Supercardioid Camera Microphone

POWERING UP

A fulfilling ‘unboxing’ experience isn’t the be-all and end-all but when you’re purchasing a topshelf product it’s becoming an expectation. Every other Apogee device I have reviewed (including the Symphony I/O MkII) has arrived packed in the most pedestrian fashion, which has made for an underwhelming unboxing experience. I am delighted to report Symphony Desktop presents a far superior first impression. The magnetic box flap opens to moulded foam covers in which the unit sits snugly with a ‘Powering Up’ guide laid over the screen. Two USB-C cables plus the power adapter are included, with a selection of plugs for the latter. I love the twist-and-lock mechanism on the DC power plug at the interface end – much neater than the screw collar and equally reassuring.

Setup wasn’t as easy as I hoped. A firmware update is the first order of business and Apogee has a slightly roundabout way of doing it – you need to copy a downloaded folder onto the included USB thumb drive, which then plugs into the Symphony Desktop to run the update. Confusingly, the interface refused to recognise the USB thumb drive. After about five attempts I realised this was because it had been formatted incorrectly. Nothing Disk Utility couldn’t fix in a jiffy but it was inconvenient nonetheless. The only other niggle is when switching on the interface, it emits a very audible hiss through the monitor outputs which disappears after the roughly 30-second power up is complete.

Any setup woes were promptly forgotten when I first pushed some tunes through the Symphony Desktop. Although I have reviewed the exact same conversion with the I/O Mk II, the clarity and purity caught me off guard and I sat there experiencing a mix of delight and surprise as my monitors leapt to life. The old ‘blanket off the speakers’ analogy has been used to death but it’s quite an accurate representation of how it feels hearing music through Symphony Desktop’s output conversion. Realism, depth, width and immersion – it’s all undeniable and utterly enjoyable.

TOUCHY BUSINESS

What’s almost as impressive as the Symphony’s output conversion is just how much control is possible with the touchscreen. At the time of writing, Apogee has not yet released the Desktop Control software which enables control of the Symphony Desktop’s functions from your computer, so at present all adjustments have to be made on the touchscreen. Thankfully, it’s a breeze and I didn’t once miss using a mouse. You can route I/O from and to virtually anywhere, create a separate tracking mix for a singer, adjust levels for each track, select which preamp emulation to apply on an input, and even tweak the DSP-powered plug-ins.

You’d think doing all of that on a little touchscreen would be laborious but that’s not the case. The GUI is laid out clearly and feels responsive. Tap one of the labelled circles along the bottom to select which input or output you want to modify then swipe left or right to access the various parameters. Apogee has wisely made the physical knob the primary source of control – as opposed to sliding your finger around to adjust on-screen dials or faders – tap a dial on-screen and the orange outline denotes which knob is ‘active’.

HORSEPOWER

Throwing real-time DSP capabilities into an interface is no longer simply fashionable – it’s almost expected. Apogee joins competing brands such as UAD, Antelope Audio, Avid, RME and MOTU by adding natively driven plug-ins to the Symphony Desktop experience. Both analogue inputs can run the Symphony ECS Channel Strip which has been tuned by Bob Clearmountain and sounds the business. Again, it was a cinch to wind up a little high-shelf sizzle and a touch of light compression and saturation all from the touchscreen. Two vintage preamp emulations are accessible on the two analogue preamps too – AP-66 (a Neve 1066 clone) and AP-57 (an Ampex 601 clone). The latter offers an edgy harmonic distortion when driven hard while the Neve is purer and fuller. The ‘vanilla’ Symphony Desktop preamps are fantastic as well; 75dB of clean and quiet gain.

MIXING IT UP

Symphony ECS Channel Strip and Clearmountain’s Spaces plug-ins come as standard with a Symphony Desktop purchase but Apogee sells optional extras such as the FX Rack and Clearmountain’s Domain. These are all powered by the DSP chip inside Symphony Desktop, and they can run natively too.

Mixing my church’s pre-recorded Christmas Eve concert was the perfect opportunity to put Apogee’s plugs to work. Apogee FX Rack is a particularly useful tool with five processors including a Pultec-modelled EQ and compressor, and an LA-3A emulation, all of which have a distinct flavour and can quickly take a track from good to great. Mod EQ-6 and Mod Comp are perfect for more surgical tasks. The Symphony ECS Channel Strip (the same one that can run as a real-time effect on Symphony Desktop’s inputs) is another easy way to pull a track together quickly. It has a neat layout, fewer parameters than FX Rack and a transparent character. I love the Drive knob in ECS to add subtle and musical distortion to a ‘boring’ track. In the reverb department, Clearmountain’s Spaces impressed me immediately with its lush and unobtrusive tails which worked wonders on lead and backing vocals. Finally, Clearmountain’s Domain is a truly stunning reverb which boils down Bob’s signal chain into a few processors: input, delay, pitch/reverb and a mixer. A variety of atmospheres and textures can be created (and recalled via presets) to effortlessly lend your mixes a sense of depth and space.

