Review: Christopher Steller

HARMONICS SECTION

The Harmonics section, with its Shimmer and Glimmer controls, offers pitch shifting (Shimmer) and harmonic modulation (Glimmer) to further enhance movement of the reverb environment and create more ghostly tones. Shimmer offers up to an octave above and below the original input (set in intervals with a button hold and knob turn), and has a Regen button, which gives more of a feedback build in the wet signal; while using Glimmer’s High Band button emphasises the upper frequencies to the point of harshness.

The results of these parameter changes vary based on the Texture type selected (the three reverb algorithms as described above). If ‘Sparse’ is selected, a more alternating left/right movement is created, which is great for a big, stereo field; while ‘Dense’ is a more traditional tail; and ‘Diffuse’ has a slow build up. All of these types benefit from some experimenting with the Decay control, especially when you are chasing huge, ambient atmospheres ie. maximum output for minimum effort.

FILTER SECTION

The Filter can be switched between a simpler low/high damp and a 24dB/octave LPF, with variable resonance for the effected signal only. This is a surprisingly useful tonal control for the overall output of the signal.

KARPLUS TUNNEL

Finally, the Delay/Karplus section’s Karplus bit: the most unusual feature of this effect module. Karplus-Strong string synthesis is a form of physical modelling that simulates hammered or plucked strings, and also certain simple percussion instruments, developed by the eponymous Messers Karplus and Strong, in an American university. With a button press and knob turn in the delay section, Karplus/Strong mode is activated. The Tap/Trig button will give you a short or long note, depending on hold duration, and the Size/Pitch and In Gate inputs become CV and Gate ins for using an external sequencer or keyboard.

Starlab can also store four presets, referred to as Favourites, which can be selected from the front panel or externally.

STAR GAZING

I whiled away the hours with Starlab being fed different left and right waveforms by a Pittsburgh Primary Oscillator (the fragments and timbre waveforms), triggered by a simple four-step sequence, with two extra LFOs from another module for some slow parameter movement. It was something of a religious experience. The first external LFO controlling the speed of the internal LFO, and the second controlling Shimmer depth… quite heavenly! Next, I’ll break out the MPU-101 MIDI/CV converter for some serious parameter control with Logic Pro X, for the proverbial ‘swine wallowing in effluent’ experience.

MODULAR VISION

Visiting the Modular Grid website for an hour or two really brings home the size and scope of the modular market, offering a bewildering array of products for all genres and budgets, but Strymon is definitely responsible for the more popular effects available. Just watch a few synth videos on YouTube and a healthy number of these will be utilising a Big Sky reverb pedal to create amazing soundscapes. Starlab? Just imagine a Big Sky or Night Sky with 18 control inputs. Hell, yeah!

Before going any further, please note that these are recommended levels for capturing individual channels or tracks that are ultimately going to be processed and mixed together, and they apply primarily to popular music genres that have relatively small dynamic range requirements. Not only will these level recommendations bring your digital audio signals into a better operating zone, they’ll streamline the mixing process and will be greatly appreciated if you’re capturing or preparing something for someone else to mix – as we’ll see in the next installment.

[For capturing music with greater dynamic range expectations (orchestral, chamber, acoustic jazz, etc.) you might want to lower the centre of the target zone to -24dBFS and broaden the optimum area to extend from -12dBFS to -36dBFS – especially if capturing live performances. We’ll look at these scenarios in the ‘Gain Bending’ installment.]

Target Zone

The illustration above shows an optimum Target Zone of 16dB between -12dBFS and -28dBFS, which offers 8dB either side of the RP-155 recommended Alignment Level of -20dBFS. The goal is to keep the signal’s Average or RMS level sitting somewhere around -20dBFS while keeping its highest and lowest Peak levels within the Target Zone whenever possible.

If you don’t have access to Average or RMS metering, aim to keep the note-to-note Peak levels sitting within the Target Zone while also remaining within the space above -20dBFS – which should help to keep the Average level useful. Let’s say you had a signal with 8dB difference between the Peak levels of its loudest and softest notes. It is better for that 8dB difference to exist between -12dBFS and -20dBFS than between -20dBFS and -28dBFS, because this has a more likely chance of keeping the Average level closer to -20dBFS (although that depends largely on the signal’s Crest Factor, as explained in the previous installment).

