His studio’s eclectic, diverse client list has grown substantially over the years, as has his Audient gear collection. “My Audient mixer ASP4816 has facilitated my work due to its routing flexibility and sound quality,” says Manolito, acknowledging the part his console plays in the studio’s success today.

With two separate live areas with different acoustics, he describes the “versatile” setup at Lito’s Place: “It combines both analogue and digital equipment with the best DAW software currently on the market: Logic Pro X and Protools. All is centralised into my Audient console, ASP4816.” He also counts surround sound monitor controller ASP510, an iD22 audio interface, the 8-channel mic pre ASP880, as well as the recently released Sono guitar recording interface amongst his gear list.

“All projects over the past 30 years have been interesting and a learning experience, and I pride myself on making each project have its unique sound,” he says. He certainly has had a mix of visitors to the studio. Recent clients have been ‘contemporary folk’ artists which include five-piece Maltese folk band, Skald. “That will be mixed by the end of May,” he says.

Another interesting project is an album of Cuban Music, with the Maltese flautist from Malta National Orchestra, Fiorella Camilleri and Cuban guitarist, Ahmed Dickinson Cardenas. He’s also just finished recording a jazz suite project; a narrative of the second world war in Malta in remembrance of the events 75 years ago, composed by Dominic Galea. This will be on the market any day now.

Manolito is equally comfortable producing Maltese musicals in his native tongue, as he is collaborating with foreign artists and media companies in Europe and the US, such as the BBC, Warner Bros & Columbia Pictures. Indeed, Brad Pitt and Emily Blunt have both been to Lito’s Place, making use of the “perfect ADR studio space” for tv/film audio post-production.

It’s important to him to stay current, by regular servicing and upgrading his equipment, as well as ensuring he is professionally up-to-date. “I make a point of increasing my knowledge and being proactive in the industry, which is why I have repeat clients who have complete faith in my work.”

He has come a long a way. “I started off with a Tascam Porta Studio 244, a set of speakers and my mother’s Shure microphone 565SD in my parents’ music room,” he says. He and his brother, a composer and jazz pianist, opened Wave Recording Studio in 1990 and then in 2008 he went solo and opened Lito’s Place.

As he embarks on his fourth decade in the industry, Manolito has great plans. “We will see Lito’s Place relocate to new premises with better facilities and a new concept. Moving away from an urban area but still enjoying the perks of the urban lifestyle. Those better facilities will include the Audient Console – ASP8024 Heritage Edition.

Now that is an excellent idea! To stay up to date with Manolito and all projects, you can find him on both Facebook or Instagram.

In a Class A amplifier the output transistor (or valve, or carbon element) is turned on all the time, even when no signal is present. This allows an audio wiggle at the input to enjoy continuous attention as the internal electronics follow along without interruption. Pretty much all other amplifier classes chop signals into bits that get processed separately and then re-assembled, which invites glitches. When dealing with very small signals like those from a microphone, it stands to reason that chopping one to bits and then making it 1000 times bigger might risk amplifying the glitches as well. That’s not a problem for Class A, since the signal is processed whole. This is considered its principal advantage. In fact, I don’t believe I’ve come across any small-signal audio input that was not configured in Class A. However, the acts of driving line outputs and speaker loads are now dominated by more recent audio classes.

The conceit with Class A is that since the signal is not interrupted, the output should be a linear recreation of the input, and it generally is, more or less. In fact, Class A’s harmonic distortion performance can be hard to beat.

The concept of ‘linearity’ is worth explaining. If say the signal at your amplifier’s input doubles in amplitude, then the output does likewise. Given a further 20% increase, and the output follows exactly. This behaviour is said to be ‘linear’ because the input and output signals wind up the same shape, with one simply bigger than the other. Now imagine if the input amplitude doubles, but the output only increases by 97%. Then it doubles again but this time the output only increases by 70%. This uneven transfer of signal trace from input to output is said to be ‘non-linear’. It is important to understand that in the above examples we are not just talking about when the music gets louder or softer; we are talking about the rise and fall of every cycle within the audio wave, and the resultant mismatch is called ‘harmonic distortion’.

Transistors, valves and carbon elements exhibit good linearity only within a portion of their overall operational range, and beyond that, amplification can be a little wobbly or compressed; sometimes on purpose. In the Fairchild 670 compressor, gain reduction is produced by moving the valves into electronic non-linearity. Circuit architecture itself can offer challenges. In its purest form, Class A is inherently lopsided, with output peaks pulled in one direction by a powerful transistor or valve, and in the other direction by a passive resistor at the mercy of downstream loads [diagram-2] When things get extreme, the result can be a lopsided signal like in this oscilloscope shot of a ‘corrupted-for-the-purpose’ Class A transistor amp. [diagram-3] The signals in and out are overlaid on each other to show their difference, and the mismatch corresponds to about 10% harmonic distortion.