SIBLING RIVALRY

After a few weeks of use, here’s why I believe the Apogee Symphony Desktop offers incredible value with even broader appeal than the Symphony I/O Mk II. 

Let’s say you’ve had enough of ‘prosumer’ interfaces and you’re ready to splash out on a substantial upgrade. Obviously, the dealbreakers would vary depending on our needs, but here’s how Symphony Desktop meets my ‘non-negotiables’.

D/A conversion? Check. In my opinion Symphony Desktop is worth the cash just to sit back and listen to your favourite tunes in all that glorious detail. 

A/D conversion? Check. 123dB is not be sneezed at, and the more you track through Symphony Desktop’s analogue inputs the more you’ll realise what you’ve been missing.

Inputs and outputs? Check. 10 of each will suffice for most home recording types, given there’s an eight-channel ADAT preamp to make the most of that digital input. 

Hardware DSP? Check. Who doesn’t love the idea of adding a tasteful touch of filtering or Neve saturation on the way in? Symphony Desktop gives it to you, then as a side benefit, it’ll power plugs within your DAW too. 

What Symphony Desktop can’t be, is the ultimate bridge between your boutique collection of preamps and the digital world of 1s and 0s. Multiple channels of line level conversion is the I/O MkII’s territory.

But all things considered, for less than A$2.5k, the Apogee Symphony Desktop is the closest I’ve come to experiencing the ultimate home studio interface. 

FACTORS AFFECTING TONALITY

Condenser microphones commonly used for professional audio come with diaphragms varying in diameter from under 5mm to over 25mm. The diaphragm’s diameter affects the performance in numerous ways. From a tonality point of view, however, the most significant affects are on the high frequency performance and the self-noise.

DIAPHRAGM DIAMETER

The high frequency limit of a condenser microphone with a circular diaphragm is primarily determined by the relationship between its diameter and the wavelength of the frequency being captured. At frequencies where the wavelength is considerably larger than the diameter, the diaphragm is too small to have any affect on the tonality of the captured sound. At frequencies where the wavelength is considerably smaller than the diameter, the diaphragm is big enough to be an obstacle and therefore have a significant affect on the tonality of the captured sound. And at frequencies where the wavelength is the same size as the diameter, a situation can occur where the diaphragm is receiving equal amounts of energy from both halves of the waveform (positive and negative), which cancel each other out and create a null (i.e. zero output). This frequency – where the wavelength is equal to the diameter of the diaphragm – theoretically determines the upper limit of a condenser microphone’s frequency response because it defines the first null point, but it’s not that straightforward…

Unlike the simple maths for determining a ribbon microphone’s high frequency limit [as shown in the second instalment], the high frequency limit of the condenser diaphragm is also affected by the angle of incidence and the way the diaphragm is mounted. For any given diaphragm, the high frequency limitations will be different in a side-address configuration than in an end-address configuration, and different again if mounted in a diffraction sphere or fitted with an equalising ring or diffusion grid.

It gets complicated when trying to define the upper frequency limit of a condenser microphone with a simple formula or rule. It is, however, worthwhile putting the concept of ‘wavelength’ into perspective with diaphragm diameter. The range of human hearing is considered to exist between 20Hz and 20kHz. Assuming a room temperature of 21°C, 20Hz has a wavelength of 17.2m – that’s how long one cycle is in the air. It’s significantly bigger than any practical microphone. At 20kHz, however, the wavelength is only 17.2mm long. It’s significantly smaller than the 25mm diaphragms commonly seen in dual-diaphragm microphones and is therefore affected by them.

Based on this knowledge, we’ll use 17mm (close enough to the wavelength of 20kHz) as a threshold point for differentiating between small and large diaphragms. There are no hard-and-fast rules for this, but condenser microphones with diaphragm diameters less than 17mm are generally described as SDCs (Small Diaphragm Condensers) while microphones with diaphragms of 20mm or more are generally described as LDCs (Large Diaphragm Condensers). Condenser microphones with diaphragm diameters less than 17mm usually exhibit extended high frequency response and better off-axis response when compared to those with diaphragms larger than 17mm.

Due to the very light weight of the diaphragm and the potential to make it very small, it is not difficult to make a condenser microphone with a frequency response extending up to 20kHz – the upper limit of human hearing. Maintaining a consistent response beyond 20kHz becomes more of a challenge, but is important for those who need to record sounds with content above the range of human hearing; whether to slow it down to create sound effects for movies and games, pitch-shift it down to analyze bat calls and insect sounds, or make ‘high resolution’ recordings for the audiophile market.

