pro engineer school vol.1

8/3/2019 Pro Engineer School Vol.1

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Volume 1


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Contents

3. Microphone Technology19. 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 Record-Producer.com

You are licensed to make as many copies as you reasonably require foryour own personal use.


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Chapter 1: Microphone Technology

The microphone is the front-end of almost all sound engineeringactivities and, as the interface between real acoustic sound travelling in

air and the sound engineering medium of electronics, receives animmense amount of attention. Sometimes one could think that the statusof the microphone has been raised to almost mythological proportions. Itis useful therefore to put things in their proper perspective: there are agreat many microphones available that are of professional quality.Almost any of them can be used in a wide variety of situations to recordor broadcast sound to a professional standard. Of course different makesand types of microphones sound different to each other, but thedifferences don't make or break the end product, at least as far as thelistener is concerned.

Now, if you want to talk about something that really will make or breakthe end product, that is how microphones are used. Two sound engineersusing the same microphones will instinctively position and direct themdifferently and there can be a massive difference in sound quality. Givethese two engineers other mics, whose characteristics they are familiarwith, and the two sounds achieved will be identifiable according toengineer, and not so much to according to microphone type.

There are two ways we can consider microphones, by construction andby directional properties. Let's look at the different ways a microphonecan 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 besaid, has evolved into a very specialized animal, but it is still commonlyfound under the bridge of an electro-acoustic guitar so it is worthknowing about.

Piezoelectric

The piezoelectric effect is where certain crystalline and ceramic materialshave the property of generating an electric current when pressure or abending force is applied. This makes them sensitive to acoustic vibrationsand they can produce a voltage in response to sound. Piezo mics (or


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transducers as they may be called - a transducer is any device thatconverts one form of energy to another) are high impedance. This meansthat they can produce voltage but very little current. To compensate forthis, a preamplifier has to be placed very close to the transducer. This

will usually be inside the body of the electro-acoustic guitar. The preampwill run for ages on a 9 volt alkaline battery, but it is worth rememberingthat if an electro-acoustic guitar, or other instrument with a piezotransducer, sounds distorted, it is almost certainly the battery that needsreplacing, perhaps after a year or more of service.

Dynamic

This is dynamic as in dynamo. The dynamo is a device for convertingrotational motion into an electric current and consists of a coil of wire

that rotates inside the field of a magnet. Re-configure these componentsand you have a coil of wire attached to a thin, lightweight diaphragm thatvibrates in response to sound. The coil in turn vibrates within the field ofthe magnet and a signal is generated in proportion to the acousticvibration the mic receives. The dynamic mic is also sometimes known asthe moving coil mic, since it is always the coil that moves, not themagnet - 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, andvice versa, using a transformer. All professional dynamic micsincorporate a transformer that gives them an output impedance ofsomewhere around 200 ohms. This is a fairly low output impedance thatcan drive a cable of 100 meters or perhaps even more with little loss ofhigh frequency signal (the resistance of a cable attenuates all frequenciesequally, the capacitance of a cable provides a path between signalconductor and earth conductor through which high frequencies canleak). It is not necessary therefore to have a preamplifier close to the

microphone, neither does the mic need any power to operate. Examplesof dynamic mics are the famous Shure SM58 and the Electrovoice RE20.The characteristics of the dynamic mic are primarily determined by theweight of the coil slowing down the response of the diaphragm. Thesound can be good, particularly on drums, but it is not as crisp and clearas it would have to be to capture delicate sounds with complete accuracy.Dynamic microphones have always been noted for providing good value


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for money, but other types are now starting to challenge them on thesegrounds.

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. When the ribbon vibrates inresponse to sound it acts as a coil, albeit a coil with only one turn. Sincethe ribbon is very light, it has a much clearer sound than the conventionaldynamic, and it is reasonable to say that many engineers could identifythe 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 doeshowever also have a low impedance and provides a current which theintegral transformer can step up so that the voltage output of a modernribbon mic can be comparable with a conventional dynamic. Examples ofribbon 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 isparalleled by a backplate. Together they form the plates of a capacitor.A capacitor, of any type, works by storing electrical charge. Electricalcharge can be thought of as quantity of electrons (or the quantity ofelectrons that normally would be present, but aren't). The greater thedisparity in number of electrons present i.e. the amount of charge thehigher will be the voltage across the terminals of the capacitor. There isthe equation:


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Q = C x V

or:

charge = capacitance x voltage

Note that charge is abbreviated as Q, because C is already taken bycapacitance.

Putting this another way round:

V = Q/C

or:

voltage = charge / capacitance

Now the tricky part: capacitance varies according to the distance betweenthe plates of the capacitor. The charge, as long as it is either continuouslytopped up or not allowed to leak away, stays constant. Therefore as thedistance between the plates is changed by the action of acousticvibration, the capacitance will change and so must the voltage betweenthe plates. Tap off this voltage and you have a signal that represents thesound hitting the diaphragm of the mic.

Sennheiser MKH 40

The great advantage of the capacitor mic is that the diaphragm isunburdened by a coil of any sort. It is light and very responsive to themost delicate sound. The capacitor mic is therefore much more accurateand faithful to the original sound than the dynamic. Of course there is a


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downside too. This is that the impedance of the capsule (the part of anymic that collects the sound) is very high. Not just high - very high. It alsorequires continually topping up with charge to replace that whichnaturally leaks away to the atmosphere. A capacitor mic therefore needs

power for these two reasons: firstly to power an integral amplifier, andsecondly 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 comeacross them from time to time. Modern capacitor mics use phantompower. Phantom power places +48 V on both of the signal carryingconductors of the microphone cable actually within the mixing console orremote preamplifier, and 0 V on the earth conductor. So, simply byconnecting a normal mic cable, phantom power is connectedautomatically. That's why it is called phantom because you don't seeit! In practice this is no inconvenience at all. You have to remember toswitch 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 ofphantom power in any way, One operational point that is importanthowever is that the fader must be all the way down when a mic isconnected to an input providing phantom power, or when phantom poweris switched on. Otherwise a sharp crack of speaker-blowing proportionsis produced.

A capacitor microphone often incorporates a switched -10 dB or -20 dBpad, which is an attenuator placed between the capsule and the amplifierto prevent clipping on loud signals.

