destruction and arrive at a balance between a high rating (in watts) that will impress potential buyers, and a low number of complaints from people who have pushed their purchases too hard. The rating on the cabinet is therefore only a guide. To get the best performance from a loudspeaker, the amplifier should be rated higher in terms of watts. It wouldn't be unreasonable to connect a 200 W amplifier to a 100 W speaker, and it won't blow the drive units unless you push the level too high. It is up to the sound engineer to control the level. Suppose, on the other hand, that a 100 W amplifier was connected to a 200 W loudspeaker (two-way, with woofer and tweeter). The sound engineer might push the level so high that the amplifier started to clip. Clipping produces high levels of high frequency distortion. In a 200 W loudspeaker, the tweeter could be rated at as little as 20-30 W, as under normal circumstances that is all it would be expected to handle. But under clipping conditions the level supplied to the tweeter could be massively higher, and it will blow. Impedance Drive units and complete loudspeaker systems are also rated in terms of their impedance. This is the load presented to the amplifier, where a low impedance means the amplifier will have to deliver more current, and hence ‘work harder’. A common nominal impedance is 8 ohms. ‘Nominal’ means that this is averaged over the frequency range of the drive unit or loudspeaker, and you will find that the actual impedance departs significantly from nominal according to frequency. Normally this isn't particularly significant, except in two situations:
  40. 40. At some frequency the impedance drops well below the nominal impedance. The power amplifier will be called upon to deliver perhaps more power than it is capable of, causing clipping, or perhaps the amplifier might even go into protection mode to avoid damage to itself. The output impedance of a power amplifier is very low – just a small fraction of an ohm. You could think of the output impedance of the amplifier in series with the impedance of the loudspeaker as a potential divider. Work out the potential divider equation with R1 equal to zero and you will see that the output voltage is equal to the input voltage. However, give R1 some significant impedance, as would happen with a long run of loudspeaker cable, and you will see a voltage loss. Make R2 - the loudspeaker impedance - variable with frequency and you will now see a rather less than flat frequency response. To be honest, the above points are not always at the forefront of the working sound engineer's mind, but they are significant and worth knowing about.
  41. 41. Check Questions • What is the difference between the terms 'loudspeaker' and 'drive unit'? • How does a moving coil drive unit work? • Comment on the two qualities of an electrostatic loudspeaker. • What is a director radiator drive unit? • What is the function of a horn? • Why are drive units larger than 200 mm sometimes used? • What is meant by the phrase 'rigid piston'? • Why is the diaphragm of a moving coil loudspeaker normally cone shaped? • Why does the diaphragm bend more at higher frequencies? • What is 'break up'? • Does breakup occur in a woofer in normal operation? • Why should a guitar drive unit distort intentionally? • Comment on the 'beaming' effect of a large drive unit. • When is a separate midrange drive unit necessary? • Comment on the two damage modes of moving coil drive units. • If a loudspeaker is rated at 100 W, what should be the power of the amplifier, according to the text?
  42. 42. Chapter 4: Loudspeaker Systems Cabinet (Enclosure) The moving coil drive unit is as open to the air at the rear as it is to the front, hence it emits sound forwards and backwards. The backward- radiated sound causes a problem. Sound diffracts readily, particularly at low frequencies, and much of the energy will 'bend' around to the front. Since the movement of the diaphragm to the rear is in the opposite direction to the movement to the front, this leaked sound is inverted (or we can say 180 degrees out of phase) and the combination of the two will tend to cancel each other out. This occurs at frequencies where the wavelength is larger than the diameter of the drive unit. For a 200 mm drive unit the frequency at which cancellation would start to become significant is 1700 Hz, the cancellation getting worse at lower frequencies. The simple solution to this is to mount the drive unit on a baffle. A baffle is simply a flat sheet of wood with a hole cut out for the drive unit. Amazingly, it works. But to work well down to sufficiently low frequencies it has to be extremely large. The wavelength at 50 Hz, for example, is almost 7 meters. The baffle can be folded around the drive unit to create an open back cabinet, which you will still find in use for electric guitar loudspeakers. The drawback is that the partially enclosed space creates a resonance that colors the sound. The logical extension of the baffle and open back cabinet is to enclose the rear of the drive unit completely, creating an infinite baffle. It would now seem that the rear radiation is completely controlled. However, there are problems: The diaphragm now has to push against the air 'spring' that is trapped inside the cabinet. This present significant opposition to the motion of the diaphragm. Sound will leak through the cabinet walls anyway. The cabinet will itself vibrate and is highly unlikely to operate anything like a rigid piston or have a flat frequency response. (Of course, this happens with the open back cabinet too).