OFF-AXIS RESPONSE & TRANSIENTS

A sound arriving from directly in front of the microphone, perpendicular to the diaphragm, is considered to be on-axis, and is said to be arriving at the diaphragm from an angle of 0°. For the purpose of this discussion any sound arriving from an angle other than 0° can be considered to be off-axis because it is no longer coming directly on-axis. (This is discussed in a later instalment where we look at microphones’ polar responses and acceptance angles.)

As the diaphragm gets larger it becomes more directional at higher frequencies. As a result, sounds arriving significantly off-axis will be duller than sounds arriving on-axis. This is generally referred to as a poor off-axis response and results in a reduction of overall high frequency energy relative to low frequency energy captured by the diaphragm. Poor off-axis response at high frequencies is a characteristic of all large diaphragm mics and fuels the myth that large diaphragms have better low frequency performance than small diaphragms, when the reality is that they have worse high frequency performance (as discussed in the previous instalment, see ‘Subkick microphones’).

As the diaphragm gets smaller its off-axis response improves. This is one of the reasons why small diaphragm condensers are often preferred for main stereo pairs when making distant-miked recordings of orchestras, choirs, pipe organs and chamber music – all situations where sounds arriving from all directions (including early reflections, reverberation and audience reaction) need to be captured with a consistent tonality. If not, the sounds arriving off-axis will be duller than those arriving on-axis and result in a sound that could be described as ‘muddy’ or ‘roomy’.

A microphone’s off-axis response is not much of an issue when close-miking a single instrument, but it can be an issue if you are close-miking a number of instruments playing together in the same space. After doing what you can to minimise spill by the strategic use of instrument placement, baffles and polar response rejection (as explained in later instalments), you’ll want the remaining spill to be captured clearly so it doesn’t muddy up the sound when all the mics are blended together in the mix. The better off-axis response of smaller diaphragms can have a cumulatively positive effect in this situation.

As a generalisation, the large diaphragm condenser’s reduced high frequency capture and accompanying slower transient response means it tends to sound mellower and potentially smoother than a small diameter condenser, but with inferior off-axis response. Regardless of the diameter, however, the condenser microphone’s diaphragm is always going to be significantly lighter than a similarly-sized dynamic microphone’s diaphragm/coil assembly and can therefore move much faster, giving it excellent high frequency performance and transient response in comparison.

EQUIVALENT NOISE LEVEL

In ribbon and dynamic microphones the primary source of noise is thermal noise, as explained in the previous two instalments of this series (1) (2). This form of noise is rarely mentioned in their specifications because it is very low and will be insignificant compared to the noise from the preamplifiers they are connected to – which tend to be used at higher gains than they would be with condensers due to the lower outputs of ribbons and dynamics.

In condenser microphones the term equivalent noise level usually refers to the combination of the diaphragm’s self-noise and the noise from the microphone’s electronic circuitry. Self-noise refers to noise caused by air particles randomly striking the surface of the diaphragm. This form of noise is not a problem with dynamic microphones because their considerably heavier diaphragms are less sensitive to the random impacts of air molecules. It’s also not a problem with ribbon microphones because their ribbon elements are not held under enough tension to make them sufficiently responsive to random impacts of air molecules.

Self-noise and equivalent noise level are explained in detail in the following instalment of this series. For now, the important thing to know is that with all other parameters being equal, a large diaphragm will have less self-noise than a small diaphragm. It will also have greater sensitivity than the small diaphragm (meaning it provides a higher output level from the same sound source) but with a lower maximum SPL. So if you need to make a very quiet recording or if you need to record something that has very low volume, a large diaphragm condenser is a better choice than a small diaphragm condenser; it is quieter due to its lower self-noise, and its higher sensitivity requires less preamp gain and associated noise. If you need to record something very loud, a small diaphragm is a better choice; it can handle higher SPLs with ease, its lower sensitivity means less chance of overloading your preamp, and its higher self-noise won’t be an issue with a loud sound source. There’s more discussion about this kind of decision making in a later instalment of this series…

The noise from the electronic circuitry inside the microphone also needs to be considered. All active electronic circuits create noise – especially when that circuit has to work with the very small signals coming from a microphone capsule. (For many years this form of noise was unique to condenser microphones, but nowadays it also exists with active ribbon mics, active dynamic mics, USB mics and any other mic that has electronic circuitry in it.)

A presentation by Neumann’s Martin Schneider compared the frequency spectrum and amplitude of the diaphragm’s self-noise and the internal electronics’ noise for a large diaphragm condenser with a very low equivalent noise level. When averaged across the audible spectrum from 20Hz to 20kHz, the noise from the microphone’s internal electronics was approximately 3dB lower than the diaphragm’s self-noise. The large diaphragm microphone used in this example has an equivalent noise level of only 7dBA, therefore the noise from the microphone’s internal electronics (being on average just 3dB lower than the diaphragm’s self-noise) could be considered significant. Microphones with very small diaphragms (i.e. 6mm or less), such as Earthworks’ QTC30 or DPA’s 4060, have equivalent noise levels of 20dBA or more, most of which is self-noise. In these examples the noise of the microphone’s internal electronics is probably insignificant.