In some old-time Class A valve amplifiers, asymmetrical harmonic distortion is more common than you might think, and since curvy distortion contains lots of even harmonics, it actually doesn’t sound that bad. In fact, it may be the thing we love best about some of that equipment. Other classes tend to distort more edgy and square, which can sound less pleasant, unless you’re a fan of German metal.

Class A has the advantage of only needing a single power supply, but the main drawback is massive inefficiency. Because the active devices are always conducting, and never turn off, most energy winds up dissipated as heat and only a small part drives the signal — typically around 30% or so. In an effort to achieve more efficiency, less heat, and better symmetry, old-time designers went back to their drawing-boards. One idea was to connect two Class A amplifiers in an opposing ‘push-pull’ arrangement, which we today refer to as ‘bridge mode’. Another was to use an active device called a ‘current-source’ in place of the load resistor. Both these techniques deliver superior harmonic fidelity, with the push-pull idea providing near perfect symmetry.

CLASS B

The logical next step was to design push-pull symmetry into a single amplifier. Initially appearing around 1930 (using valves), Class B dispenses with the load resistor, and employs two opposing transistors at its output — one pulls down and the other pulls up (maybe they should be called ‘pull-pull’ instead). [diagram-4]

Each transistor only conducts for half the signal cycle, and both devices are turned off when there is none, so the result is greater efficiency (around 50%) and less heat. The major drawback with Class B is that every time the signal crosses zero, both transistors are completely off, so part of the signal goes missing. This is called ‘crossover distortion’, and it has nothing to do with speaker crossovers. It takes the form of persistent square clipping right in the middle of the signal (rather than at the peaks) and it makes Class B a very lo-fi animal indeed.

CLASS AB

The solution to Class B crossover distortion is to doctor the circuit slightly so that instead of both transistors being turned off during zero-crossings, they are both turned on. That means one device handles the upper half of the wave, both devices work together across the middle, and the other device handles the lower half of the wave. Kind of like a baton being passed in a relay. Thus none of the signal is dropped, and extremely low harmonic distortion figures are possible. Since both transistors are now engaged for slightly longer than half of each cycle, they are said to be biased towards Class A, so this topology gets referred to as Class AB. [diagram-5]

The low output impedance and excellent symmetry of Class AB makes it perfect for driving speakers, transformers and line outputs, so eventually it turned up in everything, and is arguably the most ubiquitous amplifier architecture of the last 50 years. Go back to say 1985, and almost every hi-fi amp, professional sound reinforcement amp, mixing desk, radio, telephone, and audio op-amp was configured in Class AB. Even the Urei 1176 Compressor, revered for its Class A-ness, uses a Class AB stage to drive the output transformer. On the downside, both Class B and AB normally require a second power supply rail, which adds some extra cost and clutter.

CLASS C

Class C amplifiers operate at radio frequencies, and are not used for audio at all. The architecture looks a bit like Class A, but with the output resistor swapped for a tuned circuit usually made up of a capacitor and an inductor. [diagram-6] Class C will amplify a specific frequency while attenuating all others, so it is commonly found in radio transmitters and receivers, and also forms the basis of some oscillator circuits used for logic clocks and digital timing.

CLASS D

The 50% practical efficiency of Class AB seemed like plenty, until the advent of cell phones and mobile music playback systems in the ’80s and ’90s. Batteries cost money, and rechargeables drain quickly (even now) so designers inevitably shifted their focus to power conservation, with the goal of 100% efficiency as their target — meaning every electron consumed by an amplifier would be used to drive headphones or speakers, with zero left over to dissipate as heat. Looked at another way, if efficiency increases from 50% to 100% then your battery will last twice as long, and everyone with a smart phone wants that. Class D (first mooted in 1958) today boasts efficiencies of well over 90% and has revolutionised audio amplifier design. Sorry, the ‘D’ does not stand for ‘Digital’ — Class D does its business in analogue, but goes about it in a non-linear way, a bit like a switch-mode power supply.

Class D uses output switching to rapidly turn transistors (usually MOSFETs) completely on and off. This switching happens at a rate much higher than audio frequencies, so that each switching cycle (called a ‘duty cycle’) handles a very small section of the signal waveform. At a rate of say 200kHz, a 20kHz sine wave would be split into 10 equal sections, and a 20Hz sine wave into 10,000. When switched on, the transistors cause the output voltage to rise by supplying current (via a low-pass filter) to a loudspeaker or other load. A comparison circuit monitors this rise, and when the output gets to the right level (microseconds later) the transistor turns off and waits for the next switching cycle where the process repeats. In my diagram, the input signal is compared to a continuous sawtooth wave whose frequency sets the duty cycle rate. [diagram-7] Because the ‘pulse’ duration within each duty cycle varies according to demand, this kind of energy delivery is known as ‘pulse-width modulation’.

