FINE ART OF MUSIC

Now 37, despite his father’s influence, Jan wasn’t always convinced his path would take him in a musical direction. At 19, he was actually studying Fine Arts at the VCA where he found himself gravitating towards the jazz musicians because he “thought they were a lot more fun.”

Jan had learnt the craft in his father’s converted garage studio. It centred around Logic with a 32-track system. The interface was connected by SCSI leads, which would fray over time. When it was only good for eight channels, Cezary kicked it Jan’s way.

Playing in bands with the VCA crowd meant he started working with producers, including Andy Baldwin, a popular producer who worked on lots of Australian records as well as for internationals like Björk. “Andy called me once during the school holidays in about 2001 and told me. ‘I’m going up to Byron Bay to record a band. It’s going to be about three months, they’re called The Cat Empire. It’s going to be their debut record.’ We went up onto this old macadamia farm and converted a farmhouse into a studio. We recorded the record, and it went double platinum.”

HOOKED

On the drive home Baldwin let Jan know about an independent record label owner who was looking for an in-house songwriter/producer. It was a place called Marlin Records, owned by a Sri-Lankan born businessman, Marlon Goonawardana. “I worked a lot with Daniel Merriweather, during that time,” said Jan. “He was the same age as me. He later worked a lot with Mark Ronson, and had his own record that went to No.2 in the UK. He and I, and another friend Phrase, who was a rapper, we all just lived in the studio for a couple of years and got completely absorbed with how it all came together.”

HATS OFF

Over the years, Jan has kept his hand in a number of areas. On any given day he can draw on all facets of his production and composition background. There’s the film and television work, and he continues to play live, these days as a keys player for Dan Sultan; though along the way he started a band with Harry, from the Cat Empire, called Jackson Jackson. On the record production side he still works with The Cat Empire, and lately he’s been producing albums for John Butler, as well as an Alabama Shakes-esque soulful record for Karise Eden. Following on from his largely hip hop and pop work at Marlin, directly after our interview, Jan was due to start a session with Illy and songwriter, Cam Bluff. “I’m writing pretty much every day at the moment,” said Jan. “Something my dad really drilled into me is to get into your creative space every day. Even if it’s just for an hour. If you get in there and scribble something, there’ll be a lot of days you don’t create great content, but it means the one little idea or melody you came up with that was good will eventually find a home. Things will come together, but it only happens if you continually make yourself available to it. That’s just the discipline.”

Despite telling Marlon he knew how to work the studio, Jan had no idea when he first started. “It was a Friday and I was supposed to start on Monday,” Jan recalled. “I asked if I could grab the keys over the weekend to check the lay of the land. I spent the next 48 hours with every manual open of every piece of gear, which I had no idea how to use. I didn’t know how to use a patch bay, so I was manually patching things, and calling my mentors, like Andy and Adam Rhodes, who guided me through it. By the time Monday morning came, I had the Akai MPC coming through and recording to Pro Tools, and I had a vocal chain. From there it was a case of ‘fake it until you make it’, and I got away with it.

“I really thank Marlon, who bankrolled this hopeless bunch of kids and gave us the keys to a very expensive, well-kitted out studio on Greville St in Prahran, and told us to go for it.”

After Marlin, Jan bought a studio in Hawthorn East off Barry Palmer from the Hunters & Collectors called Way of the Eagle Studios. As part of the sale he also bought some of the gear, including a TL Audio Tubetracker and Neve preamps. “I was there for seven or eight years in a dark cave,” Jan said. “It was an awesome studio, but there was no natural light. I’d get there in the morning, leave at night, and it would be dark and rainy. I’d feel like I wasn’t part of this world.” He had partnered up with Chris Scallan, who took over the whole studio when Jan left, turning it into more of a post-production/film house.

SCORING IN BUNCHES

Jan doesn’t draw too much of a distinction between his different jobs. “In a way, film and television composition is similar to record production,” he explained. “You’ve really got to listen to what the director’s vision is then apply your aesthetic to that. Rather than slapping your aesthetic on it and saying, ‘It’s gotta be trap beats the whole way!’ You won’t get very far doing that. I’m pretty sure you’ll just get fired.”

Jan explained his process, which usually starts with a script or the first cut: “With something like Picnic at Hanging Rock, there wasn’t that much dialogue, especially in some of the earlier episodes. It’s built around this mood, and a sense of space, and mystery. Without the music or any sound designing it just looks like a bunch of girls wandering around in the bush. Often they give you a direction to steer — like ‘sad, but tense’ — then you’ll trial something. People often need to hear something to realise what they do or don’t like. It’s pretty rare they’ll write the draft off. Usually they’ll use it as a springboard to discuss the dramatic pulse or beats they want to hear.

