AT: Will there still be a standalone Fairlight audio system or will existing Fairlight users have to learn to drive a video/audio finishing tool?

BD: There will be a high level of faithfulness to existing Fairlight users. A Fairlight user is going to be able to load up Resolve and go for it.

AT: For the first time ever it may be on a Mac or Linux computer?

BD: Yeah. Last night I was working on a doco to upload to YouTube. The titles and audio levels were not quite right, so I copied the project from the PC I was using onto my Mac laptop, went home and kept working on it. Resolve archive files include used media so they are so easy to move around.

AT: Are there any assurances you can give to Fairlight users worried old projects won’t open
in Resolve?

BD: There are already scripts and tools in the background that are going to be available for that. There has to be compatibility between existing DR2 projects and Resolve. As for legacy MFX3 projects? I don’t know. A third party could write translation software for that.

AT: What about existing hardware compatibility and plans for future control and mix hardware?

BD: You’re way down the line with future hardware. Those policies are a work in progress. There are many generations of Fairlight hardware that work with version five software — older USB- and Ethernet-based models. With Resolve the line has to be drawn somewhere. I don’t see USB controllers being supported. However, USB to Ethernet upgrade kits exist for some USB controllers like the Xynergi panels. At NAB, Fairlight ethernet controllers were working with Resolve and the keys to switch to the Resolve pages were on those controllers. Also, iCan is now even more integrated with the software. [iCan is a control protocol allowing Fairlight controllers to drive other editing applications like Adobe Premiere – Ed].

AT: So if I have Resolve and Premiere on my machine, the existing Fairlight controllers can drive both NLEs via iCan?

BD: Those are all still technical possibilities. How we will use that is a work in progress. The sky’s the limit, we’re very excited.

PROBLEMS RESOLVED

Having the purchasing power and market muscle of Blackmagic behind Fairlight is a big plus for users of the DAW. As an existing Fairlight customer the future looks very good. Blackmagic brings hardware expertise, an astonishing speed of development, and a cost effectiveness that sees a full Fairlight system inside Resolve for US$299. As a Resolve user as well, it means for the first time I can do both high-end picture and sound post in the one software app without clumsily transfering between picture and audio post via unreliable interchange formats like AAF or OMF. It solves one of the weakest parts of the video post-production workflow. Exciting times for post people.

There are two gooseneck lengths — five- and nine-inch — and a hexagonal bar at the bottom of each fits into one of the many mounts. This is when the fun begins. These mounts are a gadget lover’s delight; beautifully detailed and discreetly finished in matte black paint. The AT8491U universal clip-on mount will be the most popular. It’s the general-purpose option, designed to clip onto stringed instruments or the bell of many brass and woodwind instruments. The gooseneck is inserted into the mount at the chosen angle. It looks hexagonal at a glance so I was expecting six possible angles, but on closer inspection it’s an odd shape and you only get four, which is enough, anyway. The gooseneck is secured in the mount with a nicely knurled knob. The jaws of the mount have rubber teeth with a strong bite. Another knurled knob tightens the grip and can really lock the mount onto the instrument, so there’s no chance of it flying off in the middle of even the most frenzied performance.

My old friends, My Old Friends the Chocolate Cake (hardly know them, just wanted to say it), were playing at the Theatre Royal and as luck would have it the violin player Hope Csutoros was happy to try the ATM350a. We used the AT8468 hook-and-loop violin mount first. Introduced as part of the previous model AT350 package it’s a Velcro wrap that straddles the strings between the bridge and the tail piece, the mic tucks inside a loop above the strings and points up at the instrument. Simple and quite discreet but it did seem to dampen the strings a little. Next we tried her normal technique, the clip-on mount attached to the violin’s chin-rest and pointing into the upper f-hole.

The sound of the ATM350a was immediately impressive; natural, with an appealing warmth and closeness that brought a smile to Hope’s face and won instant praise from the rest of the band. The perfect reaction for something new. The gooseneck can be used to explore mic placements to find the best combination of tone and volume, and it doesn’t wander once in position. Hope is an energetic performer but no match for the AT8491U mount once it had been locked on. It didn’t fly off once.

The frequency response has been tailored for live use with the deep low frequencies reduced to avoid rumble and limit the proximity effect. The high frequencies have been tamed a little to avoid the 10kHz-ish feedback that condensers sometimes induce, and the 2-4kHz region where live speakers like to take off has been slightly attenuated. Combine this with an accurate off-axis response and the result is excellent gain before feedback. The mic’s great… back to the mounts.