  43. 43. At this point it is worth saying that the bare drive unit is often used in theater sound systems where there is a need for extreme clarity in the human vocal range. Low frequencies can be bolstered with conventional cabinet loudspeakers. Despite these problems, careful design of the drive unit to balance the springiness of the trapped air inside the cabinet against the springiness of the suspension can work wonders. The infinite baffle, properly designed, is widely regarded as the most natural sounding type of loudspeaker (electrostatics excepted). The only real problem is that the compromises that have to be made to make this design work result in poor low frequency response. Points of order: 'Springiness' is more properly known as compliance. Another term for 'infinite baffle' is acoustic suspension. You would need a very deep understanding of loudspeakers (starting with the Thiele-Small parameters of drive units) to be able to design a loudspeaker that would work well for studio or PA use. Electric guitar loudspeakers are not so critical. The next step in cabinet design is the bass reflex enclosure. You will occasionally hear of this as a ported or vented cabinet. The bass reflex cabinet borrows the theory of the Helmholtz resonator. A Helmholtz resonator is nothing more than an enclosed volume of air connected to the outside world by a narrow tube, called the port. The port can stick out of the enclosure as in a beer bottle - a perfect example of the principle - or inwards. The small plug of air in the port bounces against the compliance of the larger volume of air inside and resonates readily. Try blowing across the top of the beer bottle (when empty) and you will see. The Helmholtz resonator can be designed via a relatively simple formula to have any resonant frequency you choose. In the case of the bass reflex enclosure, the resonant frequency is set just at the point where an equivalently sized infinite baffle would be losing low end response. Thus, the resonance of the enclosure can assist the drive unit just at the
  44. 44. point where its output is weakening, this extending the low frequency response usefully. There is of course a cost to this. Whereas an infinite baffle loudspeaker can be designed with a low-Q resonance, meaning essentially that when the input ceases the diaphragm returns straight away to its rest position, in a bass reflex loudspeaker the drive unit will overshoot the rest position and then return. Depending on the quality of the design, it may do this more than once creating an audible resonance. This can result in so- called 'boomy' bass, which is generally undesirable. Additionally, a loudspeaker with boomy bass will tend to translate any low frequency energy into output at the resonant frequency. This a carefully tuned and recorded kick drum will come out as a boom at the loudspeaker's resonant frequency. The competent loudspeaker designer is in control of this and a degree boominess will be balanced against a subjectively 'good' - if not accurate - bass response. There are other cabinet designs, notably the transmission line, but these are not generally within the scope of professional sound engineer so they will be excluded from this text. Horns We have covered horns to some degree already. There is a whole theory to horns that deserves consideration, but here we will simply list some of the basics: Whereas a direct radiator drive unit may be only 1% efficient (i.e. 100 W of electrical power converts to just 1 W of sound power), a horn drive unit may be up to 5% efficient. The air in the throat of the horn becomes so compressed at high levels that significant distortion is produced. However, some people - including the writer of this text! - can on occasion find the distortion quite pleasant. To make any significant difference to the efficiency of a loudspeaker at low frequencies, the length and area of the horn have to be very large. However, folded horn cabinets can be constructed that make enough of a difference to be worthwhile. These are sometimes known as 'bass bins'.