In an alternative variant of Class D the pulse duration is fixed, but the length of the duty cycle varies, and this scheme is called ‘pulse-density modulation’. What both schemes have in common is that they spit tiny pellets of energy at a load, to build mountains of amplitude – a bit like a 3D printer making an object out of droplets of plastic. Because the transistors are only ever turned on for the amount of time required to produce each droplet, there is very little wasted energy, and very little heat, meaning amplifiers can be made smaller because they don’t need a great big ventilation system.

In the late nineties and early noughties, Class D schemes really came of age, making self-powered monitors practical, achievable, and affordable. In pro sound reinforcement these boxes were welcomed with open arms, so much so that it is now unusual to rig a live show without them. Also, Class D amplifiers are used in many current brands of near-field studio monitors, with good ones boasting less than 0.01% Total Harmonic Distortion. In smart phones and other headphone-driving, battery-powered gadgets, Class D is a popular inclusion for its low power consumption, small size, and cool running.

CLASSES E & F

Now we’re back to radio frequencies. Class E is like a mash-up between the tuned linear circuit of Class C and the full on/off switching of Class D. They are designed to be used above 100MHz, where Class C starts to struggle. Class F is like E with fries.

CLASS G

This variant of Class AB was borrowed from some hi-fi amps like the NAD 3220 from 1989, and used to great effect in mobile phones. The basic principle is to use two separate sets of power supply rails — some lower voltage ones for the vast bulk of audio demands, and higher ones rigged to suddenly switch on only for louder peaks and head-banging. Since the higher voltage rails are completely off most of the time, the dividend is lower power consumption, less heat, and longer battery life. [diagram-8]

CLASS H

Used in some cell phones, this scheme takes the idea of Class G to the logical limit. Instead of rail switching, Class H tracks the input signal and moves a single set of power rails up and down in unison. This leads to around 80% efficiency, which is better than Classes G and AB, but not quite as good as D.

CLASS I

A push-pull variation of Class D, patented by Crown Amplifiers, using separate pulse-width modulation on two opposing switches. When the state of both switches match, they cancel each other and there is no output — but when the pulse-width on one is wider than the other, the output will swing towards it. Crown say this is more efficient than regular push-pull versions of Class D, and solves the problem of output transistors blowing up if they accidentally turn on at the same time.

CLASS S

Another version of Class D that is usually associated with radio frequency or digital input audio operation. It applies a fancy algorithm to convert multi-bit digital signals directly into pulse-width modulation, bypassing the normal D-to-A stage. Annoyingly out of sequence, the ‘S’ is generally believed to stand for ‘Switching’.

CLASS T

Yet another glorified version of Class D, in a proprietary design by a company called Tripath — hence Class ‘T’ — from 1996. Apple used them in the G4 and iMac. Their central claim was that a superior process was employed to control the pulse-width modulation, with feedback taken from the switching stage rather than the output, resulting in higher gain, and lower distortion.

ALL CLASS

This writer is not aware of any other meaningful amplifier classes, so most of the alphabet is still up for grabs, if you have any brain-waves. Not sure about the Tripath idea of simply co-opting any letter you like. Will this lead to Meyer, M-Audio, Motorola and Marshall all in litigation over Class M? Hmmm…  

IN LOUD AGREEMENT

Common sense collided with history in 1940, when some guys from NBC, CBS, and Bell Telephone got together to agree on a new standard volume indicator and reference level, and what they invented was the VU meter as we know it today. Intended to display comparative loudness or power in ‘volume units’, their aim was to create a meter that would indicate identical levels for a range of dissimilar sources which an average listener would consider to be of equal loudness. They assembled focus groups and played them male speech, female speech, piano, brass band, dance orchestra, and violin recordings, asking them to pick the level for each one where the sound system appeared to be on the threshold of distortion. Since all of these sources contain different ratios of peak to overall RMS power, a peak-reading scheme was deemed undesirable.

RMS, or Root Mean Squared, is what you get when you take an undulating AC waveform and squish it into a flat block of equivalent density. Imagine the waves at the beach are peaking all over the place while you’re trying to work out the average depth of the tide. If you had a football-field sized piece of glass, you could squish the waves down until the sea is flat and then easily measure the volume of water. RMS tells you the power density under each wave, and the VU guys saw volume in a similar way. Their new meter would measure average power density against time. They chose 0.3 seconds as their preferred sample depth, rigging an internal capacitor to take that long to charge or discharge — a process governing each full-scale deflection and recovery. The slow timing was chosen partly because it would make a moving needle comfortable to read.

The VU meter was calibrated so that 0vu — aligning with ~71% deflection — would correspond to a level produced by 1mW of power in a 600Ω system. It was intended that its sensitivity might, via the addition of an external resistor, be easily adjusted to any standard reference level, say +4 or +8vu, or to indicate 100% modulation of a broadcast signal at whatever level that happens. Both uses were accommodated on the VU scale, marked in dB from –20 to +3vu, with the 100% mark aligned with 0vu.