SCALING THE MOUNT

Next up was everyone’s favourite, the AT8491D drum mount, it attracted attention whenever I used it. More satisfying to play with than a fidget-spinner, and a lot more useful, its spring-loaded release attaches onto one of the drums tuning lugs. It’s secure, almost invisible and the mic will stay in the chosen position. I don’t think drummers like having things attached to their drums, it looks weird and those big rubber clips can change the sound. I prefer stands. This mount solves those problems. The clincher is the slip-on tension rod extender so you can still tune the drum with the mount in place. Ingenious.

The first night I used the AT8491D was with some real rock bands. Peep Tempel were the headline act with not an acoustic instrument in sight, so I thought I’d try the ATM350a on the hi hat, since it’s a condenser… but it had no mount to attach it to a mic stand, an obvious omission when Audio-Technica has tried to cover everything else, so I put it on the floor tom. The drummer approved of the mount and was suitably impressed with the discreet placement as well as the way the drum could be tuned with the mount in place. Floor tom for a thrashy rock band is a long way from a subtle violin, and the ATM350a was the opposite of the tight dynamic mic I’d normally use on the floor tom for that sort of band. The floor tom sounded huge and the spill was good too… quite a lot of it, of course, but it blended in for a big, open drum sound. The mic had no trouble with the volume or the full low-end; all in a day’s work, as they say.

The AT8491P piano mount is clever too, it may not be a unique concept but it’s an elegant solution to the problem of how to mount mics in a piano. Looking like the black queen in chess, the attractively shaped mount has a strong magnet hidden in its protective felt base and an integral shock-mount. The magnet sticks to the piano frame and secures the mount. The nine-inch goosenecks are recommended for this application. Even with the longer goosenecks the mounts are strong enough to hold the mics firmly in place and the look is super low-profile compared to two tall boom stands.

The AT8491W is a problem solver designed for woodwind instruments. Examples of mounting problems include the flare on the end of a clarinet being too wide for the basic UniMount jaws, or a flute that may be better captured by getting the mic nearer to the mouthpiece than the end. The AT8491W addresses these problems by attaching a rounded base to the body of the instrument with a hook-and-loop strap that passes through the base and wraps around the body of the instrument. The base has four soft rubber feet that touch the instrument and an integrated shock-absorber above the strap to reduce handling noise. The strap sticks to itself easily so there’s some skill required to get the fitting or adjustment process smooth, but once in place, it’s solid. The mic can now be placed to reach over the end of the flare as you would for brass instruments, or it can be placed to point back up the instrument, towards the keys, for a less aggressive sound, or aimed at the mouthpiece of a flute. Simple but well executed.

FLEXIBLE AS A GOOSENECK

The ATM350a is an excellent general-purpose instrument mic that will deliver the highest sound quality and robust amounts of gain before feedback in a live environment. Acoustic instruments and percussion have a low average level but can have surprisingly high peak transient levels. They won’t worry the ATM350a with its impressive 159dB maximum input level. The cardioid pattern gives a full and accurate view of the instrument for normal stage use but interchangeable elements are available with hyper-cardioid available for situations where players are very close to each other, and omni-directional for super-accurate proximity-less applications. The mounts let the ATM350a do its best work. Single-instrument players will choose the mount that suits their instrument, multi-instrumentalists may choose several, while venues or production suppliers could get the whole set for maximum flexibility.

ETCHED INTO HISTORY

The first recording equipment was not electronic, but entirely mechanical, and it was enough of an achievement just to have captured a sound and stored it.

In 1877, Thomas Edison invented a gizmo he called the Phonograph. It worked a bit like when you stretch a string between two tin cans and you can hear your cousin. The Phonograph allowed you to yell into a funnel, and the sheer force of your personality would cause a needle to cut a groove into a revolving tinfoil- or wax-coated cylinder. You could play it back via the same needle, using a hand-crank to rotate the cylinder, and the funnel would then serve as a horn, to amplify the sound back to the listener.

Initially there was no way of duplicating the cylinders, and if you wanted a hundred copies the artist would have to perform the work a hundred times. It took until 1901 to pull off making reproductions from a mould of the first cylinder, but it was a cumbersome and expensive process, and the recordings were only two minutes long anyway. Those difficulties aside, for fifty cents each, you could choose between marches, sentimental ballads, hymns, comic monologues, or a racist form of early ragtime known as coon songs.