Electret

The electret mic is a form of capacitor microphone. However the chargeis permanently locked into the diaphragm and backplate, just as magneticenergy is locked into a magnet. Not all materials are suited to formingelectrets, so it is usually considered that the compromises involved inmanufacture compromise sound quality. However, it has to be said thatthere are some very good electret mics available, most of which are back-electrets, meaning that only the backplate of the capacitor is an electrettherefore the diaphragm can be made of any suitable material. Electretmics do still need power for the internal amplifier. However, this cantake the form of a small internal battery, which is sometimes convenient.


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Electret mics that have the facility for battery power can also usually bephantom powered, in case the battery runs down or isnt fitted.


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the only difference being that the rear produces an inverted signal, 180degrees 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 perfectcardioid at mid frequencies, but at low frequencies the pattern will spreadout into omni. At high frequencies the pattern will tighten intohypercardioid. The significant knock-on effect of this is that thefrequency response off-axis in other words any direction but head on is never flat. In fact the off-axis response of most microphones is nothingshort of terrible and the best you can hope for is a smooth roll-off ofresponse from LF to HF. Often though it is very lumpy indeed. We willsee 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 soundwave. The diaphragm is completely enclosed, apart from a tiny slow-acting air-pressure equalizing vent, and the mic effectively compares thechanging pressure of the outside air under the influence of the soundsignal with the constant pressure within. Pressure acts equally in alldirections, therefore the mic is equally sensitive in all directions, in

theory as we said. In practice, at higher frequencies where the size of themic starts to become significant in comparison with the wavelength, thediaphragm will be shielded from sound approaching from the rear andrearward HF response will drop.

Figure-of-Eight

At the other end of the spectrum of polar patterns the figure-of-eightmicrophone is sensitive to the pressure gradient of the sound wave. Thediaphragm is completely open to the air at both sides. Even though it isvery light and thin, there is a difference in pressure at the front and rearof the diaphragm, and the microphone is sensitive to this difference. Thepressure gradient is greatest for sound arriving directly from the front orrear, and lessens as the sound source moves round to the side. When thesound source is exactly at the side of the diaphragm it produces equalpressure at front and back, therefore there is no pressure gradient and themicrophone produces no output. Therefore the figure-of-eightmicrophone is not sensitive at the sides. (You could also imagine that a


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Multipattern Microphones

There are many microphones available that can produce a selection ofpolar patterns. This is achieved by mounting two diaphragms back-to-

back with a single central backplate. By varying the relative polarizationof the diaphragms and backplate, any of the four main polar patterns canbe created. It is often thought that the best and most accuratemicrophones are the true omnidirectional and the true figure-of-eight,and that mimicking these patterns with a multipattern mic is less thenoptimal. Nevertheless, in practice multipattern mics are so versatile thatthey are commonly the mic of first choice for many engineers.

AKG C414


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Special Microphone Types

Stereo Microphone

Two capsules may be combined into a single housing so that one mic cancapture both left and right sides of the sound field. This is much moreconvenient than setting two mics on a stereo bar, but obviously lessflexible. Some stereo mics use the MS principle where one cardioidcapsule (M) captures the full width of the sound stage while the otherfigure-of-eight capsule (S) captures the side-to-side differences. The MSoutput 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 similarityin appearance to a gun barrel. The slots in the barrel allow off-axis soundto cancel giving a highly directional response. The longer the mic, themore directional it is. The sound quality of these microphones is inferiorto normal mics so they are only used out of necessity.

Sennheiser interference tube microphone


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as effectively as it does in a microphone with a larger diaphragm.Miniature microphones therefore have to be used close to the soundsource; 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 lessaccurate 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 theirenhanced 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 boomstands, tall stands up to 4 meters for orchestral recording, fishpoles asused by video and film sound recordists, and long booms with cable

operated mic positioning used in television studios. Attaching the mic tothe stand is a mount that can range from a basic plastic clip, to an elasticsuspension or cradle that will isolate the microphone from floor noise.

The other major accessory is the windshield or pop-shield. A windshieldmay be made out of foam and slipped over the mic capsule, or it maylook like a miniature airship covered with wind-energy dissipatingmaterial. For blizzard conditions windshield covers are available that


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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 usean 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 dynamicmicrophone?

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 (tworeasons)?

What precaution should you take when switching on phantompower?

Can dynamic microphones of professional quality be used withphantom 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?


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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.


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This allows the speaker to turn his or her head and still receive adequatepickup. Unfortunately, as the head moves, both microphones can pick upthe sound while the sound source the mouth is moving towards onemic and away from the other. The Doppler effect comes into play and

two slightly pitch shifted signals are momentarily mixed together. Itsounds neither pleasant nor natural. The alternative approach is to mountboth microphones centrally and use one as a backup. The speaker willlearn, through not hearing their voice coming back through the PAsystem, 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 ableto hear their amplified voice at the right level. If their voice seems tooloud, to them, they will instinctively back away from the mic. If theycant hear their amplified voice they will assume the system isntworking. I once saw the chairman of a large and prestigious organisationstand away from his microphone because he thought it wasnt working. Ithad been, and at the right level for the audience. But unfortunately, apartfrom the front few rows, they were unable to hear a single unamplifiedword he said.


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who reportedly wore a large ruby from her neck, has long gone. Thechest position is great for newsreaders but it suffers from the shadow ofthe chin and boominess caused by chest resonance. The best place for aminiature 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 notvisible to the audience beyond the second or third row. If a boom is notconsidered acceptable, then the mic may protrude a short distance fromabove the ear, or descending from the hairline. This actually captures avery good vocal sound. It has to be tried to be believed. One of thebiggest problems with miniature microphones in the theatre is that theybecome sweated out after a number of performances and have to bereplaced. Still, no-one said that it was easy going on stage. For theorchestra in a theatre musical, clip on mics are good for stringinstruments. Wind instruments are generally loud enough forconventional stand mics, closely placed. So-called booth singers canuse conventional mics.