  45. 45. The most important application of the horn is in high quality PA systems such as those used for theater musicals. The problem in theater musicals is that the sound has to be intelligible otherwise the story won't be understood by the audience (many of whom in a London West End theater would be European tourists who wouldn't have English as their first language). Also, the whole of the auditorium has to be covered with high quality sound. if director radiator loudspeakers were used in the theater, then people who were on-axis would received good quality sound. Those members of the audience who were further from the 'straight ahead' position would received lower levels at high frequency and therefore a duller sound. The solution is the constant directivity horn. (More information on directivity...). The shape of the curvature of the horn can be one of any number of mathematical functions, or even just an arbitrary shape. With careful calculation and design it is possible to produce a constant directivity horn which has an even frequency response over an angle of up to 60 degrees. This means that one loudspeaker can cover a sizable section of the audience, all with pretty much the same quality of sound. This leads to the concept of the center cluster loudspeaker system that is widely used wherever intelligibility is a prime requirement in a PA system. A number of constant directivity horn loudspeakers are arrayed so that where the coverage of one is just starting to fall off, the adjacent loudspeaker takes over. Next time you are in a theater, or large place of worship, with a quality sound system, take a look at the loudspeakers. Apart from any loudspeakers that are dedicated to bass, where directionality isn't significant, there should be one cabinet pointing almost directly at you, plus or minus 30 degrees or so, and there should be no other loudspeaker pointing at you from any other location in the building, other than for special theatrical effects. There will be more on this when we cover PA system specifically. Crossover The function of the crossover is to separate low, mid and high frequencies according to the number of drive units in the loudspeaker. A crossover can be passive or active. A passive crossover is generally internal to the cabinet and consists of a network capacitors, inductors and resistors. Having no active components, it doesn't need to be powered. An active crossover on the other hand does contain transistors or ICs and
  46. 46. requires mains power. It sits between the output of the mixing console and a number of power amplifiers - one for each division of the frequency band. A system with a three-band active crossover would require three power amplifiers. Crossovers have two principal parameter sets: the cut off frequencies of the bands, and the slopes of the filters. It is impractical, and actually undesirable, to have a filter that allows frequencies up to, say, 4 kHz to pass and then cut off everything above that completely. So frequencies beyond the cutoff frequency (where the response has dropped by 3 dB from normal) are rolled off at a rate of 6, 12, 18 or 24 dB per octave. In other words, in the band of frequencies where the slope has kicked in, as the frequency doubles the response drops by that number of decibels. The slopes mentioned are actually the easy ones to design. A filter with a slope of, say, 9 dB per octave would be much more complex. As it happens, a slope of 6 dB per octave is useless. High frequencies would be sent to the woofer at sufficient level that there would be audible distortion due to break up. Low frequencies would be sent to the tweeter that could damage it. 12 dB/octave is workable, but most systems these days use 18 dB/octave or 24 dB/octave. There are issues with the phase response of crossover filters that vary according to slope, but this is an advanced topic that few working sound engineers would contemplate to any great extent. Passive crossovers have a number of advantages: • Inexpensive • Convenient • Usually matched by the loudspeaker manufacturer to the requirements of the drive units • And the disadvantages: • Not practical to produce a 24 dB/octave slope • Can waste power • Not always accurate & component values can change over time Likewise, active crossovers have advantages: • Accurate • Cutoff frequency and slope can be varied
  47. 47. • Power amplifier connects directly to drive unit - no wastage of power & better control over diaphragm motion • Limiters can be built into each band to help avoid blowing drive units And the disadvantages: • Expensive • It is possible to connect the crossover incorrectly and send LF to the HF driver and vice versa. • A third-party unit would not compensate for any deficiencies in the driver units. Some loudspeaker systems come as a package with a dedicated loudspeaker control unit. The control unit consists of three components: • Crossover • Equalizer to correct the response or each drive unit • Sensing of voltage (and sometimes) current to ensure that each drive unit is maximally protected