“When you all work together and get it right in the end, it’s pretty cool. The only thing we were told was to steer away from the original music, because it had this very iconic pan pipe theme. They didn’t want to just rehash that, which was good, because I wasn’t in love with the original music.”

ENGINEERING BLISS

While he got into record production early on, the engineering side of the job never appealed much to Jan, “until I realised engineering wasn’t just a means to an end, but was a complete palette of colour you could inject into your recordings. Then I started blissing out. I realised it was a really creative way to make music and enhance the production.”

These days Red Moon Studios is Jan’s playground. He’s often working by himself, but doesn’t resort to working in the box. It’s one of the reasons he still prefers to use Pro Tools, “because it’s more of a tape machine to me. You have to create a great sound into it”.

While he’s not adverse to digital — using NI’s Session Strings and Embertone’s Intimate Strings among others — he tries to incorporate as much analogue gear and physical playing as possible. “I really love things that have elements of human calamity in them. That’s why I love my studio,” said Jan. “The SSL AWS console flips from being a DAW controller, to being an analogue console, with the flick of a button. That hybrid between analogue and digital is the perfect way for me to work.”

He’s got some great gear around the room, including a collection of guitars that aren’t all his. “It’s the cool thing about owning a studio, it becomes a dumping ground for all your mates’ bits of gear their wives don’t want in their houses,” laughed Jan. “Three of these guitars are Dan Sultan’s, he used to keep a lot of his guitars here because he thought it was safer.”

Some of the tasty bits include a Neumann U87, which he sends into a Neve 1073, and a mono side of the Shadow Hills Compressor for Illy’s vocals. “I used to always use the 1176 or the Retro, but then I started using the Shadow Hills and I can’t go past it,” explained Jan. “Someone can slam into it and it still sounds musical. You don’t lose all your gain. It has a great way of grabbing dynamic vocals that’s really cool.”

DRUMMING UP THE COURAGE

When Jan found that engineering bliss later in life, he immediately started tackling his biggest fear. He’d long felt insecure about recording drums, so would farm them out to another engineer. Coming up through the hip hop beat-making scene at Marlin — where he exclusively focused on soul breaks and drum samples for years — he never quite felt other engineers delivered the colour he was after. These days it’s turned into one of his biggest engineering pursuits.

“Marlin gave me an intense love of drums that weren’t clinical, generic or hi-fi sounding,” said Jan. “It was really only when I started recording them myself that I worked out how to do it with different mic techniques and different rooms. It was the hardest thing to learn, but easily the most fun aspect to get right.

“How I approach it depends on what I’m going for. Sometimes I just really like a mono overhead and often don’t mic the toms. I’ve got this old AKG D1200, it’s kind of a 57-ish mic that’s brighter. I often use it on the snare taped with an AKG C451 condenser. Then just add a kick, a mono room, and mic up the bathroom next door and I’ll just try and get a huge vibe out of that.

“On the flip side, there were a couple of songs on the John Butler record where I was running out of mic stands. We were just trying everything — mics up in the ceiling, multiple overheads, all of that. In the end, the drum sound is so huge, but it’s not hi-fi. We were really wanting that Bonham vibe where it still feels really tough, not clinical.

“Once you learn a few different setups, you can start applying them to the music you’re making. For the Karise Eden soul sound, I used a really minimal setup; mono overhead, no toms, kick, snare — a 57 through a Neve 1073 and Distressor — and some rooms. Then I’ll just really spank it with compression and get a lot of the colour from the booth.

“I usually put a mic behind the drummer’s arse, or behind their head from their perspective. Because their body usually kills a bit of the sharp tops of the cymbals, you can get the snare quite explosive without having too much wash at the top end.

“On previous Cat Empire records I’d done, I like the way the drums sounded, but sometimes too compressed. This time, I wanted to make them breathe, because I knew a lot of the other elements would be quite treated. In that way, it was creating a natural sound, which made way for other things to be more out there. You’ve just got to adapt to what things need.”

AUTOMIX UPS THE FUNCTIONS

The F8n was preceded by the F8. You can read Greg Simmons review online for a comprehensive look at the clocking and sound of the original device. The clocking tested very well then, but Zoom claims to have improved accuracy in the timecode when the F8n is powered off. Working in non-linear audio for most of my career I have seldom used timecode functions, but it will be a welcome upgrade for many users.

In the time since the original landed, Zoom updated the design a little, but most of the work was done inside the box. With a host of new features.