In the following illustrations the blue section represents the difference between the signal’s highest Peak level and its lowest Peak level.

Safety Margin

The area above -12dBFS is a Safety Margin for occasional and unexpectedly high peaks, and is roughly equivalent to the point above the analogue NOL where even the most generously calibrated analogue tape recorders would’ve entered saturation. It’s okay for the signal’s Peak level to go into this area from time to time. Percussive sounds with hard attacks, such as wood blocks and ‘80s kick drums with click pads, will regularly enter this area. Aim to keep their Peak levels below -6dBFS and make sure they stay clear of 0dBFS.

Problematic

Signals with Peak levels that sit mostly in the Problematic area but sometimes venture into the Target Zone need more gain, either in capture or downstream (clip gain, a gain plug-in, etc.) to make them audible against sounds that sit comfortably within the Target Zone.

Signals that sit mostly in the Problematic area but occasionally venture beyond the Target Zone into the Safety Margin will almost certainly require dynamic range control (compression/limiting) and/or automation to keep them audible alongside signals that sit comfortably in the Target Zone.

Application

These level recommendations should be simple to implement with any DAW or digital mixing console. Situations where they’re not easy to implement indicate that the signal might require dynamic processing (compression, limiting, automation) and/or filtering to sit comfortably alongside other signals in the mix. In these situations the best approach is to use a lower gain to ensure the signal does not clip but still delivers a useful signal for further processing and mixing. As Rupert Neve advised many years ago, “a good engineer will not let his signal clip more than once a year”.

AN ATTENUATED HISTORY OF GAIN

Throughout this series references have been made to choosing a microphone with appropriate Sensitivity to place its output level somewhere within the preamplifier’s ‘Goldilocks Zone’ – an area around the gain dial where the gain has not been turned up so high that it’s introducing excessive noise when using passive microphones, but is not turned down so low that even at minimum gain there’s a risk of overloading the circuitry and clipping the signal when using active microphones. If we know the sound’s SPL, the Target Level and the upper and lower limits of the Goldilocks Zone, we should be able to choose the right microphone Sensitivity for the job. To determine some valid upper and lower limits of the Goldilocks Zone we have to start with a quick look at some typical preamplifier designs, at least conceptually…

Preamplification Goals

The traditional microphone preamplifier requires a low noise input circuit with a relatively high impedance (typically 1500 ohms or more) that’s capable of taking the very small signal from the microphone and amplifying it up to Line Level, a job that requires considerable amounts of gain – possibly +70dB (x3162) or more for a vintage ribbon, depending on the SPL of the sound source. Making a low noise preamplifier circuit with a high input impedance input and +70dB of variable gain is difficult to do while maintaining stability, and even harder when the entire circuit has to be small enough to fit into the top of a channel strip on a mixing console. Amplifiers like that prefer to be oscillators…

In 1840 the American humorist Seba Smith (1792-1868) famously wrote “There are more ways than one to skin a cat”, a phrase that’s currently outlived him by over 150 years. The same applies to microphone preamplifiers: there are more ways than one to solve the low noise/high gain/stability problem when amplifying a microphone signal, and nobody really cares how you do it as long as it works and sounds good. Here are some examples…

Transformer

One solution is to share the total gain requirement over two lower gain amplifying circuits, or ‘stages’, as shown below.

This example uses a transformer as the first gain stage. The transformer is an excellent choice for this application because it provides a differential balanced input with voltage gain and a high Common Mode Rejection Ratio (CMRR) – which is exactly what we need to take the signal from a microphone’s balanced output via a microphone cable. Typically, transformers used for this application are ‘step-up’ transformers with turns ratios between 1:5 and 1:10, which translates to voltage gains from +14dB (x5) to +20dB (x10). The transformer in the illustration above has a turns ratio of 1:10, which gives it a voltage gain of +20dB. It’s followed by a second stage with variable gain from 0dB (x1) to +50dB (x316.2), making the total gain range +20dB to +70dB (x10 to x3162).