  48. 48. Use of Loudspeakers As mentioned earlier, there are four main usage areas of loudspeakers: domestic, hi-fi, studio and PA. We will skip non-critical domestic usage and move directly on to hi-fi. The hi-fi market is significant in that this is where we will find the very best sounding loudspeakers. The living room environment is generally fairly small, and listening levels are generally well below what we call 'rock and roll'. This means that the loudspeaker can be optimized for sound quality, and the best examples can be very satisfying to listen to with few objectionable features, although it still has to be said that moving coil loudspeakers always sound like loudspeakers and never exactly like the original sound source. Recording studio main monitors have to be capable of higher sound levels. For one thing, the producer, engineer and musicians might just like to monitor at high level, although for the sake of their hearing they should not do this too often. Another consideration is that the acoustically treated control room will absorb a lot of the loudspeaker's energy, so that any given loudspeaker would seem quieter than it would in a typical living room. It is generally true that a loudspeaker that is optimized for high levels won't be as accurate as one that has been optimized for sound quality. PA speakers are the ultimate example of this. There has been a trend over the last couple of decades for PA speakers to be smaller and hence more cost effective to set up. This has resulted in an intense design effort to make smaller loudspeakers louder. Obviously the quality suffers. If you put an expensive PA loudspeaker next to a decent hi-fi loudspeaker in a head-to-head comparison at a moderate listening level, the hi-fi loudspeaker will win easily. The most fascinating use of loudspeakers is the near field monitor. Near field monitors are now almost universally used in the recording studio for general monitoring purposes and for mixing. This would seem odd because twenty-five years ago anyone in the recording industry would have said that studio monitors have to be as good as possible so that the engineer can hear the mix better than anyone else ever will. That way, all the detail in the sound can be assessed properly and any faults or deficiencies picked up. Mixes were also assessed on tiny Auratone loudspeakers just to make sure they would sound good on cheap domestic systems, radios or portables.
  49. 49. That was until the arrival of the Yamaha NS10 - a small domestic loudspeaker with a dreadful sound. It must have found its way into the studio as cheap domestic reference. A slightly upmarket Auratone if you like. However, someone must have used it as a primary reference for a mix, and found that by some magical an indefinable means, the NS10 made it easier to get a great mix - and not only that but a mix that would 'travel well' and sound good on any system. The NS10 and later NS10M are now no longer in production, but every manufacturer has a nearfield monitor in their range. Some actually now sound very good, although their bass response is lacking due to their small size. The success if nearfield monitoring is something of a mystery. It shouldn't work, but the fact is that it does. And since so little is quantifiable, the best recommendation for a nearfield monitor is that it has been used by many engineers to mix lots of big-selling records. That would be the Yamaha NS10 then!
  50. 50. Check Questions • What problem is caused by sound coming from the rear of the drive unit? • What is a baffle? • How large does a baffle have to be to work well at low frequencies? • What is an 'open back' cabinet? • What is an 'infinite baffle' cabinet? • What problem in an infinite baffle cabinet is caused by the trapped air inside? • What is 'compliance'? • What is a 'bass reflex' enclosure? • What is the advantage of a bass reflex loudspeaker compared to an infinite baffle? • What is the disadvantage of a bass reflex loudspeaker compared to an infinite baffle? • Briefly describe a horn drive unit in comparison with a direct radiator drive unit. • What is the advantage of the horn regarding efficiency? • What is the (greater) advantage of the constant directivity horn? • What is a 'center cluster'? • What is meant by the 'slope' of a crossover? • Contrast some of the principal features of active and passive crossovers. • Comment on the use of nearfield monitors
  51. 51. Chapter 5: Analog Recording Contrary to what you might read in home recording magazines, analog recording is not dead. Top professional studios still have analog recorders because they have a sound quality that digital just can't match. This isn't really to say that they sound better; in fact their faults are easily quantifiable, but their sound is often said to be 'warm', and it is often true to say that it is easier to mix a recording made on analog than it is to mix a digital multitrack recording. The other useful feature of analog recorders is that they are universal. You can take a tape anywhere and find a machine to play it on. As digital formats become increasingly diverse, individual studios become more and more isolated with audio being subject to an often complex export process to transfer it from one studio's system to another. With tape, you just mount the reel on the recorder and press play. History Magnetic tape recording was invented in the early years of the Twentieth Century and became useful as a device for recording speech, but simply for the information content, as in a dictation machine - the sound quality was too poor. In essence, a tape recorder converts an electrical signal to a magnetic record of that signal. Electricity is an easy medium to work in, compared to magnetism. It is straightforward to build an electrical device that responds linearly to an input. As we saw earlier, 'linear' means without distortion - like a flat mirror compared (linear) to a funfair mirror (non-linear). Magnetic material does not respond linearly to a magnetizing force. When a small magnetizing force is applied, the material hardly responds at all. When a greater magnetizing force is applied and the initial lack of enthusiasm to become magnetized has been overcome, then it does respond fairly linearly, right up to the point where it is magnetized as much as it can be, when we say that it is 'saturated'. Unfortunately, no-one has devised a way of applying negative feedback to analog recording, which in an electrical amplifier reduces distortion tremendously. Early tape recorders (and wire recorders) had no means of compensating for the inherent non-linearity of magnetic material, and it was left up to scientists in Germany during World War II to come up with a solution. The tape recorder was apparently used to broadcast orchestral concerts at