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Stereo Microphone Techniques

Firstly, what is stereo? The word stereophonic in its original meaning itsuggests a solid sound image and does not specify how many

microphones, channels or loudspeakers are to be used. However, it hascome to mean two channels and two loudspeakers using as few or asmany microphones that are necessary to get a good result. When itworks, you should be able to sit in an equilateral triangle with thespeakers, listen to a recording of an orchestra and pinpoint where everyinstrument is in the sound image. (By the way, some people complainthat stereophonic, as a word, combines both Greek and Latin roots. Justas well perhaps, because if it had been exclusively Latin it would havebeen crassophonic!)

When recording a group of instruments or singers, it is possible to usejust 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 ofthe sound stage (and, due to the rear pickup of the figure-of-eight mic, tothe left and right of the area where the audience would be also). Morepractically, two cardioid microphones can be used. They would be angledat 120 degrees were it not for the drop off in high frequency response atthis angle in most mics. A 110-degree angle of separation is a reasonablecompromise. This system was originally proposed in the 1930s andmathematically inclined audio engineers will claim that this gives perfectreproduction of the original sound field from a standard pair of stereoloudspeakers. However perfect the mathematics look on paper, the resultsdo not bear out the theory. The sound can be good, and you can with


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effort tell where the instruments are supposed to be in the sound image.The problem is that you just dont feel like you are in the concert hall, orwherever the recording was made. The fact that human beings do nothave 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 itsounds a whole lot better.

Near-coincident crossed pair

The ORTF system, named for the Office de Radiodiffusion TelevisionFrancaise, uses two cardioid microphones spaced at 17 cm angledoutwards at 110 degrees, and is simply an extended near-coincidentcrossed pair.


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the left and right signals closer to the centre. Therefore it is in reality noadvantage at all.

Binaural stereo attempts to mimic the human hearing system with a

dummy head (sometimes face, shoulders and chest too) with twoomnidirectional microphones placed in artificial ears just like a realhuman head. It works well, but only on headphones. A binaural recordingplayed on speakers doesnt work because the two channels mix on theirway to the listener, spoiling the effect. There have been a number ofsystems attempting to make binaural recordings work on loudspeakersbut none has become popular.

In addition to the stereo miking system, it is common to mic up everysection 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, withthe other microphones used to compensate for the distance to the rear ofthe orchestra, and to add just a little presence to instruments whereappropriate. Sectional mics shouldnt be used to compensate for poorbalance due to the conductor or arranger. Sometimes however classicalcomposers dont get the balance quite right and it is not acceptable tochange the orchestration. A little technical help is therefore called for.

Instruments

We come back to the two golden rules of microphone placement, asabove. It is worth looking at some specific examples:

Saxophone

There are two fairly obvious ways a saxophone can be close miked. Oneis close to the mouthpiece, another is close to the bell. The difference insound 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 toposition microphones for all manner of instruments, but the key is toexperiment 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 thenatural sound of the instrument, if thats what you want. Close micpositions almost never do. If you move the mic further away, up to


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The kick drum should have its front head removed, or there should be alarge 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 thudthat 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 distanceinside the drum.

The snares on the underside of the snare drum may rattle when otherdrums are being played. Careful adjustment of the tension of the snares isnecessary, and perhaps even a little damping.

Microphones should be spaced as far apart from each other as possibleand directed away from other drums. Every little bit helps as thecombination of two mics picking up the same drum from different

distances leads to cancellation of groups of frequencies. The brute forcetechnique is to use a noise gate on every microphone channel, and this iscommonly done. Noise gates will be covered later.

Perhaps this is a brief introduction to the use of microphones, but its astart. And to round off Ill give away the secret of getting good soundfrom your microphones:

Listen!


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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 arigid piston.

The diaphragm could be flat and still produce sound. However, since themotor is at the center and vibrations are transmitted to the edges, thediaphragm needs to be stiff. The cone shape is the best compromisebetween stiffness and large diameter.

High frequencies will tend to bend the diaphragm more than lowfrequencies. It takes a certain time for movement of the coil to propagate

to the edge of the diaphragm. Fairly obviously, at high frequencies thereisn't so much time and at some frequency the diaphragm will start todeviate from the ideal rigid piston.

200 mm is a good compromise. It will produce enough level at lowfrequency for the average living room, and it will produce reasonablydistortion-free sound up to around 4 kHz or so. When the diaphragmbends, it is called break up, due to the vibration breaking up into anumber of different modes. Break up, in this context, doesn't mean

severe distortion or anything like that. In fact most low frequency driveunits are operated well into the break up region. It is up to the designer toensure that the distortion created doesn't sound too unpleasant. By theway, it is often thought that a larger drive unit will operate down to lowerfrequencies. This isn't quite the right way to look at it. Any size of driveunit will operate down to as low a frequency as you like, but you need abig drive unit to shift large quantities of air at low frequency. At highfrequency, the drive unit vibrates backwards and forwards rapidly,moving air on each vibration. At low frequencies there are feweropportunities to move air, therefore the area of the drive unit needs to begreater 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 werenot particularly stiff. Modern drive units use plastic diaphragms, or pulpdiaphragms that have been doped to stiffen them adequately. Of course,the ultimate in stiffness would be a metal diaphragm. Unfortunately, itwould be heavy and the drive unit would be less efficient. Carbon fiber


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may break apart the turns of the coil, or shift it from its central positionwith respect to the magnet (mechanical damage). The drive unit willstill function, but the coil will scrape against the magnet producing a veryharsh 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 theloudspeaker. It will tell you when it is under stress if you listen carefullyenough.

One common question regarding damage to loudspeakers is this: Whatshould the power of the amplifier be in relation to the rated power of theloudspeaker? In fact, although the power of an amplifier can be measuredvery accurately, the capacity of a loudspeaker to soak up this power isonly an intelligent guess, at best. During the design process, themanufacturer will test drive units to destruction and arrive at a balancebetween a high rating (in watts) that will impress potential buyers, and alow number of complaints from people who have pushed their purchasestoo hard. The rating on the cabinet is therefore only a guide. To get thebest performance from a loudspeaker, the amplifier should be ratedhigher in terms of watts. It wouldn't be unreasonable to connect a 200 Wamplifier to a 100 W speaker, and it won't blow the drive units unless youpush 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 a200 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 Wloudspeaker, the tweeter could be rated at as little as 20-30 W, as undernormal circumstances that is all it would be expected to handle. Butunder clipping conditions the level supplied to the tweeter could bemassively higher, and it will blow.