One problem with this design approach is that the voltage gain of the transformer determines the minimum gain possible, which can be too high for some situations. Also, in some smaller and/or lower cost transformer designs a high peak level from the microphone can cause transformer saturation – a form of magnetic clipping that’s akin to tape saturation and, similarly, treads a fine line between euphonic and useless.

With or without saturation, a high peak level going into a step-up transformer means an even higher peak level coming out of it that can overload the variable gain stage and/or other circuits further downstream, resulting in clipping distortion. The signal coming out of the preamplifier is too high even with the gain turned all the way down to minimum, which, in this example, is the transformer’s voltage gain of +20dB. This problem is solved by adding a switchable -20dB (÷10) pad to the input, as shown below:

The pad could be inserted before or after the transformer; for the purposes of the illustration we’ll put it before. The result is a preamplifier with variable gain from 0dB to +70dB (0dB to +50dB with the pad, and +20dB to +70dB without the pad). Add phantom power and a high pass filter and you’ve got a versatile ‘take on anything’ microphone preamplifier similar to those seen in professional mixing consoles ever since Neumann invented phantom power in the mid ‘60s.

Transformerless

In the late ‘70s there was a move away from transformers. This saved considerable space on each channel strip (allowing more channels in the same space), while reducing the cost per channel, improving some specifications and worsening others – although, of course, the marketeers focused on the improvements only. The early ‘transformerless’ preamplifier designs essentially replaced the transformer with an electronic circuit, as shown below:

Numerous circuits were designed to emulate and replace the transformer in this application, or even to create an entirely new microphone preamplifier design using transistors throughout rather than transformers. These were often packaged as small circuit-board-in-a-box designs containing discrete components potted in resin with terminals ready to mount directly onto the circuit board of a mixing console’s channel strip, rather like a giant chip. Some were ultimately built into chips. These included products from Analog Devices, John Hardy Company, Benchmark Media, Harrison, Valley People and Philips.

Padless

With the advent of the home studio and the project studio came the need for a preamplifier that was cheaper to make and simpler to use. It didn’t need the 0dB to +70dB gain range of the professional consoles because the intended market was recording musicians using dynamic and condenser microphones to close-mic drums, guitars and vocals at home. [Royer had not yet resurrected the ribbon microphone as a viable option, so there was no need for high gain/low noise preamplifiers.] This lead to the ‘padless preamplifier’, which is now commonly found on most affordable mixing consoles and interfaces.

The illustration below shows a padless version of the preamplifier shown earlier. The gain range is variable from +10dB to +60dB, which is sufficient for most close-miking applications using condenser and dynamic microphones when recording popular music genres. As an added benefit there is no pad switch for the non-technical user to misunderstand and misuse, which solved an on-going problem for the after-sales support people: no more angry calls from technically-illiterate users saying “the preamplifier is too noisy” or “the preamplifier doesn’t have enough gain” when the real problem was that the pad was switched on when it was not needed, and no more angry calls saying “the preamplifier keeps distorting” when the real problem was that the pad was not switched on when it was needed.

Using the pad switch when it is not needed means unnecessarily more preamplifier gain is required, which can result in unnecessarily noisier signals. Conversely, not using the pad switch when it is needed allows signal peaks to go higher than necessary, increasing the risk of clipping distortion. The padless preamplifier’s ‘happy medium’ gain range of +10dB to +60dB simplifies the circuit and reduces costs, but with the trade-off of lower versatility.

If the minimum gain of +10dB is too much you’re going to need a pad; either one that’s built in to the microphone, or an external pad like the simple circuit shown at the end of this installment…

Discrete & Class A

When a preamplifier or other audio circuit is described as ‘discrete’ it means it is made from individual electronic components (resistors, capacitors, transistors, etc.) rather than integrated circuits aka ‘chips’. In a discrete circuit the designer can select and hand-match electronic components to minimise noise, and can also use higher voltage power supplies than chips allow, which, in turn, translates to greater Headroom (among other things).