RESONANCE

Because the diaphragm is usually a circular membrane held under tension, it behaves like a tiny drum (imagine a miniature roto tom) with a strong primary resonant frequency, or note, of its own. Any sound that contains frequencies at or near the diaphragm’s resonant frequency will cause the diaphragm to resonate, emphasizing those particular frequencies above others and thereby adding to the microphone’s characteristic tonality.

The condenser microphone designer tweaks the diaphragm’s diameter, mass and tension to determine its resonant frequency – which typically ends up somewhere between 5kHz and 9kHz – and then decides how much damping should be applied to control the resonance. Increasing the damping reduces the strength of the resonance (which is equivalent to lowering the amount of boost on a peak/dip equaliser), but also reduces the diaphragm’s overall sensitivity (which is like lowering the fader). There is always a compromise between resonance and sensitivity, hence the market is full of condenser microphones with a characteristic boost in the upper midrange or high frequencies – often engineered and marketed as a feature to provide enhanced detail or to compensate for the loss of high frequencies through the air when distant miking. This balance of resonance and damping is another factor that determines how the condenser mic affects the tonality of the captured sound.

Most condenser diaphragms are ‘edge terminated’ which means the electrical connections (or ‘terminations’) to the diaphragm and backplate are made at the edges, leaving the diaphragm free to move. Some are ‘centre terminated’, which means the electrical connection is made to a terminal in the centre of the diaphragm – as seen in Neumann’s classic U47 and U67 mics respectively. This is like lightly placing your finger in the centre of a drum skin and listening to how it affects the resonances and therefore the tonality; it changes the balance of the diaphragm’s resonant modes, but in the case of the U47 and U67 it’s just one of many things that affect or determine their tonality. Another is, of course, their tube circuitry…

INTERNAL ELECTRONICS

Every circuit that an audio signal passes through imparts its own sonic fingerprint on that signal (typically as a combination of its inherent harmonic and intermodulation distortions), and every condenser microphone includes an impedance converter circuit that the signal must pass through in the process of becoming useable. That circuit will contain numerous electronic components, but the primary component will be an amplifying device of some kind (tube, FET, transistor, etc.) that imparts its own sonic fingerprint, contributing to the microphone’s overall tonality.

As with ribbon and dynamic mics, for many years a transformer was used to give the condenser microphone a balanced differential output suitable for sending a signal down a microphone cable and into a preamp. Contemporary designs use a solid state electronic circuit and don’t require an output transformer, although many still use one. The acronym ‘TLM’, used in the model names for many of Neumann’s contemporary microphones, stands for ‘Transformer Less Microphone’. Similarly, when DPA made an updated version of their classic 4006, ‘TL’ was appended to the model number to indicate it was ‘Transformer Less’. According to DPA’s documentation, removing the transformer allowed the 4006TL’s low frequency response to be extended downwards another octave from 20Hz to 10Hz.

Some condenser microphone manufacturers have returned to using transformers primarily for their ‘old school’ tonal properties, or simply to take advantage of the transformer’s relative simplicity for providing a balanced differential output. When Audio-Technica designed the 5047 – a premium version of their 5040 – they added a transformer output to maintain a ‘constant load output impedance’ and provide a ‘smooth sonic character’. [The effect of a varying load impedance was discussed in the second instalment of this series; it is the reason why the tonality of a passive ribbon microphone can be affected by the preamp it is connected to.]

DYNAMIC TRANSDUCTION

Like the ribbon microphones described in the previous instalment, dynamic microphones use magnetic induction to convert a sound into a signal. Instead of using a corrugated ribbon of aluminium, however, a coil of wire is attached to the back of a diaphragm (a circular membrane suspended in the air). The diaphragm is dome-shaped which, among other things, provides the structural integrity needed to support the coil. As with a ribbon element, the coil is immersed in a magnetic field. Sound vibrations cause the diaphragm to vibrate, moving the coil within the magnetic field and inducing an electrical current into it. For this reason, dynamic microphones are sometimes referred to as moving coil microphones. Every winding in the coil has a current induced into it, which collectively adds up to a higher induced current than could be created with a ribbon element in the same magnetic field (a ribbon is essentially just one winding of a coil).