STANDARD ISSUE

The 0vu standard was overwhelmingly accepted in all corners of industry, and later referred to as 0dBm (dB milliwatts). A 1mW sine wave into a 600Ω load produces a level of 0.775vrms, and today we disregard the load and just call the voltage level 0dBu. The ‘u’ originally meant ‘unloaded’ (ie. we don’t care what the load is, only the voltage level) but that’s all so long ago, now we simply regard dBu as dB ‘units’.

Some other people decided they didn’t like a zero reference of 0.7745966692… volts RMS, and upped it to 1 Volt even, calling it 0dBV. Today our domestic hi-fi gear uses –10dBV as its 0vu reference point (around –8dBu), whereas most professional systems use +4dBu, and some gadgets give you the choice of both.

SCALING THE EUROPEAN PEAKS

By 1950, the dBm was a smash hit world-wide, but the VU meter was not so successful outside the USA. Those pesky Europeans were less interested in subjective comparisons of sound sources, and more interested in knowing exactly how signals were being handled by actual equipment. In Germany, Scandinavia, and England, engineers had been separately developing peak level meters with the principal aim of monitoring broadcast modulation. Nobody saw the benefit in using an RMS-type device to do this monitoring, since it is the program peaks which cause over-modulation. Engineers at the BBC were even unkind enough to refer to the VU meter as the ‘Virtually Useless’ meter.

Peak-reading meters were around before VU, but to this day there has never been a single standard scale, timing or reference level to which all Peak Program Meters (PPM) abide. However, they do all calibrate to so many dBu, and to the naked eye all appear to sort of behave the same way-ish, with fast attack times (to catch the peaks) and usually a slow decay. In the days of analogue valves, overloading produced distortion that was not terribly harsh, and the feeling was if it lasted less than a few milliseconds nobody would notice. A milliamp meter could deflect to full scale in 1mS, so it became standard practice to slow down the attack slightly to remove these aurally meaningless peaks — the Brits chose 4mS and the Germans 10mS. With today’s digital equipment, hitting full-scale makes a loud bang, so quicker attack times can be useful.

PPM decay might last anything from 1.5 (EBU) to 2.5 (BBC) seconds, mainly to make things easy on the eye. In time, it became fashionable to include a peak-hold of between one and three seconds, which is helpful if you have to look away momentarily, and don’t want to miss what happened.

The front panel scales and alignments, however, are largely dreamt up. The BBC used a scale which was simply numbered one to seven, where four corresponded to 0dBu, and six to their maximum permitted signal of +8dBu. The EBU scale was marked in dB from –12 to +12dBu, with 0dBu marked as ‘Test’. There were many others — Diagram 2 provides a taste.

RE-BUILT FROM THE TOP-DOWN

Today with digital audio, it is very important to avoid 100% bit saturation of the digital converter. The kind of scale we’re most used to refers to that point as 0dBFS, or 0dB Full Scale, and proceeds downward in negative decibels. The point at which any analogue ins or outs on a digital device agree with 0dBu could be –24, –20, –18, –12dBFS, or any other figure that takes the fancy of the manufacturer who’s gear you are using. Again, there is no single universal standard. On stand-alone devices a dBFS meter is likely to be constructed using LEDs, as opposed to a moving needle, but on more complex equipment will probably occupy part of a graphical display.

Some meters now display both VU and PPM simultaneously, with VU expressed as a solid bar, and PPM as a moving dot above it. Such schemes were proposed as far back as 1981 by Michael Dorrough of Dorrough Electronics, who has since gone on to invent a new type of ballistic which he calls the ‘Loudness Meter’, or LUFS (Loudness Units Full Scale).

Why not create a new meter? If anything is clear from all of the above it is that anybody who ever created a meter scheme was really just interested in getting some visual assistance to see what was happening to the signal, or the equipment. There’s nothing intrinsically empirical about any of the PPM schemes; people just decided to do things a certain way. If it helps an audio engineer perform their job well, there is nothing to stop you, me, or Mrs Tech Bench meddling with timing and alignment, or applying new wacky algorithms to create any kind of meter we want.

Who knows — if we come up with a good one, it might catch on.

FREQUENCY THROUGH TIME

In 1892 Alexander Graham Bell’s telephone system had connected New York to Chicago with a pair of wires over 1400km long. The two parallel wires effectively functioned as capacitor plates, and pretty soon it got up everyone’s nose how the whole thing behaved as a geographically-sized, high-cut filter. To make up for the resultant loss of sibilance and intelligibility, Bell Laboratories developed and installed fixed-frequency, non-adjustable filters to restore the top end, and thus ‘equalise’ all frequencies in the telephone signal.

The first user-adjustable equaliser is credited to John Volkmann from RCA, who created one for movie playback systems in the 1930s. Most picture theatres of the day were built in the silent era, or were converted song and dance venues. The ‘talkies’ brought electronic audio into those venues for the first time. Between the primitive speaker systems, the sound recordings themselves, and the lucky dip of theatre acoustics, it often added up to an unsatisfactory day out for the ticket-paying punter. Volkmann’s gadget offered boost and cut of multiple selectable frequencies, and to theatre managers the altering effect must have seemed almost magical.