You will often see the term ‘discrete’ used alongside the term ‘Class A’, referring to a circuit design that amplifies the entire waveform with one amplifying stage – as opposed to splitting the signal into positive and negative half-cycles that are amplified individually and then joined back together (Class B), which can result in crossover distortion due to tiny level mismatches between the half-cycles when they are re-connected. Class A is often considered the most pure form of amplification, but it is also the least efficient. Efficiency is important when working with low-powered devices (e.g. bus-powered interfaces) and with amplifier chips where low efficiency means more heat and therefore more thermal noise and increased risk of component failure. Most of the chips used for amplification use ‘Class AB’, which uses Class A when signal levels are small and crossover distortion will be significant, but switches to the more efficient Class B when signal levels are high and crossover distortion will be insignificant.

Preamp-On-A-Chip

Contemporary preamplifier designs – especially those found in interfaces – often use the ‘preamp-on-a-chip’ designs from manufacturers such as Burr-Brown, THAT Corporation, Analog Devices and others. These put all of the important components of a microphone preamplifier circuit into a single chip and allow the manufacturer to customise some performance characteristics with the use of external electronic components. You will often see the same preamp-on-a-chip inside a budget interface and an expensive interface, but they’ll offer different levels of performance due to how the manufacturer has configured the external circuitry and the quality of the AD converter they’re following it with.

In products where the preamp-on-a-chip’s output goes directly into an AD converter within the same device (e.g. an interface or a digital mixing console), some interesting design possibilities arise. Because the output of the preamplifier is going directly into the AD converter and does not need to be made available to the user as an analogue signal at that point, there is no need for the preamplifier or the AD converter to conform to the +4dBu industry standard NOL. This means the preamplifier and AD converter can be designed to run with lower power supply voltages (e.g. bus powering) and lower NOLs, and the preamplifier chip only needs enough gain to bring the microphone signal up to the required level to drive the AD converter chip. It also means the total gain can be shared between the preamplifier and the AD converter; in this situation the preamplifier’s maximum gain only needs to be enough to get the signal sufficiently above the AD converter’s noise floor. The rest of the gain is provided by DSP, raising the level of the preamplified signal along with any noise it contains and any noise present in the AD converter’s analogue input circuit.

The illustration above shows the EIN curves for Burr-Brown’s long-established PGA2500 (red) as used in interfaces from Apogee and Antelope to RME and Universal Audio, and THAT Corp’s very recent 6261 (green). The PGA2500 has digitally-controlled gain that’s variable in 1dB steps from +10dB to +60dB, giving it a range of 50dB. The 6261 also offers digitally-controlled gain in 1dB steps from -8dB (attenuation) to +34dB, giving it a range of 42dB.

The illustration below is from the data sheet for THAT Corporation’s 6261 preamp-on-a-chip; note that the digitally-controlled gain is shared between the preamplifier and the ADC driver (a circuit designed to provide a signal to the AD converter). Conceptually, it can be viewed as a modern-day return to the two-gain-stage approach except now both stages have variable gain. Also note that the output of the ADC driver is rated at 2V_RMS for Full Scale (0dBFS), which puts the -20dBFS Alignment Level at 0.2V_RMS rather than the 1.228V_RMS required to meet the industry-standard Alignment Level of-20dBFS (which is aligned to +4dBu). It’s 15.8dB lower than +4dBu, and therefore requires 15.8dB less gain to reach its NOL. With that in mind, the maximum gain of +34dB does not seem so low. At the same time, having a minimum gain of -8dB means it can handle a higher input level without the need for a pad.

While the preamp-on-a-chip designs are affordable and offer numerous advantages, they can’t match the performance that’s possible with discrete designs. It’s also worth noting that many of the preamp-on-a-chip designs (such as the PGA2500) offer a minimum gain of around +10dB, similar to the padless preamplifiers described earlier. If the minimum gain of +10dB is too much, see the pad circuit below.