  52. 52. all hours of day and night, to the consternation of opposing countries who wondered how Germany could spare the resources to have orchestras playing in the middle of the night. (Obviously, recording onto disc was possible, but the characteristic crackle always gave the game away). After hostilities had ceased, US forces brought some captured machines back home and development continued from that point. There is a lot of history to the analog recorder, which we don't need here, but it is certainly interesting as the development of the tape recorder coincides with the development of recording as we know it now. The Sound of Analog There are three characteristic ingredients of the analog sound: • Distortion • Noise • Modulation noise • Distortion The invention that transformed the analog tape recorder from a dictation machine to a music recording device, during the 1940s, was AC bias. Since the response of tape to a small magnetizing force is very small, and the linear region of the response only starts at higher magnetic force levels, a constant supporting magnetic force, or bias, is used to overcome this initial resistance. Prior to AC bias, DC bias was used courtesy of a simple permanent magnet. However, considerable distortion remained. AC bias uses a high frequency (~100 kHz) sine wave signal mixed in with the audio signal to 'help' the audio signal get into the linear region which is relatively distortion-free. This happens inside the recorder and no intervention is required on the part of the user. However the level of the bias signal has to be set correctly for optimum results. In traditional recording, this is the job of the recording engineer before the session starts. It has to be said that line up is an exacting procedure and many modern recording engineers have so much else to think about (their digital transfers!) that line-up is better left to specialists. Despite AC bias, analog recording produces a significant amount of distortion. The higher the level you attempt to record on the tape, the more the distortion. It isn't like an amplifier or digital recorder where the signal is clean right up to 0 dBFS, then harsh clipping takes place. The
  53. 53. distortion increases gradually from barely perceptible to downright unpleasant. Most analog recordings peak at a level that will produce around 1% distortion, which is very high compared to any other type of equipment. At 3%, most engineers will be thinking about backing off. More is unacceptable. It may not sound promising to use a medium that produces so much distortion, but the fact is that it actually sounds quite pleasant! It is also different in character than vacuum tube (valve) distortion so it is an additional tool in the recording engineer's toolkit. Noise As well as producing more distortion than any other type of audio equipment, the analog tape recorder produces more noise too - a signal to noise ratio of around 65 dB is about the best you can hope for and represents the state of the art since tape recorders matured around the early 1970s. It is debatable whether noise is a desirable component of analog recording, but it is certainly a feature. Noise isn't really the ogre it is made out to be. If levels are set correctly to maximize the use of the available dynamic range up to the 1% or 3% distortion point, then there is no reason why it should be troublesome in the final mix, although some 'noise management' will be necessary of the part of the mix engineer. Modulation Noise There have been digital 'analog simulators', but to my ears, unless this aspect of the character of analog recorders is simulated, they just don't same the same. Modulation noise is noise that changes as the signal changes, and has two causes. One is Barkhausen noise which is produced by quantization of the magnetic domains (a gross over-simplification of a phenomenon that would take too much understanding for the working sound engineer to bother with). The other - more significant - cause of modulation noise is irregularities in the speed of tape travel. These irregularities are themselves caused by eccentricity and roughness in the bearings and other rotating parts, and by the tape scraping against the static parts. We some times hear of the term 'scrape flutter', which creates modulation noise, and the 'flutter damper roller', which is a component used to minimize the problem. If a 1 kHz sine wave tone is recorded onto analog tape, the output will consist of 1 kHz plus two ranges of other frequencies, some strong and
  54. 54. consistent, others weaker and ever-changing due to random variations. These are known in radio as 'sidebands' and the concept has exactly the same meaning here. Modulation noise, subjectively, causes a 'thickening' of the signal which accounts for the fat sound of analog, compared to the more accurate, but thin sound of digital. It has even been known for engineers to artificially increase the amount of modulation noise by unbalancing one of the rollers, thus creating more stronger sidebands containing a greater range of frequencies. Don't try it with your hard disk!