Impedance

Drive units and complete loudspeaker systems are also rated in terms oftheir impedance. This is the load presented to the amplifier, where a lowimpedance means the amplifier will have to deliver more current, andhence work harder. A common nominal impedance is 8 ohms.Nominal means that this is averaged over the frequency range of thedrive unit or loudspeaker, and you will find that the actual impedancedeparts significantly from nominal according to frequency. Normally thisisn't particularly significant, except in two situations:


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At some frequency the impedance drops well below the nominalimpedance. The power amplifier will be called upon to deliver perhapsmore power than it is capable of, causing clipping, or perhaps theamplifier might even go into protection mode to avoid damage to itself.

The output impedance of a power amplifier is very low just a smallfraction of an ohm. You could think of the output impedance of theamplifier in series with the impedance of the loudspeaker as a potentialdivider. Work out the potential divider equation with R1 equal to zeroand you will see that the output voltage is equal to the input voltage.However, give R1 some significant impedance, as would happen with along run of loudspeaker cable, and you will see a voltage loss. Make R2 -the loudspeaker impedance - variable with frequency and you will nowsee a rather less than flat frequency response.

To be honest, the above points are not always at the forefront of theworking sound engineer's mind, but they are significant and worthknowing about.


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At this point it is worth saying that the bare drive unit is often used intheater sound systems where there is a need for extreme clarity in thehuman vocal range. Low frequencies can be bolstered with conventionalcabinet loudspeakers.

Despite these problems, careful design of the drive unit to balance thespringiness of the trapped air inside the cabinet against the springiness ofthe 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 compromisesthat have to be made to make this design work result in poor lowfrequency 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 (startingwith the Thiele-Small parameters of drive units) to be able to design aloudspeaker that would work well for studio or PA use. Electric guitarloudspeakers are not so critical.

The next step in cabinet design is the bass reflex enclosure. You willoccasionally hear of this as a ported or vented cabinet.

The bass reflex cabinet borrows the theory of the Helmholtz resonator. AHelmholtz resonator is nothing more than an enclosed volume of airconnected to the outside world by a narrow tube, called the port. The portcan stick out of the enclosure as in a beer bottle - a perfect example of theprinciple - or inwards. The small plug of air in the port bounces againstthe compliance of the larger volume of air inside and resonates readily.

Try blowing across the top of the beer bottle (when empty) and you willsee.

The Helmholtz resonator can be designed via a relatively simple formulato have any resonant frequency you choose. In the case of the bass reflexenclosure, the resonant frequency is set just at the point where anequivalently sized infinite baffle would be losing low end response.Thus, the resonance of the enclosure can assist the drive unit just at the


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requires mains power. It sits between the output of the mixing consoleand a number of power amplifiers - one for each division of thefrequency band. A system with a three-band active crossover wouldrequire three power amplifiers.

Crossovers have two principal parameter sets: the cut off frequencies ofthe bands, and the slopes of the filters. It is impractical, and actuallyundesirable, to have a filter that allows frequencies up to, say, 4 kHz topass and then cut off everything above that completely. So frequenciesbeyond the cutoff frequency (where the response has dropped by 3 dBfrom normal) are rolled off at a rate of 6, 12, 18 or 24 dB per octave. Inother words, in the band of frequencies where the slope has kicked in, asthe frequency doubles the response drops by that number of decibels. Theslopes mentioned are actually the easy ones to design. A filter with aslope of, say, 9 dB per octave would be much more complex.

As it happens, a slope of 6 dB per octave is useless. High frequencieswould be sent to the woofer at sufficient level that there would be audibledistortion due to break up. Low frequencies would be sent to the tweeterthat could damage it. 12 dB/octave is workable, but most systems thesedays use 18 dB/octave or 24 dB/octave. There are issues with the phaseresponse of crossover filters that vary according to slope, but this is anadvanced 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


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Power amplifier connects directly to drive unit - no wastage ofpower & better control over diaphragm motion

Limiters can be built into each band to help avoid blowing driveunits

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 dedicatedloudspeaker 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


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all hours of day and night, to the consternation of opposing countries whowondered how Germany could spare the resources to have orchestrasplaying in the middle of the night. (Obviously, recording onto disc waspossible, but the characteristic crackle always gave the game away).

After hostilities had ceased, US forces brought some captured machinesback home and development continued from that point. There is a lot ofhistory to the analog recorder, which we don't need here, but it iscertainly interesting as the development of the tape recorder coincideswith 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 dictationmachine 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 forcelevels, a constant supporting magnetic force, or bias, is used to overcomethis initial resistance. Prior to AC bias, DC bias was used courtesy of asimple permanent magnet. However, considerable distortion remained.AC bias uses a high frequency (~100 kHz) sine wave signal mixed inwith the audio signal to 'help' the audio signal get into the linear regionwhich is relatively distortion-free. This happens inside the recorder andno intervention is required on the part of the user. However the level ofthe bias signal has to be set correctly for optimum results. In traditionalrecording, this is the job of the recording engineer before the sessionstarts. It has to be said that line up is an exacting procedure and manymodern recording engineers have so much else to think about (theirdigital transfers!) that line-up is better left to specialists.

Despite AC bias, analog recording produces a significant amount ofdistortion. The higher the level you attempt to record on the tape, themore the distortion. It isn't like an amplifier or digital recorder where thesignal is clean right up to 0 dBFS, then harsh clipping takes place. The


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The Anatomy of the Analog Tape Recorder


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The Studer A807 pictured here is typical of a workhorse stereo analogrecorder, sold mainly into the broadcast market. Let's run through themajor components starting from the ones you can't see:

Three motors, one each for the supply reel, take-up real andcapstan. The take-up reel motor provides sufficient tension tocollect the tape as it comes through. It does not itself pull the tapethrough. The supply reel motor is energized in the reverse directionto maintain the tension of the tape against the heads.

The capstan provides the motive force that drives the tape at thecorrect 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 frequencyresponse.