Meanwhile, Emil Berliner of Washington DC had come up with a flat disc version of Edison’s idea which he called the Gramophone, which possessed an important advantage. After the needle scratched away the wax, you could acid-etch into the zinc plate underneath to easily create a reproduction master with at least three minute’s worth of material on it — more if you used a larger disc. This was technically the beginning of the record industry as we know it, but whether you were a banjo player or a symphony orchestra, you still had to shout down that funnel.

Enter Alexander Graham Bell. Back in 1877 when Edison was inventing the Phonograph (also making him the world’s first audio engineer), Bell was inventing the Telephone — a revolution in communications which would change the world as dramatically as the internet did again a hundred years later. Like Gates and Zuckerberg in powdered wigs, Bell and Edison were riding at the helm of a new era.

A year later Edison created the Carbon Microphone, which was arguably the first serious piece of pro audio kit ever. It consisted of carbon granules squished between two metal plates. Sound waves striking the plates varied the pressure on the granules, which changed the electrical resistance between them. Carbon microphones were a great leap forward. They were not high-fidelity devices, so their dominance in professional audio was limited, but their inherent simplicity and durability made them a standard component in telephones for the next hundred years.

Edison’s carbon microphones were also ingeniously reconfigured as the first electronic amplifiers. [1] Instead of using sound waves, a mechanical plunger was pressed against the carbon granules and, similar to a loudspeaker, varied its pressure magnetically in response to an audio source. The resulting signal that emerged on the wire could be as much as 100 times (40dB) louder than the original. These carbon amplifiers were used extensively in the new telephone networks, to boost diminishing signals in long-distance cables.

In the years that followed, it was all about the telephone, and most of the important advances in electronic audio and amplification were made by people working for two subsidiaries of AT&T, the company founded by Bell. Western Electric and Bell Laboratories variously developed the vacuum tube, the first condenser microphone, some of the first valve amplifiers, and even invented the decibel (named after Bell) and negative feedback, but it took a while for these advances to find their way back into the record industry.

The oldest known electronic recording is a document of the burial of the Unknown Soldier in Westminster Abbey, London, in 1920. The equipment used was a high output carbon microphone connected directly to a record cutting lathe. The result is noisy, fuzzy and indistinct, and like everything else, you can find a copy of it on YouTube. It also marks a starting point in the appearance of the first electronic recording systems, which were all ‘direct to disc’ and quickly saw the old mechanical technology wiped out of existence.

By 1925, Western Electric’s ‘Westrex’ electrical recording system had established itself as the market leader, and did away with carbon microphones in favour of the sonically superior condenser mics. Class A valve amplifiers then boosted the weaker signal up to cutting lathe level, for the production of a wax master. These kinds of systems were commissioned into purpose-built studios all around the world, including the brand new Abbey Road facility in London (at that time, the largest recording studio in the world), and so the stage was set for the beginning of the musical recording era as we know it.

Over the next 20 years the design of better sounding and higher fidelity microphones and amplifiers was to follow, with new researchers like EMI Hayes Laboratories setting many benchmarks, and manufacturers like Neumann producing classic equipment that is still hallowed today. Still, direct to disc recording remained the standard until 1948 when Ampex, 3M and Bing Crosby’s money finally developed magnetic recording tape into a viable alternative.

NO GAIN NO PLAY’N

The principal and most tangible requirement of a microphone amplifier is to apply gain in the audio band of frequencies; lots of gain, and useful amounts tend to range between 40 and 70dB. In voltage terms, that is an increase in amplitude of up to 3000 times, which is quite a lot.

The oldest systems accomplished this amplification using a combination of input transformer to class A valve circuitry, and that was the way it stayed until the invention of transistors. The transformers were wound in such a way that a small amount of gain was produced across the windings, and then output to the valve amp which would provide whatever additional gain was required. Various other schemes have been used over the years, and I have listed some common ones below, with examples of devices that have made use of those topologies.

• Transformer to class A valve (Bill Putnam’s original UA100D and UA610 control panels)
• Transformer to class A transistor (Neve 1073 preamp)
• Transformer to op amp (Chilton CM series Broadcast desks)
• Transformerless differential transistor input (Soundcraft 400B desk)
• Transformerless instrumentation amp IC (Klark Teknik Midas XL200 desks)

Interesting, at least to me, is that right from the get-go all of the above methods employed a ‘differential input’ as a way of suppressing unwanted external noise — and noise is the second most important matter a microphone amplifier must face.