Gain-Ranging

Some field recorders (e.g. Zoom, Sound Devices) offer the ability to record in ‘32-bit float’ format, which provides huge dynamic range – far more than a microphone could ever deliver. To achieve this they use a technique known as ‘gain-ranging’. Instead of using one preamplifier and one AD converter on each microphone input, they use two. One preamplifier/converter combo has a fixed gain that’s optimised for capturing and converting low signal levels, the other has a fixed gain that’s optimised for capturing and converting high signal levels. The outputs of both converters overlap to ensure a smooth transition between them, and the rest is done in DSP to provide a 32-bit floating point signal that is essentially impossible for any contemporary microphone to drive beyond 0dBFS. It’s possible to overdrive the analogue inputs of the preamplifiers if you try hard enough, but that’s unlikely from a microphone output.

Gain-ranging isn’t new — you’ll see it in the AES42 digital microphones from over a decade ago — but it’s making its way into affordable audio devices and de-emphasising the need to focus on getting the gain right from the start. Speaking of which…

TOWARDS GOLDILOCKS

How does all of this information help us to define the Goldilocks Zone? It’s all about choosing a microphone with the right Sensitivity for the sound source and preamplifier so that we can reach the Target Level given in the recommendations above (-20dBFS Average) without introducing more noise than necessary when using low Sensitivity passive microphones, and without risking distortion when using high Sensitivity active microphones.

Next installment we’ll delve into the Goldilocks Zone, define upper and lower limits for gain, and include some practical examples…

Photos: Robert Hambling

PICKING UP 20 YEARS ON

“I love working with The Oils,” affirms the English-born Livesey, on working with the band again. “It is amazing, even after such a long break, how quickly it felt like there had been no time at all since we last worked together. I guess the fundamental relationships don’t change much so it was very comfortable and creatively exciting. They have always been one of my favourite bands to work with and these recordings were no different. But the one thing I noticed is that every member of the band had become even better as musicians, singers and writers. And that’s bearing in mind they were already amazing before.

“We had about 21 songs and we recorded 19 of them and kind of half recorded the last two,” continues Moginie. “In six weeks of working nine-to-five, Monday to Friday, we barrelled through the songs. Then we went back to Oceanic for the final two weeks to lay down some piano and Hammond C3, add some strings, horns, vocals from Pete, Jess Mauboy, Bunna Lawrie, Adam Goodes, Stan Grant and Tasman Keith and some extra guitar bits that were needed at the last minute. Then we transferred all the drums we had done at Rancom St to my Studer A80 16-track two-inch tape machine and then flew them back into the Pro Tools sessions for some extra juice and thump.”

MICROPHONE OPTIONS

When it came to the recording sessions, Livesey set up multiple microphones around the studio environs to capture the best possible sounds. “When we were recording with Warne, he would position drum mics everywhere and anywhere in the room,” reveals Moginie. “And when it came to the mix he’d say, ‘we need a bit more of this on the kick and a bit more space’ or ‘we need a bit more room on the crash’ to which he’d add some sort of room mic that was compressed to the max but had incredible clarity, so the close sounds themselves were very clean with the drums. Then he would dial in the room mics which then became another thing again. Different microphones can totally alter the acoustic space of anything it occupies. The guitars went through the MCI JH console, or some APIs, which was kind of grainy and very good for guitars, and the bass went through a Neve 1073. The lead vocals we used my Neumann U67 through a Neve 1084 and a Distressor.”

“The vocals were generally recorded with Jim’s Neumann U67,” confirms Livesey. “We had a few on the session but Jim’s was the best one for Pete’s vocals. But there were a couple of songs where we wanted a sharper, more aggressive tone where we used a Shure SM7.

ACOUSTIC GUITARS

“For acoustic guitars I used a combination of a Neumann U67 and a Neumann CMV563. Often we recorded Martin and Jim playing together with those mics on each guitar. And on songs where we only used a single acoustic guitar I used a mid-side pair of Neumann U67s [one set to cardioid pattern facing the instrument and a second mic on figure-8 pattern aligned with the first mic but at 90 degrees to the first mic. That second mic is then routed to two channels panned left and right with one of those channels phase reversed].”