The playback head plays back the recording.Magnetic Tape

Magnetic tape comprises a base film, upon which is coated a layer of ironoxide. Oxide of iron is sometimes, in other contexts, known as 'rust'. Theoxide is bonded to the base film by a 'binder', which also lubricates thetape as it passes through the recorder. Other magnetic materials havebeen tried, but none suits analog audio recording better than iron, or moreproperly 'ferric' oxide. There are two major manufacturers of analog tape(there used to be several): Quantegy (formerly known as Ampex) andEmtec (formerly known as BASF).


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Cleaning: the heads and all metallic parts that the tape contacts arecleaned gently with a cotton bud dipped in isopropyl alcohol. Isopropylalcohol is only one of a number of alcohol variants, and it has goodcleaning properties. It is not the same as drinking alcohol, so don't be

tempted. Also, drinking alcohol - ethanol - attracts additional taxes insome 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 becleaned with isopropyl alcohol, but it often is. You can buy specialrubber cleaner from pro audio dealers but in fact you can use a mildabrasive household liquid cleaner. Just one tiny drop is enough.

Demagnetizing the heads: After a while, the metal parts will collect aresidual magnetism that will partially erase any tape that is played on the

machine. A special demagnetizer is used for which proper training isnecessary, otherwise the condition can be made even worse.

Line-up: Line up, or alignment, has two functions - one is to get the bestout of the machine and the tape; the other is to make sure that a tapeplayed on one recorder will play properly on any other recorder. Thefollowing 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 ofthe head - zenith, wrap and height are not so critical and therefore do notneed 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 andthe 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 isrecorded 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.


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Chapter 6: Digital Audio

Why digital? Why wasn't analog good enough? The answer starts withthe analog tape recorder which plainly isn't good enough in respect of

signal to noise ratio and distortion performance. Many recordingengineers and producers like the sound of analog now, because it is achoice. In the days before digital, analog recording wasn't a choice - itwas a necessity. You couldn't get away from the problems. Actually youcould. With Dolby A and subsequently SR noise reduction, noiseperformance was vastly improved, to the point where it wasn't a problemat all. And if you don't have a problem with noise, you can lower therecording level to improve the distortion performance of analog tape. Arecording well made with Dolby SR noise reduction can sound very goodindeed. Some would say better than 16-bit digital audio, although this isfrom a subjective, not a scientific, point of view. Analog record also hadthe problem that when a tape was copied, the quality would deterioratesignificantly. And often there were several generations of copies betweenoriginal master and final product. Digital audio can be copied identicallyas many times as necessary (although this doesn't always work as well asyou might expect. More on this in another module).

In the domestic domain, before CD there was only the vinyl record. Wellthere was the compact cassette too, but that never even sounded good

even with Dolby B noise reduction. (Some people say that they don't likeDolby B noise reduction. The problem is that they are usually comparingan encoded recording with decoding switched on and off. The extrabrightness of the Dolby B encoded - but not decoded - soundcompensates for dirty and worn heads and the decoded version soundsdull in comparison!). People with long memories will know that theyused to yearn for a format that wasn't plagued with the clicks, pops andcrackles 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 inboth signal to noise ratio and distortion performance. That is why theyare used in both the professional and domestic domains. When thequestion arises of why the other parts of the signal chain have mostlybeen changed over to digital, any possible improvement in sound qualityis hardly relevant. Everything else performs as well as anyone couldpossibly want. Well almost anyone, the only exceptions being the


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multiplication, require 30 Megabytes per stereo minute. Of course,Megabytes are getting cheaper all the time. There is another problemhowever - data bandwidth. When recording onto a hard disk system,there is a certain data throughput rate beyond which the system will

struggle and possibly fail to record or playback properly. A standardmodern hard drive should be easily capable of achieving 24 tracks ofplayback under normal circumstances (the track count is affected, for onething, by the 'edit density' - the more short segments you cut the audiointo, and the more widely the data is physically separated on the disk, theharder it will be to play back). Try this at three times the data rate and thetrack count, or the reliability is bound to suffer. However, disks aregetting ever faster and most of the problems of this nature are in the past.Before long it will be possible to get virtually any number of tracks quiteeasily. It's worth a quick look at Digidesign's comments on hard diskspecifications to maximize track count.

Digital Interconnection

Digital interconnection comes in a number of standards, which aresummarized here:

AES/EBU

Also known as AES3 1985 (the year it was implemented) Standard for professional digital audio Supports up to 24-bit at any sampling rate Transmits 2 channels on a single cable Uses 110 ohm balanced twisted wire pair cables usually terminated

with XLR connectors Can use cables of length up to 100 meters Electrical signal level 5 volts Standard audio cables can be used for short distances but are not

recommended as their impedance may not be the standard 110ohm and reflections may occur at the ends of the cable

Data transmission at 48 kHz sampling rate is 3.072 Megabit/s (64xthe sampling rate)

Self clocking but master clocking is possibleS/PDIF


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Check Questions

To which type of sound engineering equipment was digital audiofirst applied?

In relation to the question above, why was this the most pressingneed?

What types of equipment are currently not available in digitalform?

Describe 'sampling rate'. What is the minimum sampling rate for a digital system capable of

reproduction up to 20 kHz (ignoring any 'safety margin').

What is 'aliasing'? What two sampling rates are most commonly used in digital

audio?

Describe quantization. What is the signal to noise ratio, in theory, of a digital system with

20-bit resolution?

Why is coding necessary? Give two reasons. Why does a digital to analog convertor need a filter? What is error correction? What is error concealment? What happens (or at least should happen) if an error is neither

corrected nor concealed?

How many Megabytes of data, approximately, are occupied by oneminute of CD-quality stereo digital audio?

Why, in a hard disk recording system, is it likely that fewer tracks can bereplayed simultaneously at the 24-bit/96 kHz standard, than at the CD-


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q u a l i t y 1 6 - b i t / 4 4 . 1 k H z s t a n d a r d ?


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Chapter 7: Digital Audio Tape Recording

The original purpose of DAT (Digital Audio Tape) was to be areplacement for the Compact Cassette (or simply 'cassette', as we now

know it). Since DAT was intended to be a consumer product right fromthe start, the cassette housing is very small, 73 x 54 mm and just 10.5mm thick. For professional users, this is rather too small, not just becauseit makes the cassette easier to lose, but because there will always be afeeling that DAT could have been a better system if there had been a bitmore space for the data. This would allow for error concealment to beminimized, and tracking tolerances could be such that a tape recorded onone recorder could be absolutely guaranteed to play properly on anyother. This is generally the case for professional machines, but notnecessarily so for semi-pro 'domestic' recorders.