Theoretically, the dynamic microphone doesn’t need a transformer because the diaphragm’s coil is fundamentally the same as the output of a transformer and can be connected directly to an XLR socket to provide a balanced differential output. However, many dynamic microphones rely on a transformer nonetheless for similar reasons as the ribbon microphone: to increase the signal level, provide a more appropriate output impedance, and provide a balanced differential output that’s suitable for driving the signal down a microphone cable and into a preamp. Electro-Voice’s RE20 and Shure’s SM58 both use transformers, while Electro-Voice’s RE320 and Shure’s Beta 58 do not. (Interestingly, Shure went back to using a transformer in the Beta 58A.)

One popular modification for Shure’s SM57 is to remove the transformer, which reportedly smooths out the frequency response and improves the high frequencies – at the expense of losing the 10dB of voltage gain provided by the transformer. There is plenty of ‘how to’ information online for this mod, and also for replacing the transformer with alternative versions, but be aware that these mods will void the microphone’s warranty and also change its characteristic tonality – which was probably what you bought it for in the first place. An in-line booster (see below) might be a better option…

FACTORS AFFECTING TONALITY

Although the dynamic microphone uses the same principle of magnetic induction as the ribbon microphone, its considerably heavier coil/diaphragm assembly results in reduced sensitivity to high frequencies and slower response to transients. Unlike the gentle high frequency roll-off of ribbon microphones or the extended high frequency response of condenser microphones, dynamic microphones roll-off rapidly – rarely reaching 20kHz at any notable level.

RESONATORS & DAMPERS

The dynamic microphone has a number of significant resonances; in fact, its entire frequency response could be considered as a series of controlled resonances. The first is the resonance of the diaphragm itself, which is often illustrated as being centred somewhere around 800Hz, depending on the mass, diameter and suspension. This resonance creates a large peak in the frequency response that resembles an inverted ‘V’, and is far from the straight horizontal line that’s considered ideal. Acoustic damping is applied to reduce this peak considerably. Although it is possible to damp the diaphragm’s resonance down to a flat response, increasing the damping decreases the diaphragm’s sensitivity and results in lower output so it’s always a compromise.

Other resonating systems and dampers are added as a form of passive EQ to extend the overall frequency response. One of these is a narrow tube that connects the air chamber behind the diaphragm to a larger acoustic chamber inside the mic body, forming a Helmholtz resonator (similar to a ported speaker enclosure) that boosts the low frequencies and balances them against the diaphragm’s damped resonant peak. Likewise, a resonating cavity is built into the dome-shaped space inside the diaphragm to boost the high frequencies and balance them against the low and midrange frequencies. The result of these three resonating systems – the diaphragm, the Helmholtz resonator and the resonating cavity behind the diaphragm – provides reasonably good bandwidth.

Additional resonators and dampers may be added to further enhance the performance and/or give the microphone its characteristic tonality. For example, beneath the wire mesh and internal foam wind filter of Shure’s SM57 is a small plastic disc with holes around the perimeter. It sits immediately above the diaphragm, assumedly to protect it from the foam. The underside of this disc, however, has a concave dip in the centre with a curvature that follows the top of the diaphragm; this creates a small resonating chamber immediately above the diaphragm that’s tuned to extend the high frequency performance. This is known as a resonator cap and is commonly used in dynamic microphones – look for a small dome of some kind sitting immediately above the diaphragm.

The result of these controlled resonances working together is a far better frequency response than the diaphragm alone can provide. Although the quoted frequency response for a typical dynamic microphone extends higher than that of a typical ribbon microphone, once the dynamic microphone reaches its quoted upper limit the high frequency roll-off is usually steeper. Fortunately, dynamic microphone designers are usually able to place the high frequency roll-off somewhere above 12kHz, which is high enough to avoid terms like ‘dull’ and ‘muffled’ – descriptions that would certainly be used if the same steep roll-off occurred just a bit lower.

The inside of a dynamic microphone is a work of miniature acoustic engineering art. A look inside Sennheiser’s 421 quickly explains its size and shape, with a bass pre-emphasis pipe from the back of the capsule feeding into a large acoustic chamber shaped to form a broadband resonator, along with four rear vents/ports, a resonator cap and numerous damping systems. There’s also a humbucking coil, a transformer and a five-position bass roll-off control. Even Shure’s visually simple SM57 contains numerous resonating cavities and damping materials, including the cavity inside the handle itself – as Granelli Audio Labs discovered while prototyping their G5790 (a modified Shure SM57 with a right-angle bend to make it easier for close-miking drums). They had to re-work their initial design to make sure it didn’t alter the original tonality of the SM57 – otherwise they wouldn’t be able to market it as “The mic you’ve always loved, the mod you’ve always needed.”