“The bass cab was miked with a Neumann tube U47 and there was also a DI. In mixing I often used that DI’ed into a Sans Amp or sometimes I re-amped the bass for more grit.

”Obviously, the Hammond organ and piano were miked, with close mics and stereo room mics in the case of the piano. And we did amp up the Moog as well as printing a DI. Other synths were DI’ed and there’s a bit of soft synths and samples here and there too. The Mellotron was also DI’ed.”

DRUMS: IT’S COMPLICATED

“Drums is where it got complicated,” continues Livesey. “We went through a bunch of different mic positions and choices throughout the bed tracks. But generally, I’d use a Beyer M201 on the snare top. AKG D12 and a sub kick (NS10) on the resonator head and a Shure Beta 52 inside the kick where possible or on the beater side of the batter head where not. Toms were miked with Neumann U87s. Overheads were Telefunken/Shoeps M221Bs. A Shure SM7 on the hats and KM84 on the ride. I had many room mics set up and used different combinations on different songs. But amongst those were my original STC (BBC) 4038 just a couple of feet in front of the kit, a PZM on the floor behind the kit, two AKG 414s as stereo rooms low down and about three metres in front of the kit, and up high a mono Neumann U67 set to omni. I think I also used a Shure SM57 sometimes as well”.

DON’T FIX IT IN THE MIX

Livesey also adheres to the philosophy of getting as close to the final sound as possible when tracking. “I don’t like the ‘printing flat’ idea because I think you need to be working to something close to the final tones whilst you are tracking overdubs,” he says. “You need to shape the sounds of the overdubs to the track. There would be some EQ on almost everything and compression on things like drum room mics, bass, some guitar parts and vocals. I generally don’t print very extreme compression although some of the more edgy vocals were hit pretty hard. But there is always extra processing that happens in mixing as you fine tune the tones.”

Once the recording and mixing were completed, plans were put into motion for a prospective release date. All were in favour initially of releasing the entire batch of songs recorded, but when the pandemic hit, a rethink was considered. “At first we were going to put the whole lot out in the early days of the pandemic,” recalls Moginie, “but just like everybody else, we decided that if you put an album out, then you should be touring behind it, which we couldn’t do. It was frustrating but in the end we decided to wait a year and get all the pieces in place and to do it the right way. In the meantime, we decided to put the ‘The Makarrata Project’ out first which had ‘First Nation’ and ‘Gadigal Land’ and those sort of songs on it with a First Nation theme. And we left the rest of the other stuff intact for the next album, ‘Resist’.

“It was very much all about the subject matter of the songs which was why ‘The Makarrata Project’ was all about First Nation issues, changing the date, and the ‘come on down sit around the camp fire and talk as one people’ idea, and included collaborations from Dan Sultan, Leah Flanagan, Kaleena Briggs, Ursula Yovich, Alice Skye, Troy Cassar Daley, Frank Yamma as well as the ones previously mentioned. We had a posthumous recording of Gurrumul which we laid over ‘Change The Date’ that fitted like a glove and his magic took that song to another level. Bones went back to the USA after the recording and died of cancer late in 2020 just after ‘Makaratta’ got to No.1. It’s still hard to believe such a beautiful man, singer and bass player with us over 30 years has gone. At least he’s all over ‘Makaratta’ and ‘Resist’ and played magnificently.”

AT: Tell us about the new studio’s design.

Tony Buchen: For starters, it’s a slightly coffin-shaped room, to deal with some standing waves. It’s also the biggest control room that I’ve ever had. I mean, the couch is actually off the back wall, which is a luxury. And it fits all my gear.

It’s a mix room with my SSL console, but I wanted it to be more versatile than that. I didn’t want it to be a dead acoustic space. If I’m recording an acoustic guitar I wanted the environment to be conducive to that, or a vocal performance, even. 