At the same time, other companies were developing equalisers to be used in audio production and broadcast. Art Davis of the Cinema Engineering Company hit an early home run by designing the first proper graphic equaliser, the Type 7080. Featuring six bands with 1.5 octave spacing, the unit offered 8dB of boost and cut, zero insertion loss, and faders fashioned from linear 17-position switches sporting ‘typewriter key’ knobs. The unit was an early active device, which employed variable passive filters to alter the cathode resistance of a tube amplifier, and therefore its gain. At the back end there was some make-up gain followed by a push-pull section, driving a transformer that delivered a floating differential output. It was clever stuff for its day. Later, Art Davis defected to Altec, where he designed the seven band Model 9062A graphic equaliser, which remained a benchmark device till the 1970s.

In 1952, Englishman Peter J. Baxandall published his scheme for a negative-feedback tone control circuit using potentiometers as opposed to switches, thus allowing full user control. It also ushered in a new era of fully-active EQ circuitry and became the most reproduced tone control in history. To this day there has scarcely been a hi-fi unit anywhere in the world which fails to incorporate it behind those knobs marked Bass and Treble. Baxandall received no royalties for his invention, but in 1950 the British Sound Recording Association gave him a $25 gold watch, in honour of an earlier version. That watch would be worth something now.

Meanwhile in Texas, Professor Wayne Rudmose had figured out that public address systems needed more than just a bass and treble control to cure feedback. His 1958 paper Equalization of Sound Systems proposed that PA systems could actually be tested for frequency response (using non-existent equipment that would need to be devised), and thereafter filters might be created and installed to equalise that response. His theories were successfully put into practice at Dallas Love Field Airport, and from then on ‘tuning’ a PA became a thing to do.

The next decade saw a lot of work done to further tame PA systems. Rock ’n’ roll was in full bloom, and program material was getting louder. In 1967, Altec Lansing introduced the first 1/3-octave passive notch filter set specifically for PA tuning called the Acousta-Voice system, and from there it was only a hop and a jump to create a 1/3-octave, cut and boost-style graphic equaliser.

The last truly major development in EQ was the parametric equaliser.  Historically, graphic equalisers were useful studio tools, but had big faders, and took up lots of space. What engineers craved was a smaller and more flexible device which could be built right into a channel strip. In 1971, Daniel Flickinger invented a sweepable EQ which allowed adjustment of frequency and gain, in three overlapping bands of fixed width. Nowadays, manufacturing a mixing desk without at least one band of sweepable EQ is unthinkable. In 1972, George Massenberg presented a paper on a similar invention that he and two associates had been developing which, besides frequency and gain, allowed user adjustment of the third parameter — bandwidth. Massenberg called his device a ‘Parametric Equalizer’ and he continues to sell them to this day, under the GML brand name.

INDUCTED INTO OBSCURITY

Looking at the work of Massenberg and his contemporaries, one aspect that never gets talked about is how they stood astride a historical line of demarcation for the electrical component called an Inductor. Before Massenberg, if you took the lid off an equaliser you would probably see inductors. After Massenberg, probably not.

Inductors and capacitors are components with electrical properties that vary with frequency, and they are what shapes the response of any frequency or time dependent circuit. They work in equal and opposite ways, so it was always natural to see them together in the same devices, the EMI REDD desk EQ from the ’60s being one fine example.

However, inductors are essentially a coil of wire with properties similar to a radio antenna. In Massenberg’s paper, he described them as marginal performance components which inherently invite parasitic electromagnetic noise into a system, and he proposed the use of several pre-existing electronic Indian rope tricks to do away with them completely. One was an op-amp circuit called a ‘gyrator’, which takes a capacitor and, more or less, draws a moustache and glasses on it to fool the neighbouring electronics into thinking it’s an inductor. The ideas he talked about increasingly appeared in the work of others, and gyrators soon replaced inductors in graphic equalisers. Since then, inductors have become a sad and sorry underclass in audio, and now you only really encounter them inside passive speaker crossovers. If you come across one, loan it a dime.

ALL THINGS BEING EQUAL

(Fig-1) Rudmose used fixed frequency notch filters to equalise the PA in Dallas, and that was that, no knobs. These days fixed EQ is less common, but the RIAA curve has been a solid stayer. On a vinyl record, the grooves wiggle from side to side, and the bass notes (which contain the most energy) wiggle the widest. So in order to fit a useful amount of material on the vinyl the bass is attenuated by 40dB at 20Hz, rising to unity at 20kHz. The filter slope zig-zags between those two points, but what you hear is closer to a straight line. All records are made that way, so a phono input employs a fixed boost filter of the opposite slope, to put everything back to normal.