TAKE A LONG LINE…

Because a microphone cable is made from a long bit of wire it works pretty well as an antenna, and will receive all manner of emissions from radio broadcasts, digital TV, light switches sparking and the guy next door operating his hand drill. Then on top of all that, you have your mains AC voltage humming along at 50 or 60Hz (depending where you live). If all that junk infects our mic amp and gets amplified 3000 times as well, then we may as well call off the session and spend the night drinking. Differential inputs are our main defence against these troublesome unwanted signals, and a typical audio transformer can attenuate them by 100dB or more.

In addition to our external woes, the components themselves produce noise, so every system has it, and you can hear it as hiss when you turn up the volume. For frequencies under 10Hz, semiconductors will produce a kind of crackle, like the slow grinding of metal on stone, as random charge carriers within the silicon combine or break apart. In a mic amp where the designers haven’t troubled themselves about noise, there can easily be something lurking in the background with an amplitude well into the hundreds of microvolts. We call that the ‘noise floor’ and what we want is for our audio signal to be as far above it as possible. That is one of the reasons why most of the downstream gear you are likely to use is designed to operate with signals of around one volt or so (professional line level signals of +4dBu correspond to an amplitude of 1.228 volts). Processing our wanted audio at line levels in the region of 80dB above the noise floor means it is far easier to keep it clear of all the unwanted crap, internal and external. [2] A superior mic amp will have a very low noise floor, but in our inferior system a delicate ribbon microphone might only be in the clear by 20dB or less, and when we turn it up will sound like our artist is playing a tune in the back yard, while watering the garden.

This helps explain why you should avoid plugging a line level output into mic level input. Every time you apply gain to a signal, you are also amplifying noise. [3] If you have audio that is a happy distance from the noise floor, then it is not a good idea to turn it down, just so you can amplify it up again and bring the noise floor with it. It’s much wiser to connect line outputs to line inputs at the same level (see diagram 3).

We no longer live in the era when it was good enough simply to achieve a result, and now when designers set out to develop a microphone amplifier that performs better than the average, noise management is a major subject that needs to be considered from the very first sketches. There is a subtle trade-off between noise and gain, and a more palpable one between noise and bandwidth. It can sometimes play out that several low-gain stages cascaded or added together might produce the same overall amplification with a lower noise figure than one single high-gain stage. But will that muck up the input impedance seen by the microphone? Or make our external noise rejection less effective? And how much frequency bandwidth do we want? Do we restrict it in order to lower noise, especially under 10Hz, or will that upset our many customers doing whale music? Or should we give them a switch that makes our amplifier operate down to DC?

So finally, here we are at the big table where one argument starts to affect another, and hopefully the designer will find that process satisfying, and feel energised as they look deep into all the challenges for a unifying solution. Failing that, they might look deep into a beer.

In the second part of this article, I will explain how differential inputs go about their important work of attenuating noise, introduce several other mic amp essentials, and talk about the design process, and the pursuit of what some people refer to as the ghost in the machine: that elusive quality of sounding ‘good’.

It also helps that Jónsi has a good set of pipes on him, despite mostly delivering a sweet falsetto. “The music is really dynamic and he flows with it,” explained Jónsson. “On the loud parts, he is really loud. Some parts he sings really low, but there’s usually not much going on at the same time, so I don’t have to really raise him.”

The current show has about 45 inputs coming off stage, with a few extra channels for playback click tracks at the monitor position. The console package was all Midas, and the crew use two 24-channel Midas DL431 stage box splitters to have separate gain stages for FOH and monitors. “I’m a Midas guy,” said Jónsson. “I like the sound of Midas. How they make the preamps is better than other brands.

“The show is really dynamic. I would say the softest part would probably be about 85dB, and the loudest part could be peaking around 103-105dB. Some songs might have a really soft piano and it’s not tasteful to try and make that loud! It would sound silly.

“I’m not pushing the low parts up, I just let them be low and then when they play harder you really get the impact of more level and more expression. In general it’s not really a loud show, but in parts it’s rock ’n’ roll level loud.”