DIAPHRAGM PARAMETERS

Diaphragm diameter also plays a role in the tonality of a dynamic microphone, especially when considered with the size/weight of the coil. A larger diaphragm can support a larger coil which potentially means a higher induced current, but with reduced high frequency sensitivity due to the increased weight. The smaller and lighter the diaphragm/coil assembly is, the faster it can move – resulting in better high frequency performance and transient response. BeyerDynamic’s M201 is a relatively small dynamic microphone (outer dimensions measure 160mm x 24mm), presumably containing a relatively small diaphragm – or, at least, a very light one. Often recommended for snare and acoustic guitar, the M201 offers high frequency performance and transient response approaching that of a large diaphragm condenser.

You can’t always judge the diaphragm size by looking at the microphone itself, however. A dynamic microphone’s physical size is usually due to the resonating chambers and other acoustic work going on inside it, and is not an indication of the diaphragm size itself. This is especially the case with microphones designed for capturing low frequencies (e.g. kick drums) where large resonating acoustic chambers are used inside the microphone to extend and enhance its low frequency performance.

A look inside a physically large dynamic microphone such as Shure’s Beta 52A kick drum mic or AKG’s D12 series (D12, D12E, D112, etc.) reveals a much smaller diaphragm than the microphone’s physical size would suggest, but there’s a lot of other things going on inside to support it acoustically, electronically and physically. Many people are surprised to see that the capsule in Shure’s Beta 52A is approximately the same size as the capsule in Shure’s Beta 57A handheld vocal mic, while the capsule in AKG’s D12 is marginally larger but still significantly smaller than its housing would suggest. The myth that you need a large diaphragm to capture low frequencies is explored in more detail below (see ‘Subkick microphones’).

Most dynamic microphones use diaphragms between 12mm and 18mm diameter. Going larger than 18mm presents engineering challenges to keep the diaphragm/coil assembly light enough to do its job without flexing or collapsing under its own weight. Heil Sound’s PR series feature very large 38mm diaphragms with varying thicknesses to keep them as light as possible while providing strength where necessary; other dynamic microphone manufacturers use similar approaches when making large diameter diaphragms.

NOISE & DISTORTION

As with ribbon microphones, the dynamic microphone’s transducer has no inherent source of noise apart from the thermal noise (aka Johnson noise) that exists in all electrical circuits, which is due to the effect of temperature on the microphone’s electrical circuitry. The current induced into the dynamic microphone’s coil is considerably higher than the current induced into a ribbon element, resulting in a greater signal level (from the same source) that makes the thermal noise less significant than it is in a ribbon microphone. Also, dynamic microphones are generally used for close-miking relatively loud sound sources (drums, guitar amps, live vocals, etc.), resulting in a higher induced current that further reduces the significance of the thermal noise. In any sensible dynamic microphone application, the dominant noise should be from the chosen preamplifier.

The main forms of distortion in dynamic microphones are due to the diaphragm/coil assembly being pushed to extremes so it is no longer behaving in a linear manner; flexing the diaphragm or even reaching the end of its excursion and ‘bottoming out’ – however, it would take a very high SPL to push a dynamic microphone to audible extremes, and even more to damage it. There is also low-order harmonic distortion introduced by the transformer – particularly due to saturation from transients and high levels of low frequencies. The harmonic distortion of the transformer may be a contributor to the microphone’s overall tonality – particularly with transients and higher signal levels.

SPECIALISATION

The engineering challenges contained within the dynamic microphone’s transducer make it difficult to create a theoretically perfect microphone, but they’re also a convenient justification for making specialised microphones that exploit the transducer’s strengths and avoid its weaknesses. By juggling the diameter and mass of the diaphragm, the number of windings on the coil, the size and strength of the magnet, and the use of resonators, dampers and acoustic chambers, the dynamic microphone designer can create all kinds of specialised microphones – from large mics with mellow tonalities and rich low frequency responses suited for kick drums and basses, to small mics with bright tonalities and fast transient responses for snares and acoustic guitars. From a tonal point of view, some dynamic microphones rival condenser microphones when used in their specialised applications.

ADVANCES & ODDITIES

Although the dynamic microphone as we know it today can be traced back to the 1930s there have been many advances and improvements, along with one or two oddities…

RARE EARTH MAGNETS

As with ribbon microphones, the use of rare earth magnetic materials such as neodymium and samarium-cobalt – with their considerably stronger magnetic fields – offers the dynamic microphone designer numerous options.

Electro-Voice was the first microphone manufacturer to make a feature of neodymium magnets; their N/DYM series of dynamic microphones, released in 1986, had surprisingly higher output signals and better high frequency performance than most other dynamic microphones on the market at the time.

Shure’s Beta series, released in 1989, began life as neodymium versions of their most popular dynamic microphones (wisely released as entirely new mics rather than replacements), with higher outputs and better high frequency performance. For example, the Beta 58A offers 3dB higher output than the SM58 and its quoted frequency response extends to 16kHz rather than the SM58’s 15kHz.