(Fig-2) Most mixing desk channels feature a sweepable or switchable high-pass filter to attenuate sub-bass and proximity issues in microphones. Occasionally you will see a sweepable low-pass filter, to help remove unwanted treble from bass instrument channels. These filters provide a flat response in their pass band, and generally roll off down a 6dB or 12dB per octave slope beyond their cutoff frequency. Filters of a higher order than 12dB per octave are less common, and tend to imply a special purpose. Linkwitz and Riley famously use 24dB per octave filter slopes in their electronic speaker crossovers.

(Fig-3) If you want to affect a specific band of frequencies, you need a bandpass filter. The frequency response is usually bell-shaped, with an upper and lower roll-off slope, and there are two main varieties — Constant Q and Variable Q.

‘Q’ stands for Quality Factor, and is simply the ratio of -3dB bandwidth in Hertz to a centre peak frequency. For example, if we have a centre frequency of 1000Hz with -3dB roll-offs at 414Hz and 2414Hz, then the bandwidth (upper minus lower) equals 2000 Hz, therefore Q=0.5 (1000/2000). For amplitudes less than 3dB there is no -3dB point, and subsequently all kinds of disagreement about how to calculate bandwidth. I don’t want to start a bar room fight, so I will stick with the above as a general principle, and just say that whenever we increase Q, we decrease bandwidth, resulting in sharper curves.

Bandwidth can also be measured in octaves, which are very roughly the inverse of Q — but always slightly more, because they are calculated from either extremity rather than the centre. Since an octave is merely a doubling of frequency, the distance between 414Hz and 2414Hz winds up being about 2.5 octaves.

CONSTANT Q VS VARIABLE Q

When boost or cut is applied to a bandpass filter, bandwidth and Q will either be constant or variable, and the term ‘variable’ does not mean adjustable.

The frequency response curves for a variable Q equaliser (Fig-4) show that the bandwidth or Q of the equaliser changes with the amount of boost or cut used. Sometimes this is called ‘Proportional Q’. At low settings the equaliser exhibits a wider relative bandwidth than at high settings, and only achieves its specified bandwidth at maximum boost or cut, so minor adjustments made at a specified frequency will cause changes to other frequencies some distance away. This type of response is desirable in situations where you want to change the overall tonal balance of a sound, without over-emphasising any particular frequency. Variable Q circuitry is common in old-fashioned tone controls, guitar amps, and most professional equalisers that fail to declare themselves as being ‘Constant Q’.

The frequency response curves for a constant Q equaliser (Fig-5) show that, even for small amounts of boost or cut, the equaliser maintains its stated bandwidth. That makes it very useful for applying changes to specific frequencies, in circumstances where you would prefer to leave nearby tones undisturbed. Examples might include removal of feedback, or enhancement of a particular resonance in a musical instrument.

GRAPHIC EQ

In live sound reinforcement, the 31-band graphic equaliser has left its footprint more than any other EQ. With evenly spaced 1/3-octave bands, it gets its name from the way the response curve is so directly displayed (like a graph) by the faders that control it. When you are battling wild frequencies in a pesky room, it’s helpful to quickly see where you’ve been.

Live sound techs talk about their equipment the way a tradesperson talks about their ute. ‘I only nudged the fader, but it produced enough grunt to uproot this tree.’ In that context a good graphic equaliser will be one for which each fader exhibits a solid grasp of its band of frequencies, so that when you move one ‘something happens’ (I have used equipment where nothing happens). Graphic equalisers mainly employ constant Q equalisation, so that bands several faders apart from one other don’t interact, but a superior design will allow bands immediately adjacent to ‘combine’ slightly when the faders are moved together, eliminating any dip that might occur between them. Combining was invented by Art Davis when he created his Acousta-Voice system. He reasoned that if your P.A. is feeding back at a frequency half-way between two adjacent bands, then you will attenuate both, with the expectation that all frequencies between the two ought to be grabbed in the process.

PARAMETRIC EQ

In recording studios, room sound and feedback are not quite as high on the worry list, and EQ tends to be more purely applied to affect the timbre or balance of an instrument. With independent control over bandwidth (Q), frequency and amplitude, a parametric equaliser is able to produce almost any shape of EQ curve one might desire — something the fixed frequency bands of graphic EQ cannot do. Circuits tend to employ a state-variable filter topology using op-amp integrated circuits, and I have never heard of a parametric EQ using valves.

BAXANDALL TONE CONTROL

This classic tone control provides a shelf, as opposed to a peak-type response. A stock standard configuration might produce 15dB cut and boost at 100Hz and 10kHz, with the slopes returning to unity not far either side of 1kHz, but there are countless variations. (Fig-6)

VIRTUAL

In this day and age of software-based recording, there are any number of proprietary algorithms that exist to mimic all of the above, and more. I’m with the peeps who favour minimal EQ and careful mic placement during tracking, though for mixing I will admit to being quite a fan of software equalisers. There are many who love to mix or print through analogue EQ equipment. In the end, if one way sounds better than another, then it is.