Jónsson used to run a Waves package, but abandoned it for the internal Midas processing and a couple of TC Electronics M3000 outboard reverbs; one for vocals and the other for drums. Likewise, his scene recalls are mostly there to get him in the ballpark and change reverb presets via MIDI. “It’s programmed for each song, but it’s mostly reverb changes, delay times, and fader positions for the beginnings to give me a fair start,” he said. “I’m not recalling mid-song and we’re not running time-codes.”

While he has compression on every channel, they’re often not very active. He also mostly just high passes and adds little bits of EQ. Likewise, the drum gates are only set to trigger at the level of the stage noise, everything else gets through. “They’re happy working on how they want to sound and I think it’s not my position to change that,” said Jónsson. “I’d rather try to capture the sound and feeling they’re creating and expand it to others without messing with it too much. Not adding my personal taste and flavour into it.” Really, said Jónsson, the Sigur Rós show is all about “reverb, reverb, reverb and then more reverb.”

SCREEN OFFS KILLSWITCH

The generously recessed rear control panel of the K10 has been made shallower and smaller as the higher power amp and new low-noise fan have seemingly pushed their way to the surface. The control panel has changed to reflect the times. Retained are the two combo XLR/jack inputs with XLR links and a single XLR mix out. Gone are the switches. The one I miss the least is the sudden-death mic/line switch on Input 1. The preset EQ options on two switches are handy on the K10, and I can find them in the dark, but gone, replaced by a screen and nudge buttons — I’ll get my glasses. To be fair, anything that makes the mic/line switch harder to accidentally engage is good and QSC recognises this by making you confirm your choice before enabling the switch to mic input mode.

The screen also significantly increases the number of controls and options available. The original K10 had a few preset options on a couple of switches, the K10.2 has 11 factory presets. These tweak the speaker’s frequency response and dynamics processing to suit different physical environments or applications. There are settings for live sound, EDM, speech and monitors, both stage and studio. There are settings for using the speaker as a bass or acoustic instrument amp, and settings for handheld or wireless mics.

The delay function is a great inclusion and works for me, I’m often running delay speakers from an analogue desk so it saves gear. Four-band EQ is available but is sensibly limited to cut only, and only up to 6dB at that. These speakers shouldn’t need EQ beyond the broad presets anyway and this minimises the effects of any crazy or accidental settings. Scenes can be saved to store user settings if multiple users are involved.

The K10.2 is easy to use and navigating the screen is intuitive enough. Input 1 switches between line and mic level, via the screen. Input 2 switches between line and Hi-Z, to become a handy high impedance input for instruments with pickups. Input 3 has changed from a pair of RCAs to a stereo 3.5mm minijack. Not sure about this, its sort of more modern but the common 3.5mm to RCA cables are readily available compared to 3.5mm male to 3.5mm male cables, and the reliability of minijack sockets worries me. Commendably there’s no Bluetooth, that really encourages low quality sound. There’s no 48V phantom power on the mic channel either. 48V on the line is nasty when the cable inevitably gets accidentally unplugged while its turned on but condenser vocal mics are common now… there’s a problem looking for a solution.

BETTER, NOT BIGGER

The technology that gave the QSC K Series its superior quality sound is still there. QSC’s Directivity Matched Transition (DMT) synchronises the HF and LF driver’s coverage angle. Intrinsic Correction keeps the speakers tidy in demanding environments. With new drivers and more power I was interested to see how much difference there was in use, and yes, its new and improved. In an A/B comparison with the K10 the K10.2 produces the same type of warm, clear sound but it’s noticeably bigger. The low-end is fuller, the -6dB LF response is quoted as having improved from 60Hz to 56Hz and this extra depth is particularly noticeable on music playback. It’s louder overall and more resistant to feedback.

The presets add another dimension, particularly if there’s no mixer or drive processing in the system or installation. Dance music on the dance setting sounds full and thumping. For live music shows with mixers the default setting is a flat start for FOH where EQ and compression will come from the desk or drive rack. For foldback, the stage monitor settings use sharp filtering to reduce un-wanted low-end buildup. Setting 1 reduces the boominess but retains strong lows for instruments. Setting 2 cuts more to focus on the vocals. In use I preferred the new, steeper cabinet monitor angle, it kept the performer closer to the speaker so the sound was more direct and used less space on stage. It can also be hung from M10 installation points and M5 yoke attachment points or pole-mounted, horizontally or angled down by 7.5 degrees. The K10.2 is a genuine multi-purpose speaker you could take anywhere. It looks neater and sounds better than the K10. Its more powerful and more flexible in use. Its reputation will be maintained.