James Webeck’s instinct is to start live and it’s easier to shut the acoustic down if you need to. But start with some life. That said, the approach was to start with it as dead as possible near the console. That way, when I’m in the sweetspot I’m just hearing the studio monitors. And then as you go to the back of the room, it opens up a little bit. We have some nice diffusers at the back of the room, with 1960s perforated tiling that I brought back from L.A. I’m really happy with the result.

ORIGIN: THE SPECIES

AT: The SSL Origin purchase must have been a big decision. What’s the rationale?

Tony Buchen: It all stems from how I learnt to mix. I learnt with an engineer called James Cadsky,  a name familiar to many of your readers, I’m sure. James had a studio in Sydney called The Vault, which had an SSL E series — a classic ’80s mixing console that you could really drive — as a result it  sounded more vintage than subsequent SSL releases. I learnt on that console sitting alongside James.

When I moved to Studios 301, I was mixing on the SSL ‘K’ Series there, which had some very powerful features and automation but it didn’t sound as crunchy — it’s not a console you can drive; it’s built to provide pristine sound.

Alongside that I was working in the box more and more, albeit using a Shadow Hills summing mixer.

At that time, I missed not being on a mixing console, and like many others who learnt their craft on a large mixer, I guess I felt like bit of a fraud. I was delivering good mixes and clients were happy with my work but the nagging feeling was: I’m not really doing what I truly know I can do. A mixing console gives you an extra level of control and you’re relinquishing some of that by mixing in the box, for sure.

The point is, I’ve always wanted to get back on a board but I was conflicted because I just couldn’t get my head around buying a mixing console… even ethically, I’m not going to own a console that requires an air conditioner to cool a power supply 24/7. I was resigned to giving up on that dream.

I was at the NAMM show in 2018 when SSL was launching the Origin. I couldn’t resist hopping on. I grabbed some EQ and immediately knew, yes, this was an SSL. And it also has a Drive feature on the channels, which makes it a bit more like the E-series consoles, where you can drive the input a bit more. It also has power saving features, which impressed me. It was ticking boxes. I knew I had to have it… Screw it!

AT: Mixing consoles are enjoying a mini renaissance. Were you quite sure the Origin was the pick of the bunch compared to the equivalent Neve or API?

Tony Buchen: I was at Vintage King in L.A. and it’s gotta be the one place in the world where you actually have the three consoles in front of you, all connected. I tried the API and it’s beautiful. If it’s a competition to be best summing mixer, the API would win every time because it has this depth to it and it’s beautifully made. But in terms of usability, there’s no question for me that the SSL is a way more creative console than both the Neve or the API… at least for the way I like to work. Plus, for me, there’s an instant familiarity with SSL — I immediately feel at home behind an SSL.

AT: How do you incorporate the SSL Origin into your workflow?

Tony Buchen: I’ll do 90 percent of the mix in ProTools. I then put it up on the board (run with most of the SSL large faders at unity) and route some aspects of the mix through speciality analogue outboard, and use some hardware mix bus processing. It’s like using the Origin as a kind of summing mixer with EQ and routing benefits.

AT: Would you insert your hardware processing into the individual channels?

Tony Buchen: More likely I’ll use the small faders to send to parallel compression, for example. Sometimes I have hardware inserts on channels and sometimes I’ll still use software inserts in Pro Tools where appropriate. Having those analogue returns coming up the desk help me to inform my mix decisions more than in the Pro Tools environment. I acknowledge that it’s an esoteric world that I’m talking about but for me it’s more beautiful — those sounds on the console are more present and exist for me in the real world.

AT: And presumably the inserts on your mix bus are pretty consistent from mix to mix.

Tony Buchen: Interestingly, I’ve worked out a pretty unorthodox hybrid approach to the mix bus. Some people might think it’s just a bit nuts, but like I go out and in again through plugins such that when I press the large fader insert on the mix bus, I’m actually inserting plugins as well as analogue outboard. The reason for that? There are plugins that are required on the mix bus that are very hard to replicate in the analogue domain. I have a Manly VariMu, an old peak limiter and other hardware items that are great on the mix bus there are certain ProTools mix bus plugins that are so exciting to me that I can’t live without them.