Sony professional DAT

Having said that DATs size is a disadvantage for professional users, itreally is amazing how it achieves what it does working at microscopicdimensions. DATs full title, R-DAT, indicates that the system uses arotary head like a video recorder. Unlike analog tape which records thesignal along a track parallel to the edge of the tape, a rotary head recorder

lays tracks diagonally across the width of the tape. So even though thetape speed is just 8.15 millimeters/second, the actual writing speed is amassive 3.133 meters/second. The width of each track is 13.591millionths of a meter. Unlike an analog tape, the tracks are recordedwithout any guard band between them. In fact, the tracks are recorded byheads which are around 50% wider than the final track width and eachnew track partially overlaps the one before, erasing that section. Since thesame heads are used for recording and playback, this may seem to


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kHz. This converts the original analog audio signal to a stream of binarynumbers representing the changing level of the signal. But since thedimensions of the actual recording on the tape are so small, there is a lotof scope for errors to be made during the record/replay process, and if the

wrong digit comes back from the tape it is likely to be very much moreaudible than a drop-out would be on analog tape. Fortunately DAT, likethe Compact Disc, uses a technique called Double Reed-SolomonEncoding which duplicates much of the audio data, in fact 37.5%, in sucha way that errors can be detected, then either corrected completely orconcealed so that they are not obvious to the ear. If there is a really hugedrop-out on the tape, then the DAT machine will simply mute the outputrather than replay digital gibberish. As an extra precaution againstdropouts, another technique called interleaving is employed whichscatters the data so that if one section of data is lost, then there will beenough data beyond the site of the damage which can be used toreconstruct the signal.

The pulse code modulated audio data is recorded in the centre section ofeach diagonal track across the tape. There is other data too:

'ATF' signals allow for Automatic Track Finding which makessure that the heads are always precisely positioned over the centreof the track, even if the tape is slightly distorted and the track

curved. Sub Code areas allow extra data to be recorded alongside the audio

information. Not all of the capacity of the Sub Code areas is in useas yet, allowing for extra expansion of the DAT system. Those atpresent in use include:

A-time, which logs the time taken since the beginning of the tape P-time, which logs the time taken since the last Start ID. Start ID marks the beginning of each item; Skip ID tells the machine to go directly to the next Start ID, thus

performing an instant edit. End ID marks the end of the recording on the tape. There is also provision for SMPTE/EBU timecode


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The original Sony 3324, and recent 24-track machines, use the normaldensity geometry on 1/2 tape which allows twenty-four digital audiotracks, two analog cue tracks, a control track and a timecode track. (Thecue tracks are there so that audio can be made available in other thannormal play speed +/- normal varispeed). The tape speed at 44.1 kHz is70.01cm/s. The 3324 is totally two-way compatible with the larger 3348which can record forty-eight digital tracks on the same tape. To give anexample, you may start a project on a 3324, of any vintage, and then theproducer decides as the tracks fill up that he or she really needs moreelbow room for overdubs. So you hire a 3348, put the twenty-four track

tape on this and record another twenty-four tracks in the guard bands leftby the other machine. Continuing my (hypothetical) example, when it isdecided that the project is costing too much and going nowhere, theproducer is sacked and another one brought in who decides that the extratwenty-four tracks are unnecessary embellishments and the originaltracks, with a little touching up, are all that are required. Off goes the3348 back to the hire company, the tape - now recorded with forty-eighttracks - is placed back on the 3324 and the original twenty-four tracks aresuccessfully sweetened and mixed with not a murmur from the tracks that

are now not wanted. We are now accustomed to new products andsystems which offer new features yet are compatible with materialproduced on earlier versions. This must be audio history's only exampleof forward as well as reverse compatibility. It shows what thinking aheadcan achieve.

DASH Operation

The first thing you are likely to want to do with your new DASHmachine is of course to make a recording with it, but it would beadvisable to read the manual before pressing record and play. Some ofthe differences between digital and analog recording stem from the factthat the heads are not in the same order. On an analog recorder we areused to having three heads: erase, record and play. DASH doesn't need anerase head because the tape is always recorded to a set level ofmagnetism which overwrites any previous recordings without further


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intervention. So the first head that the tape should come across should bethe record head. Right?

Wrong. The first head is a playback head, which on an basic DASH

machine is followed a record head only. If this seems incorrect, you haveto remember that while analog processes take place virtuallyinstantaneously, digital operations take a little time. So if you imagineanalog overdubbing where the sync playback signal comes from therecord head itself, you can see why this won't work in the digital domain.There will be a slight delay while the playback signal is processed, andanother delay while the record signal is processed and put onto tape. 105milliseconds in fact, which corresponds to about 75 mm of tape. Toperform synchronous overdubs there has to be a playback head upstreamof the record head otherwise the multitrack recording process as we knowit just wont work. For most purposes two heads are enough, and a thirdhead is available as an option if you need it, and you'll need it if you wantto have confidence monitoring. (There are no combined record/playbackheads, by the way, all are fixed function).

On any digital recording medium the tape has to be formatted to be used.On DAT the formatting is carried out during recording, but on DASH itis often better to do it in advance. The machine can format whilerecording - in Advance Record mode - but this is best done in situations

where you will be recording the whole of the tape without stopping. Ifyou wish, you can pre format a tape but this obviously takes time. Youcan take comfort from the fact that it can be done in one quarter of realtime, and the machine will lay down timecode simultaneously.

Since there are different ways to format a tape and make recordings, the3342S has three different recording modes: Advance, Insert andAssemble. Advance mode is as explained above. Insert is for when youhave recorded or formatted the full duration of the material and you want

to go back and re-record some sections. Assemble is when you want toput the tape on, record a bit, play it back, record a bit more etc, as wouldtypically happen in classical sessions.

Converter Delay

The main text deals with some of the implications of delays causedby the process of recording digital signals onto tape and playingthem back again. There is another problem caused by delays in the


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A/D conversion itself. The convertors used in the Sony 3324S, forexample, while being very high quality, have an inherent delay ofabout 1.7 milliseconds.