OVERCOMING PROXIMITY EFFECT

Electro-Voice have been tackling the proximity effect in directional microphones for decades, dating back to their 664 dynamic cardioid from the mid 1950s. Initially intended for sound reinforcement applications, it was the first microphone to feature their ‘Variable D’ (‘D’ for ‘distance’) technology which conceptually uses multiple frequency-selective rear ports (rather than a single port as is commonly used) to create a cardioid response. The result is a cardioid microphone with almost no proximity effect; there is no significant bass boost or bass loss when moving closer or further from the microphone, meaning an instrument on stage could be very closely miked without the exaggerated bass boost and related low frequency feedback issues.

The Variable D technology continued with the 665 and 666, both intended for the broadcast market. The 667 followed and came supplied with its own transistorized preamplifier that offered numerous frequency response curves via two rotary switches: a four-position rotary switch for the low frequency response and a five-position rotary switch for the high frequency response. Together, these switches offered 20 different frequency response options, including flat, and numerous options for boosting and cutting the low, high-mid and high frequencies. In a simple yet practical design move, the different response options were displayed on the front panel so it was easy to ‘dial up’ the desired choice.

The most recognised of the Variable D microphones would be the RE20 and its derivatives. Its lack of any significant proximity effect has made it popular with radio announcers, television presenters, podcasters and others who need to move around while sitting and talking – rather than talking directly into the microphone from a fixed distance at all times. It also has excellent high frequency extension and a very consistent polar response across a wide range of frequencies – two distinctions it holds with few other dynamic microphones (Sennheiser’s MD441 comes to mind). No wonder Electro-Voice nicknamed it ‘the condenser killer’…  

HUMBUCKING

Although humbucking or ‘hum compensation’ coils are generally associated with electric guitar pickups, their first use in sound applications dates back to Electro-Voice’s V1 ribbon microphone from 1934 (see previous instalment), a year before the first patent for the humbucking guitar pickup. Humbucking coils are used in many of the most popular dynamic microphones currently on the market, and are usually listed among the microphone’s features.

The basic concept is simple: add an extra coil with the same characteristics as the coil that’s attached to the diaphragm, put it on the same axis as the diaphragm’s coil but don’t attach it to the diaphragm so it does not move through the magnetic field and therefore does not get the microphone signal induced into it. Any electrical interference (hum, lighting buzz, etc.) should be induced equally into both coils. Invert the polarity of the new coil, and then add the signals from the two coils together. The interference (which is in both coils but with inverted polarity in one of them) will cancel out, while the signal (which is only in one coil) will pass through to the output.

TWO-WAY MICROPHONES

As seen earlier, one of the problems with dynamic microphones is achieving an extended high frequency response and fast transient response with their relatively heavy diaphragm/coil assemblies. AKG addressed this problem with the use of two dynamic diaphragm/coil assemblies working together – one optimised for low frequencies and one optimised for high frequencies – with a filtering circuit merging the two signals together to create a two-way microphone. Discontinued long ago, there were three mics in the series – the D200E, the D202E and the D224E – with the most memorable being the D202E ‘sound rocket’, which dates back to 1966 and was ultimately replaced by the sleeker-looking D222E. Crossing over at 500Hz, the D202E offered a very flat response and remarkably consistent rejection to sounds arriving from the rear.

AudioTechnica took a similar approach with their AE2500 ‘dual element’ kick drum mic, which combines a dynamic capsule and an electret condenser capsule in the same microphone, but with a separate output for each capsule so they can be blended as desired. Sony’s C-100 uses two capsules in a two-way configuration with a single output, but they’re both condenser capsules. The C-100 is discussed further in the next instalment of this series.

PRINTED RIBBONS

An interesting but short-lived development in the design of dynamic microphones can be found in Fostex’s Regulated Phase (RP) series. These microphones used a flat circular diaphragm (as used in condenser microphones) with the ‘coil’ etched into a layer of foil bonded to the diaphragm itself – similar to the tracks on a printed circuit board – earning them the description of ‘printed ribbon’. The diaphragm was placed within the field of two circular magnets, and magnetic induction created the signal. The result was a series of microphones with the ruggedness of dynamic microphones, a transient response approaching that of condenser diaphragms, and a mellow tonality similar to a ribbon microphone. Among aficionados, Fostex’s M88RP remains an affordable collector’s item. Discontinued long before the ribbon microphone revival, it is worth speculating how successful the RP series would be if re-released today and taking advantage of later technologies such as rare earth magnets.

ACTIVE DYNAMICS

As with active ribbons, an active dynamic microphone includes internal electronic circuitry to buffer the signal from the voice coil, provide a stronger output signal, and remove the effect of the preamp on the microphone’s tonality. Blue were early adopters and perhaps pioneers of this idea with their Ball and Kickball studio microphones; their Encore 200 is a phantom powered handheld dynamic for live vocal applications. Lewitt Audio’s MTP 840 DM and AKG’s D12VR are both dynamic mics with active circuitry that can be turned on if desired, allowing them to be used as passive or active dynamics. Aston’s Stealth goes a step further: it’s a dynamic mic with a built-in Class A microphone preamplifier, again switchable in and out of circuit, that provides four switchable tonalities and a massive 50dB of gain.