EQ PHONE HOME

All that’s useful, of course, but Mrs Tech Bench asked me if there’s a way to manage your woofs and tweets without having to think about it. Well… I tell her about a junior conference technician I knew named Mark. Mark is a genius. On his first real test, he is left alone with a PA and a medical professor on stage, who blathers into his lapel mic about the exciting world of cortico-steroids. Mark thinks he can hear the onset of feedback, but what to do about it? He pulls out his phone and calls his colleague Dave, who is famous for being able to identify any feedback frequency instantly. Mark sings in a whisper down the phone line, ‘Wooooo’. ‘Three Hundred Hertz,’ says Dave. Mark reaches over to the graphic equaliser, pulls down the 300Hz fader, and the problem is gone. ‘Thanks Dave.’

…Now that’s the way to EQ.

SIX SILVER SWANS SIBILANTLY SQUAWKED

I suspect most readers of this article already own a digital recording program with a de-esser plug-in included, and may have had a bit of a fiddle. Those readers will have seen that the plug-in enables them to apply instantaneous gain reduction to sibilance and fricatives, which are the whistly and spitty bits of human speech. They often appear in our recordings with un-natural prominence above the average power of supporting dialogue, and can be as annoying as a finger that emerges from the loudspeaker to PoKe you in the TemPle wiTH eFFry SSSSSyllable.

‘Fricative’ is a jazzy word, which would be better suited to describe a style of dancing or cooking. Instead, it refers to sounds which emerge from our mouths that use friction as a substantive part of their mechanism, rather than just tonal resonance. The letter ‘T’ is a good example, as it involves blocking the airflow with your tongue, and then suddenly releasing it. Fricatives that don’t entirely block airflow, but which involve the friction of air through a narrow gap or across the teeth, tend to have a hissy nature, and are referred to as ‘sibilant’. The letter ‘S’ is the poster boy for sibilance, and it is from him we get the term ‘essy’ and the generic name for any device designed to reduce the amplitude of such fricatives when we don’t like them.

These noises all contain the bulk of their spectral energy in the 4-10kHz band of frequencies, while the main body of non-fricative speech resides four or five octaves lower, in the 100-500Hz band. Fricatives are a natural part of speech, but they can present challenges — and management or intervention is sometimes required.

TESST ONE TSSOOO

Some fricatives contain plosive sounds which from the point of view of a microphone can look like a shock-wave. Earlier I described the letter ‘T’ as being produced by a sudden un-blocking of the airway by your tongue. This produces not only the high ‘t’ sound, but also a sudden pressure wave of somewhat greater energy, and it can hit the mic element with a tiny thump. It’s as forceful as a bullant knocking on your front door, but when added to the intelligible part of the sound, the resultant recorded ‘T’ may exhibit more emphasis than initially heard by the naked ear.

Once that sound enters the recording equipment, it faces additional challenges. It is not uncommon for a vocal track to receive an equalisation boost in the top end, to improve intelligibility and enhance natural harmonics which may be pleasing to the ear. However, since fricatives reside in the same range, they will also be boosted, increasing their elevation above the average level of non-fricative content.

Furthermore, it’s quite common to compress a vocal track, to make it sound more powerful and consistent. Compressor attack times favoured for vocal performances are generally too slow to grab leading edges, with gain reduction applied only to the main body of the sound, making the fricative louder still.

Turning down the treble is an obvious way to reduce fricatives in a recording, but the audio track is also dulled as a consequence, which may be undesirable. We really want to be able to remove them while leaving the surrounding audio intact.

DE-OLDEN DAYS

As we saw in the article about compressors (Issue 123), Hollywood was taking control of audio dynamics from the very first days of the ‘talkies’ and de-essing appeared about 10 years later. In 1939, Warner Brothers Films developed a system for reducing excessive sibilant energy in speech as full modulation was approached on the optically-recorded film stock. This was probably the first time the idea had been put to industrial use, and control of sibilance in films has been ongoing ever since.

The same idea eventually crossed over into the record industry, in the area of disc cutting and mastering. If a high frequency signal has too much amplitude, it can be difficult for a playback needle to physically track the groove at speed, so it made sense to place an upper limit on such undulations. One example from the early 1960s was a record cutting lathe manufactured by Danish company Ortofon, which included its STL631 Treble Limiter as part of the valve-amplified cutter control circuit. With a patch, it was possible to use the 631 as a stand-alone unit, and some are still in use today — not just in vinyl mastering, but also recording.

The earliest reference I can find to a proprietary stand-alone unit is the Orban 516EC Dynamic Sibilance Controller made in San Francisco in the mid-1970s. It was a FET model designed, says Orban, for “the recording and motion picture industries.”

Prior to the wide availability of purpose-built units, sibilance reduction was usually achieved by employing a compressor with a high pass filter inserted in its side-chain. David Nicholas (INXS, Pulp) says that way back at the dawn of time he used, “something or other by Valley People, probably a Dynamite from the ’70s configured to work as a de-esser, and it was all a bit troublesome and complicated to set up. But when the classic dbx 902 came out in the ’80s, that was it, and every rack in the world had a pair of them.”