Imagine the situation where you are punching into a track on ananalog recording to correct a mistake. You will probably set up themonitoring so that you and the performer can hear both the outputfrom the recorder and the signal to be recorded. The performer willplay along with his part until the drop in, when the recorder willswitch over to monitor the input signal. This will be returned to theconsole and you will hear the level go up by approximately 3dBbecause you are now monitoring the same signal via two paths.

On the 3324S you can make a cross fade punch in of up to about

370 milliseconds. This is a good feature, but when you have madethe punch in - using the monitoring arrangement described above -you will hear the input signal added to the same signal returnedfrom the recorder but delayed by about 1.7ms. This will causedphase cancellation and an odd sound. Fortunately, Sony haveincluded an analog cross fade circuit which will imitate what ishappening in the digital domain, but without the delay.

Editing

DASH was designed to be a cut-and-splice editing format. Briefly,this is possible but it was found in practice that edits were oftenunreliable. Editing of DASH tapes is now done by copying betweentwo machines synchronized together with an offset. Twosynchronized 24-track machines are obviously more versatile in thisrespect than one 48-track.

Maintenance

Although an analog recorder can be, and should be, cleaned by therecording engineer in the normal course of studio activities, a DASHmachine should only be cleaned by an expert, or thousands of dollarsworth of damage can be caused. The heads can be cleaned with a specialchamois-leather cleaning tool, wiping in a horizontal motion only. Cottonbuds, as used for analog records will clog a DASH head with their fibers.Likewise, an analog record can be aligned by a knowledgable engineer,but alignment of a DASH machine is something that is done every six


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months or so by a suitably qualified engineer carrying a portable PC anda special test jig in his tool box. The PC runs special service softwarewhich can interrogate just about every aspect of the DASH machinechecking head hours, error rates, remote ports, sampler card etc etc. With

the aid of its human assistant it can even align the heads and tape tension.

Current significance

The current significance of DASH is as a machine that can record onto arelatively cheap archivable medium, with confidence that tapes will bereplayable after many years. Also, when an analog project is recorded ontwin 24-track recorders, it is often considered more convenient forediting to copy the tapes to a Sony 3348. The single 3348 is far faster andmore responsive than synchronized analog machines, making the mixing

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Very maintenance-intensive. For a 24-track system, four machines(4 x 8 = 32) are necessary to account for the one that will alwaysbe on the repair bench.

High resolution versions available (ADAT 20-bit, DTRS 24-bit, 96kHz, 192 kHz, with reduced track count)

The differences are these: Maximum record time: ADAT - 60 minutes, DTRS - 108 minutes ADAT popular in budget music recording studios DTRS popular in broadcast and film post-production

One further difference is that it is probably fair to say that the ADAT hasreached the end of its product life-cycle, although there are undoubtedlystill plenty of them around and in use. DTRS however is still useful as atape-based system offering a standard format and cheap storage.


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Check Questions

Was DAT originally intended as a professional or a domesticrecording medium?

What is the sampling rate of standard DAT? What is the resolution of standard DAT? What is 'azimuth recording'? Describe the head wheel in DAT recorder. What is SCMS? What is the distinguishing feature of a DAT machine capable of

near-simultaneous off-tape monitoring?

What is the sub-code area of the DAT tape used for? What is 'interleaving'? What is the width of the tape used for 24-track DASH? What is the width of the tape used for 48-track DASH? Describe how 24-track and 48-track DASH machines are

compatible.

How are DASH tapes edited? In DASH, why does a playback head come before the record head

in the tape path?

Comment on the cleaning requirements of DASH How many tracks does a modular digital multitrack (MDM) have? How can more tracks be obtained? Comment on the types of usage of ADAT and DTRS machines.


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Technical sound engineers abhor inconsistencies like this, so a new unitwas invented: dBu, where 0 dBu is 0.775 V, without any reference toimpedance. Once again, the u doesn't stand for anything. dBu issometimes written dBv (note lower case v). Confusingly there is also

another reference: dBV (note upper case V), where 0 dBV is 1 volt. Insummary:

0 dBm = 1 mW

0 dBu = 0.775 V

0 dBv = 0.775 V

0 dBV = 1 V

There are more:

dBr is a measurement in decibels with an arbitrary quoted reference level

dBFS is a measurement in decibels where the reference level is the fulllevel possible in a specific item of digital audio equipment. 0 dBFS is themaximum level and any measurement must necessarily be negative, forexample 20 dBFS.

All of the above (with the exception of dBFS) refer to electrical levels.We also need levels for magnetic tape and other media. Analog recordingon magnetic media is still commonplace in top level music recording,and outside of the developed countries of the world. Magnetic level ismeasured in nWb/m (nanowebers per meter). Nano is the prefixmeaning one thousandth of a millionth. The weber (Wb) is the unit ofmagnetic flux. Wb/m is the unit of magnetic flux density, or simply fluxdensity. Wilhelm Weber the person (pronounced with a v sound inEurope, with a w sound in North America), by the way, is to magnetism

what Alessandro Volta is to electricity.

There are a number of magnetic reference levels in common use. Ampexlevel, named for the company that developed the tape recorder fromGerman prototypes after World War II, is 185 nWb/m. NAB (NationalAssociation of Broadcasters, in the USA) level is 200 nWb/m. DIN(Deutsche Industrie Normen, in Europe) level is 320 nWb/m. Insummary:


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Ampex level: 185 nWb/m

NAB level: 200 nWb/m

DIN level: 320 nWb/m

Its worth noting that none of these reference levels is better than anyother, but NAB and DIN are the most used in North America and Europerespectively.

Operating Level

An extension of the concept of level is operating level. This is the levelaround which you would expect your material to peak. Much of the time

the actual level of your signal will be lower, sometimes higher. Its just afigure to keep in mind as the roughly correct level for your signal. Inelectrical terms, the standard operating level of professional equipment is0 dBu. There is also a semi-professional operating level of 10 dBV.This does cause some difficulty when fully professional and semi-professional equipment is combined within the same system. Either youhave to keep a close eye on level and resign yourself to makingcorrections often, according to what combination of equipment youhappen to be using, or buying a converter unit that will bring semi-pro

level up to pro level.