RECALL SCRAMBLE

AT: One of the most compelling advantages of working in the box is instant mix recall. Does the SSL Origin have any recall features?

Tony Buchen: No, and it’s something I’ve had to give a lot of thought about. For me it’s started with having a great assistant. Ollie Brown has his own studio across the hall and is a real up and coming talent in his own right. Together we’ve been working on a process for recalling a mix. We’ve got it down to about a 10 or 15 minutes turnaround.

AT: What’s the secret beyond taking a bunch of photos of knobs? 

Tony Buchen: It’s about taking photos of knobs [laughs]! Because the console doesn’t have a computer like the SSL K does, where it tells you the position of pots and faders, you need to have your own system. And in the end it comes down to how I use the console, and it being compatible with a super-fast recall. 

This isn’t an original idea but you must be disciplined in making sure your bass parts always comes up the same channel; same with vocals, drums etc. It’s about keeping everything you can locked into a set signal flow. That way the EQ and routing remains largely unchanged and I can make adjustments quickly and creatively rather than spending time at the patchbay. This means, in terms of the recall, I’m just making small adjustments to the analogue gear a lot of the time. The radical changes happen in Pro Tools sessions which are 100% recallable, obviously.

AT: Finally, Tony, what sort of pandemic did you have? Do you fit into the ‘what doesn’t kill you…’ category?

Tony Buchen: It’s undoubtedly been a shocking few years for the creative industries. And I’ve learnt a lot about what makes me tick in the studio when a lot of what you hold dear gets taken away. But I’ve been very fortunate to have made some great records remotely and have never been wanting for work. I think the biggest ‘take home’ for me is to always see setbacks as opportunities. If something gets cancelled in my schedule, I’m thinking, okay, I’ve got free time to create. Sure, it’s a bummer that the gig was cancelled. And, even though you could very easily go down that rabbit hole of thinking the universe has turned on you, it doesn’t actually do you any good to think about things in those terms. So Covid definitely tested that philosophy in ways I never thought anything would. But, ultimately, it’s been a great opportunity to do some interesting shit, notwithstanding the pain and suffering that was going on in the world which has weighed very heavily on all of us.

BUNDLE

Aside from these alterations there’s not a lot separating the Player model from its American predecessor. Both sport GraphTech Tusq nut and deep C-shaped 12-inch radius mahogany neck, with only 1mm differences here and there for neck width. Fretboard material is rosewood for the Player, whereas the American is ebony, so there’s some difference in tone there – nothing much to write home about. Both feature ‘narrow-tall’ frets eking out to 22 frets (should you really need that 22nd), and both models come with Fender staggered height tuners.

Each instrument also arrived bundled into Fender’s Deluxe 1225 gig bag. Some may wonder, especially at this price, why the Acoustasonics don’t include a hard case, but I imagine due the the nature of the Acoustasonics (a take-it-anywhere instrument you’re likely to be traipsing between gigs and your cohort’s studios), a gig bag is the more prudent choice. And as far as gig bags go, the Deluxe 1225 is the business – extreme padding, super comfortable back-straps, plenty of pockets, and, when combined with the Acoustasonic’s 2.3kg weight, the package comes in at a featherweight 4.8kg. Fender also include a Boveda two-way humidity control pack designed to keep your plank of wood between 45 and 55% humidity. A nice touch.

GO PLAY

You don’t have to look far across the web to find enthusiastic references to the Acoustasonic’s versatility and playability, so I’ll not go too far into my impressions, apart from confessing my reluctance to put the thing down. Price-wise there’s an appreciable discount for the Player compared with the U.S. of A. model – in Australia you’re saving a good $1200 or so. Still, the Player Acoustasonic isn’t cheap at a $1700 street price ($2199 retail). You’d collect a Mexican Player electric Telecaster for around a grand if you hunt around. Regardless, if the Acoustasonic is what you’re after – and rest assured it could cover you in countless scenarios (writing, recording, gigging – amped or otherwise), maybe the asking price isn’t so steep. In short: go play one.