  55. 55. The Anatomy of the Analog Tape Recorder
  56. 56. The Studer A807 pictured here is typical of a workhorse stereo analog recorder, sold mainly into the broadcast market. Let's run through the major components starting from the ones you can't see: • Three motors, one each for the supply reel, take-up real and capstan. The take-up reel motor provides sufficient tension to collect the tape as it comes through. It does not itself pull the tape through. The supply reel motor is energized in the reverse direction to maintain the tension of the tape against the heads. • The capstan provides the motive force that drives the tape at the correct speed. • The pinch wheel holds the tape against the capstan. • The tach (short for tachometer) roller contains a device to measure the speed of the tape in play and fast wind. • The tension arm smooths out any irregularities in tape flow. • The flutter damper roller reduces vibrations in the tape, lessening modulation noise. • The erase head wipes the tape clean of any previous recording. • The record head writes the magnetic signal to the tape. It can also function as a playback head, usually with reduced high frequency response. • The playback head plays back the recording. Magnetic Tape Magnetic tape comprises a base film, upon which is coated a layer of iron oxide. Oxide of iron is sometimes, in other contexts, known as 'rust'. The oxide is bonded to the base film by a 'binder', which also lubricates the tape as it passes through the recorder. Other magnetic materials have been tried, but none suits analog audio recording better than iron, or more properly 'ferric' oxide. There are two major manufacturers of analog tape (there used to be several): Quantegy (formerly known as Ampex) and Emtec (formerly known as BASF).
  57. 57. Tape is manufactured in a variety of widths. (It is also manufactured in two thickness - so-called 'long play' tape can fit a longer duration of recording on the same spool, at the expense of certain compromises.). The widths in common use today are two-inch and half-inch. Oddly enough, metrication doesn't seem to have reached analog tape and we tend to avoid talking about 50 mm and 12.5 mm. Other widths are still available, but they are only used in conjunction with 'legacy' equipment which is being used until it wears out and is scrapped, and for replay or remix of archive material. Quarter-inch tape was in the past very widely used as the standard stereo medium, but there is now little point in using it as it has no advantages over other options that are available. Two-inch tape is used on twenty-four track recorders. A twenty-four track recorder can record - obviously - twenty-four separate tracks across the width of the tape, thus keeping instruments separate until final mixdown to stereo. Half-inch tape is used on stereo recorders for the final master. The speed at which the tape travels is significant. Higher speeds are better for capturing high frequencies as the recorded wavelength is physically longer on the tape. However, there are also irregularities (sometimes known as 'head bumps, or as 'woodles') in the bass end. The most common tape speed in professional use used to be 15 inches per second (38 cm/s), but these days it is more common to use 30 ips (76 cm/s), and not care about the massive cost in tape consumption! At 30 ips, a standard reel of tape costing up to $150 lasts about sixteen minutes.