USB MICROPHONES

An extension of the active microphone idea, the USB mic builds a preamp, converter and USB interface into the microphone so it can be directly connected to and powered by the USB input on a personal computer, laptop or mobile device. Many include a built-in headphone socket to allow latency-free monitoring while also working around the lack of headphone sockets on many mobile devices. USB mics eliminate the need for an interface for users who only need one microphone (solo musicians, podcasters, vloggers, etc.) and simplify the ‘plug and play’ recording concept considerably. It will be interesting to see if USB mic manufacturers adopt 32-bit gain-staging technology [as used in the AES42 digital microphones (see the following instalment) and in field recorders like Zoom’s F6 and Sound Devices’ MixPre series] to eliminate the need for gain control during recording.

The dynamic transducer is well-suited to the USB microphone because it does not require the high polarising voltage of the condenser microphone, has a higher output than a ribbon microphone, and is considerably more rugged than both. It’s also a popular choice for spoken voice applications that benefit from the simplicity of a direct-to-USB connection – such as podcasting and vlogging. Røde’s Podcaster is a good example; it uses a 28mm dynamic capsule, offers 18-bit resolution at sampling rates from 8k to 48k, and includes a 3.5mm stereo headphone output with volume control.

IN-LINE BOOSTERS

Although dynamic microphones don’t need in-line boosters to the same extent that vintage and passive ribbon microphones do, they can still benefit from them for the same reasons to preserve their characteristic tonality (see previous instalment). Any of the in-line boosters marketed for ribbon microphones should also work well with dynamic microphones, including Royer Lab’s dBooster, Triton Audio’s FetHead, Cloud Microphone’s CloudLifter, Radial Engineering’s McBoost and Klark Teknik’s Mic Booster CT1. It’s not unusual to see podcasters using in-line boosters with Shure’s SM7B, Electro-Voice’s RE20 and similar announcer/presenter microphones.

SUBKICK MICROPHONES

A moving coil loudspeaker is essentially a dynamic microphone operating in reverse, so it is conceptually simple to use one as a microphone by placing it in front of a sound source and connecting its terminals to a preamp input. The original subkick microphone was a DIY device made by placing a speaker in front a kick drum – something studio engineers have been doing for decades (Geoff Emerick reportedly used a speaker cabinet as a subkick mic on The Beatles’ ‘Paperback Writer’). In later years it became common to see the LF driver from Yamaha’s NS10 monitors clamped to a mic stand and being used as a subkick mic (this was convenient because every professional studio kept spare NS10 LF drivers on hand for emergency repairs).

Yamaha’s SKRM100 went considerably further than the DIY approach. In a design motivated by drummer Russ Miller, Yamaha improved and commercialised the subkick idea by mounting a 6.5 inch low frequency driver (white cone, of course!) into a standard 10-inch 7-ply birch/mahogany drum shell, fitting it with mesh heads and supplying it with a sturdy stand made from drum hardware. The drum shell, mesh heads and under-damped speaker cone all resonated in response to the energy from the kick drum and added their own tonalities to the sound captured from the kick drum, thereby creating a unique device for enhancing the low frequency reproduction of the kick drum (some well-known session drummers carry their own SKRM100 fitted with vintage hoops, claiming it extends the tone of the mesh heads). When mixed with a traditional kick mic placed inside the kick’s shell to capture the impact of the beater, the result can be impressive – particularly when given a small boost around 30Hz, which is an octave below the SKRM100’s significant 60Hz peak. Yamaha showed good judgement by describing the SKRM100 as a ‘low frequency capture device’ rather than a microphone. It has since been discontinued, but similar devices such as Solomon’s LoFReQ and DW’s Moon Mic are still on the market.

A Google search reveals plenty of DIY information for those interested in building their own subkick microphone, but be warned: the DIY subkick world is full of pseudo-science. While it is true that a large cone area is required to reproduce low frequencies, it is not required to capture them. A quick look at DPA’s 4060 lavalier microphone, with its tiny 5.4mm diaphragm that measures flat down to 30Hz before dropping to -2dB at 20Hz, puts a convincing end to that myth – as does comparing the capsules in Shure’s Beta 52A kick drum mic and Beta 57A handheld vocal mic, which are virtually the same size. The DIY subkick’s speaker cone does not capture any more low frequencies than any normal mic would capture in the same position; it simply captures less high frequencies and has very poor transient response. Blending this with a traditional kick drum mic placed inside the kick drum obviously results in more low frequency energy overall.