I asked him about the early reputation of de-essers, as a box that didn’t really do much. His response was quick, and to the contrary. “One of my worst nightmares was too much de-esser during the tracking of Midnight Oil’s Blue Sky Mining, making the word ‘save’ sound a bit like ‘shave’. It was tape, so we couldn’t doctor it like you can with digital. The fans picked up on it and used to sing along at the gigs… “who’s gonna shave me!” It became a hit, but I lost a great deal of sleep over it. Since then I have never used a de-esser during tracking; only on the back end. With a de-esser, if you can hear it working, then it’s too much.”

FACTORY FAB’D FRICATIVE FIXERS

Simplified, a de-esser is a compressor in which the control circuitry is triggered by excessive amplitude within the fricative range of frequencies. As in the example above, a normal compressor can be used if the audio path to the gain reduction element is interrupted by a high pass or band pass filter, causing the compressor to attenuate only when the signal amplitude crosses a threshold within that band. If the attack and release times are set to react quickly, short fricative transients can be reduced substantially without affecting the audio that precedes or follows. Typical attack times are usually a millisecond or less, with release times varying between two and 50ms. In a proprietary de-esser these times are fixed, depending on what the designer finds pleasing to the ear.

These days, a device like the makeshift one above is usually described as a ‘high frequency limiter’ and a reasonable quality de-esser is likely to include additional refinements. Principal among these is ‘dynamic threshold’ where fricative energy is compared with overall energy, so that gain reduction only happens when high frequencies exceed some given ratio against the full bandwidth signal. If a singer crooning a verse suddenly belts out a much louder chorus, the increase in general volume would put more signal across the threshold of a high frequency limiter, and result in more de-essing. However, since louder singing causes negligible change in the ratio of high to low vocal frequencies, a dynamic threshold promises to keep the amount of de-essing more or less consistent.

In 1980, dynamic threshold was a big feature of the dbx 902, and worked so well that it was assumed to be a dbx invention. But other manufacturers had already employed versions of it, including the Orban 516 of the previous decade. The real achievement of the 902 was to get the ratio comparison bits working over a 64dB range — wider than any likely musical performance — so the unit could almost be treated as ‘set and forget’ within the signal chain. In the 1980s such functionality was welcomed, since it was still common practice to record a whole band at once, and a sound engineer had plenty of other places to look during a take.

The dbx 902 uses a voltage controlled amplifier, or VCA, as its gain reduction element. A VCA is an amplifier in which gain is kept proportional to an external voltage applied to its control terminal. Valley People boasted a proprietary VCA in its Dynamite compressors of the late ’70s, but again it seems dbx was the one to perfect this technology, which it also used in the legendary dbx 160 compressor. These days if you want to buy a VCA to solder into your gadget, you would look no further than those made by THAT Corporation, a company founded by guys who used to work at dbx.

The 902 uses a 12dB/octave active crossover circuit to divide the audio signal into high, low, and full bandwidth signals, which are dealt with separately (see diagram). Today the dbx 902 is superseded by the 520, made to the same recipe but with updated components.

The 902 set a benchmark, but other manufacturers remind us there are different ways to do it.

The BSS DPR-402 dynamics controller incorporates two separate side-chains — one for compression and one for de-essing — which simultaneously dictate gain reduction to a
single VCA.

Empirical Labs makes a number of VCA de-essers. The Lil FrEQ includes de-essing as one function within a tone equaliser unit, and allows operators to switch between dynamic threshold de-essing, or high frequency limiting, as desired. The DerrEsser includes a ‘listen’ button that allows you to preview the audio content being removed.

The Avalon VT-747SP is an optical compressor which allows independent threshold control for high and low frequencies, so it can be set up as a high frequency limiter.

My own Prodigal de-esser uses a FET as the control element, with dynamic threshold created by adding the averaged full bandwidth signal to a user threshold setting — the ‘ESS’ knob. Turning clockwise, the result is more essy, and anti-clockwise, less essy.

DE-ESSING UP FOR FAMILY OCCASIONS

De-essing is not limited to the reproduction of the human voice, but is also useful for other sounds containing high frequency fricatives, such as a plectrum as it thrashes across acoustic guitar strings, or to remove hi-hat bleed from a snare drum mic.

I recently recorded Mrs Tech Bench, who is a great lover of Himalayan music. She plucked through a few hot takes on her dramyin using a traditional bone plectrum, which can sound a bit peaky, especially if you are using a condenser mic. My solution was to slap a de-esser across it, which worked a treat. Playing it back, I was impressed at how the subtle reduction in pick noise gave way to the warm, full-bodied sound of the instrument. “How’s that?” I asked. “It sounds the same as before,” she said.

Perhaps Fred was an ’essing genius after all.