Magnetic tape also has a standard operating level - several of them infact. To simplify a little since analog magnetic tape is now a minoritymedium, albeit an important minority: In a studio where VU meters areused, then it is common to align the VU meters so that 0 VU equals +4dBu. Tape recorders would be aligned so that a tone at 200 nWb/m givesa reading of 0 VU. In short:

200 nWb/m on tape normally equates to +4 dBu and 0 VU

Most brands of tape can give good clean sound up to 8 dB above 200nWb/m and even beyond, although distortion increases considerablybeyond that.

Digital equipment also has an operating level, of sorts. In some studios- mainly broadcast - digital recorders such as DAT are aligned so that18 dBFS (18 dB below maximum level) is equivalent to +4 dBu and 0


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RMS and Peak Levels

How do you measure the level of an AC (alternating current) waveform?Or to put it another way, how do you measure the level of an AC

waveform meaningfully? A simple peak-to-peak measurement, or peakmeasurement, shows the height (or amplitude) of the waveform, but itdoesn't necessarily tell you how much subjective loudness potential thewaveform contains. A very peaky waveform (or a waveform with ahigh crest factor, as we say) might have strong peaks, but it will not tendto sound very loud. A waveform with lower peaks, but greater areabetween the line and the x-axis of the graph will tend to sound louder ondelivery to the listener. The most meaningful measurement of level is theroot-mean-square technique. Cutting out all the math, the RMSmeasurement tells you the equivalent heating capability of a signal. Awaveform of level 100 Vrms would bring an electric fire element to thesame temperature as a direct (DC) voltage of 100 V. A waveform of level100 Vpeak-to-peak would be significantly less warm.

Frequency Response

It is generally accepted that the range of human hearing, taking intoaccount a selection of real live humans of various ages, is 20 Hz to 20kHz, and sound equipment must be able to accommodate this. It is not

however sufficient to quote a frequency range. It is necessary to quote afrequency response, which is rather different. In addition, we are notlooking for any old frequency response, we are looking for a flatfrequency response which means that the equipment in questionresponds to all frequencies, within its limits, equally and any deviationsfrom an equal response are defined. The correct way to describe thefrequency response of a piece of equipment is this:

20 Hz to 20 kHz +0 dB/-2 dB

or this:

20 Hz to 20 kHz 1 dB

Of course the actual numbers are just examples, but the concept ofdefining the allowable bounds of deviation from ruler-flatness is the key.


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Another way of measuring the noise performance of equipment is EIN orEquivalent Input Noise, and this is mainly of relevance to microphonepreamplifiers. An example spec might be 'EIN at 70 dB gain: -125 dBu(200 ohm source)'. This means that the gain control was set to 70 dB and

the noise measured at the output of the mic preamp - in this case themeasurement would be 55 dBu. When the set amount of gain issubtracted from this we get the amount of noise that would have to bepresent at the input of a noiseless mic amp to give the same result. The'200 ohm source' bit is necessary to make the measurement meaningful.If the EIN figure does not give the source impedance, then I am afraid themeasurement is useless. Perhaps it is giving the game away to say thatthe reason a gain of 70 dB is quoted is because mic preamps normallygive their optimum EIN figures at a fairly high gain. The lower the gainat which a manufacturer dare quote the EIN, the better the mic inputcircuit.

Modulation Noise

Noise as discussed above is a steady-state phenomenon. It is annoying,but the ear has a way of tuning out sounds that dont change. However,there is another type of noise that constantly changes in level, and that ismodulation noise. One source of modulation noise is that which occurs inanalog tape recorders. The effect is that as the signal level changes, the

noise level changes. This can be irritating when the signal is such that itdoesn't adequately mask the noise. A low frequency signal with fewhigher harmonics is probably the worst case and will demonstratemodulation noise quite clearly. Noise reduction systems, as mainly usedin analog recording, also have the effect of creating modulation noise.Noise reduction systems work by bringing up the level of low-levelsignals before they are recorded, and reducing the level again onplayback at the same time reducing the level of tape noise.Unfortunately, the noise level is now in a state of constant change and

thereby drawing attention to itself. Some noise reduction systems havemeans of minimizing this effect. All of the various Dolby systems, forexample, work well when properly aligned.

Quantization noise in digital systems is also a form of modulation noise.At very low signal levels it is sometimes possible to hear the noise levelgoing up and down with the signal.


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Sine wave - the simplest possible sound with no harmonics

The effect of even-order harmonic distortion on a sine wave


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control, 2) gain due to EQ, or perhaps 3) unexpected enthusiasm on thepart of a musician. Also, when signals are mixed together, the resultinglevel isn't always predictable. Professional equipment can handle levelsup to +20 dBu or +26 dBu, therefore there is always plenty of headroom

to play with. Of course, the more headroom you allow, the worse thesignal to noise ratio, so it is always something of a compromise.

In recording systems, it is common to reduce headroom to little or zero.The recording system is at the end of the signal chain and there are fewervariables. Nevertheless, it does depend on the nature of the signal source.If it is a stereo mix from a multitrack recording, then the levels areknown and easily controllable therefore hardly any headroom is required.If it is a recording of live musicians in a concert setting, then much moreheadroom must be allowed because of the more unpredictable level of thesignal, and also because there isn't likely to be a second chance ifclipping occurs.

Wow and Flutter

The era of wow and flutter is probably coming to an end, but it hasn'tquite got there yet so we need some explanation. Wow and flutter areboth caused by irregularities in mechanical components of analogequipment such as tape recorders and record players. Wow causes a long-

term cyclic variation in pitch that is audible as such. Flutter is a fastercyclic variation in pitch that is too fast to be perceived as a rise and fall inpitch. Wow is just plain unpleasant. You will hear it most often, and at itsworst, on old-style juke boxes that still use vinyl records. Flutter causes adirtying of the sound, which used to be thought of as whollyunwelcome. Now, when we can have flutter-free digital equipment anytime we want it, old-style analog tape recorders that inevitably sufferfrom flutter to some extent have a characteristic sound quality that isoften thought to be desirable. Wow and flutter are measured in

percentage, where less than 0.1% is considered good.


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pro engineer school vol.1

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