  58. 58. Analog Recorders in Common Use Otari MTR90 Mk III There have been many manufacturers of analog tape recorders, but the top three historically have been Ampex, Otari and Studer. In the US, you will commonly find the Ampex MM1200 and occasionally the Ampex ATR124, which is often regarded as the best analog multitrack ever made, but Ampex only made fifty of them. All over the world you will find the Otari MTR90 (illustrated with autolocator) which is considered
  59. 59. to be a good quality workhorse machine, and is still available to buy. The Studer range is also well respected. The Studer A80 represents the coming of age of analog multitrack recording in the 1970s. It has a sound quality which is as good as the best within a very fine margin, but operational facilities are not totally up to modern standards. For example, it will not drop out of record mode without stopping the tape. The Studer A800 is still a prized machine and is fully capable, sonicly and operationally, of work to the highest professional standard. The more recent A827 and A820 are also very good, but sadly no longer manufactured. Multitrack Recording Techniques How to set about a multitrack recording session is a topic in itself and will be explained later. However, there are certain points of relevance to the equipment itself. The first is the necessity to be able to listen to or monitor previously recorded tracks while performing an overdub. The problem here is that there is a gap between the record head and the playback head. If the singer, for example, sings in time with the output from the playback head, the signal will be recorded on the tape a couple of centimeters away, therefore causing a delay. To get around this problem, while overdubbing, the record head is used as a playback head. In this situation we talk about taking a 'sync output' from the record head. The sync output isn't of such good sound quality since the record head is optimized for recording, nevertheless it is certainly good enough for monitoring. The playback head is used for final mixdown. Also, it is commonplace to 'bounce' several tracks, perhaps vocal harmonies, to one or two tracks (two tracks for stereo), thus freeing up tracks for further use. This has to be done using the sync output of the record head, otherwise the bounce won't be in time with the other tracks. The slight loss of quality has to be tolerated. Another technique worth mentioning at this stage is editing. As soon as tape was invented, people were cutting it apart and sticking it back together again. In fact, with the old wire recorders, people used to weld the wire together, although the heat killed the magnetism at the join. The most basic form of tape editing is 'top and tailing'. This means cutting the tape to within 10 mm or so of the start of the audio, and splicing in a section of leader tape, usually white (about two meters). Likewise the
  60. 60. tape is cut ten seconds or so after the end of each track and more leader inserted between tracks. At the end of the tape, red leader is joined on. No blank tape is left on the spool once top and tailing is complete. Editing can also be used to improve a performance by cutting out the bad and splicing in the good. Even two inch tape can be edited, in fact it is normal to record three or four takes of the backing tracks of a song, and splice together the best sections. The tape is placed in a special precision- machined aluminum editing block, and cut with a single-sided razor blade, guided by an angled slot. Splicing tape is available with exactly the right degree of stickiness to join the tape back together. When the edit is done in the right place (usually just before a loud sound), it will be inaudible. It takes courage to cut through a twenty-four track two-inch tape though. Compared to modern disk recorders, the main limitation of tape-based multitrack - analog and digital - is that once they are recorded, all the tracks have a fixed relationship in time. In a disk recorder, it is easy to move one track backwards or forwards in time, or copy it to a new location in the song. The equivalent technique in tape-based multitrack recording is the 'spin in'. In the original sense of the term, a good version of the chorus, or whatever audio was required to be repeated, would be copied onto another tape recorder. The multitrack would be wound to where the audio was to be copied. The two machines would be backed up a little way, then both set into play. At the right moment, the multitrack would be punched into record. Of course, the two machines had to be in sync, and this was the difficult part. If the two machines were identical mechanically, then a wax pencil mark could be made on corresponding rotating tape guides and the tapes backed up by the same number of revolutions. It sounds hit and miss, but it could be made to work amazingly quickly. When the digital sampler became available, it was used in place of the second recorder. Maintenance There is a difference between the maintenance of an analog recorder and a digital recorder. Firstly you can do a lot of first-line maintenance on an analog machine. You can't do more than run a cleaning tape on a digital recorder. The second is that you have to do the maintenance, otherwise performance will suffer. These are the elements of maintenance:
  61. 61. Cleaning: the heads and all metallic parts that the tape contacts are cleaned gently with a cotton bud dipped in isopropyl alcohol. Isopropyl alcohol is only one of a number of alcohol variants, and it has good cleaning properties. It is not the same as drinking alcohol, so don't be tempted. Also, drinking alcohol - ethanol - attracts additional taxes in some countries, therefore it would not be cost-effective to use it. The pinch wheel is made of a rubbery plastic. In theory it shouldn't be cleaned with isopropyl alcohol, but it often is. You can buy special rubber cleaner from pro audio dealers but in fact you can use a mild abrasive household liquid cleaner. Just one tiny drop is enough. Demagnetizing the heads: After a while, the metal parts will collect a residual magnetism that will partially erase any tape that is played on the machine. A special demagnetizer is used for which proper training is necessary, otherwise the condition can be made even worse. Line-up: Line up, or alignment, has two functions - one is to get the best out of the machine and the tape; the other is to make sure that a tape played on one recorder will play properly on any other recorder. The following parameters are aligned to specified or optimum values: Azimuth - the heads need to be absolutely vertical with respect to the tape otherwise the will be cancellation at HF. The other adjustments of the head - zenith, wrap and height are not so critical and therefore do not need to be checked so often. Bias level - optimizes distortion, maximum output level and noise. Playback level - the 1 kHz tone on a special calibration tape is played and the output aligned to the studio's electrical standard level. High frequency playback EQ - the 10 kHz tone on the calibration tape is played and the HF EQ adjusted. Record level - a 1 kHz tone at the studio's standard electrical level is recorded onto a blank tape and the record level adjusted for unity gain. HF record EQ - adjusted for flat HF response. LF record EQ - adjusted for flat LF response.
  62. 62. The line-up procedure used to be considered part of the engineer's day-to- day routing, but is now often left to a specialist technician. To conclude, this is certainly far from a complete treatise on analog tape recording, but it is enough for a starting point considering that analog recorders are now quite rare. Even so, analog recording has a long history and will almost certainly have a long future ahead. In fact the machines are so simple and are infinitely maintainable - a fifteen year old Studer A800 will still be working for its living in fifteen years time. You can't say that for digital recorders. Also, the sound of analog is very much the sound of recording, as we understand it. Does it make sense therefore to use digital emulation to achieve a pale shadow of the analog sound, or would it be better to use the real thing?
  63. 63. Check Questions • Give two reasons why analog recorders are still in use in top professional studios. • Comment on distortion in analog recording. • Comment on noise in analog recording. • Comment on modulation noise in analog recording. • What is the function of AC bias? • What is the distortion level of peaks in an analog recording? • Why is the concept of clipping not relevant in analog recording? • Why is the supply reel motor driven in the opposite direction to the actual rotation of the reel? • What is the capstan? • What is the pinch wheel? • What is the tach roller? • What two tape widths are in common top-level professional use? • Name three twenty-four track analog tape recorders, make and model. • What is 'bouncing'? • Comment on cut and splice tape editing. • What are the two functions of line-up?
  64. 64. Chapter 6: Digital Audio Why digital? Why wasn't analog good enough? The answer starts with the analog tape recorder which plainly isn't good enough in respect of signal to noise ratio and distortion performance. Many recording engineers and producers like the sound of analog now, because it is a choice. In the days before digital, analog recording wasn't a choice - it was a necessity. You couldn't get away from the problems. Actually you could. With Dolby A and subsequently SR noise reduction, noise performance was vastly improved, to the point where it wasn't a problem at all. And if you don't have a problem with noise, you can lower the recording level to improve the distortion performance of analog tape. A recording well made with Dolby SR noise reduction can sound very good indeed. Some would say better than 16-bit digital audio, although this is from a subjective, not a scientific, point of view. Analog record also had the problem that when a tape was copied, the quality would deteriorate significantly. And often there were several generations of copies between original master and final product. Digital audio can be copied identically as many times as necessary (although this doesn't always work as well as you might expect. More on this in another module). In the domestic domain, before CD there was only the vinyl record. Well there was the compact cassette too, but that never even sounded good even with Dolby B noise reduction. (Some people say that they don't like Dolby B noise reduction. The problem is that they are usually comparing an encoded recording with decoding switched on and off. The extra brightness of the Dolby B encoded - but not decoded - sound compensates for dirty and worn heads and the decoded version sounds dull in comparison!). People with long memories will know that they used to yearn for a format that wasn't plagued with the clicks, pops and crackles of vinyl. The release of the CD format was eagerly anticipated, and of course the CD has become a great success. Done properly, digital audio recorders can greatly outperform analog in both signal to noise ratio and distortion performance. That is why they are used in both the professional and domestic domains. When the question arises of why the other parts of the signal chain have mostly been changed over to digital, any possible improvement in sound quality is hardly relevant. Everything else performs as well as anyone could possibly want. Well almost anyone, the only exceptions being the