The Automix function is one of those. It’s incredibly useful for podcast producers or anyone using multiple microphones for different sources. With Automix on, the F8n detects signal input from each channel and attenuates unused mic channels. This reduces unwanted noise as well as reducing the chances of feedback. The real benefit is in the time saved during post-production. I could unload all my channels and clean them up for my podcast, but the Automix feature does this at recording time and cuts the time required to get your content out to your audience.

It’s significant because the sound quality gap for hardware is continually decreasing, and users are looking at what other advantages their equipment offers. Professionals in audio production are always burning the candle at both ends, so features that save us time start to look like real game changers.

Other improvements on its predecessor include: a new Fader Mode view, instead of the knobs on the home screen; Digital Boost for the headphone output (up to +24dB); selectable volume curves for headphone output (I really like this one); and a maximum fader level increased from +12db to +24dB. Thankfully, with all these being software updates, they’re now available to F8 users, too.

NOT A LIMITING FACTOR

A big addition to the F8n is a new set of advanced limiters. Zoom claims these provide protection against incoming signals peaking, and it seems to work as advertised.

The ‘look-ahead’ limiters take advantage of the F8n’s digital architecture. By delaying the signal a few milliseconds, it can take the content directly through the mic, assess it, process it and then pass it on through the rest of the signal chain uncorrupted.

Obviously, this cannot compensate for a signal too loud for your mic diaphragm to handle, but beyond that it works well. I would have loved to test it with something like gunfire sounds as the very quick changes and huge output levels would really show the effectiveness of the advanced limiters, but I spent a while whispering and then suddenly shouting into the mics and the final recordings went from very distorted with the limiter off, to strong but clear with the limiter on. This works for me.

CLOCKING THE NOISE

At the other end of the spectrum was my ‘quiet place’ test — an old wind up clock miked with a Sanken CS1e in a dead quiet space. I’m always more interested in how these devices handle the softest of sounds, which the F8n did quite well.

The Zoom F8n has an input gain with a range from +10dB to +75dB which is a broad range. The clock came through nice and clear, and there was still almost no unwanted noise when boosted up to 60dB. Above 65dB the noise was at a level I wouldn’t desire, and the range from 70-75dB was unusable. However, that’s less than 10% at the top end where bad noise was introduced; comparable with devices that are much more expensive than the H8n.

ADDED EXTRAS

This device uses eight AA batteries, and when using alkaline batteries that is going to get pricey pretty fast. Rechargeable batteries are the way to go, and Zoom gives you the option to attach an external battery back via a Hirose connector. Having both AA and an external rechargeable battery maximises my options, especially if I’m on a remote location where a faulty built-in battery would kill the entire job.

The various functions for routing allow for a variety of input setups, like the previously mentioned mono signal split into two channels. It also supports ambisonic input and even converts A-format mic signals to B-format within the device. The F8n also supports a range of recording formats, sample and bit rates up to 24-bit/192k.

When I first started recording, I kept wondering why everything was distorting. Turns out the default routing is post-fader, which was simple to change when I figured it out and returned to my habit of monitoring pre-fader.

You can control the F8n remotely via your mobile phone, which sounded like a brilliant idea until I read it was iOS only. It’s a pity considering the prevalence of the Android operating system. I hope those lucky iOS users are taking advantage of how useful this could be.

Finally, the F8n includes a built-in slate microphone, which I think is a cool idea as it allows you to add notes while recording even when your mics might be at a distance from the recorder. This is another function I would have used over the years if I’d had it.

ZOOM, ZOOM

While some manufacturers desperately flog useless features to sell a sequel product, this update to the F8 is a pleasant antidote. It’s a solid recording device with nice, quiet preamps, and the range of functionality gives it a broad user base. It can comfortably capture live musicians or concerts, record in remote places, cover jobs that need SMPTE for linear sync, or it can simply be a convenient device to use in the studio for foley or voice recording.

Zoom has always crammed as much as it possibly could into its devices. Within its price point — probably even up to twice what you would pay for the F8n — I would be confident selecting this device over other available choices to get the job done.

WARMING IT UP

Over the last couple of months of use, the Warm Audio 273-EQ has been faultless. The internal power supply doesn’t induce any noticeable noise, and makes it simple to rack. All the buttons have a nice deep travel, and all the dual concentric switch pots are from Blore Edwards, a UK manufacturer that makes Neve-specific switches and pots. The switches all felt good, but the pots were inconsistent. Even over a small sample size of six EQ knobs, some had a quality feel with decent resistance, others felt a little elastic, while one was very loose. They were not a match for the original knobs. Likewise, the click of the main switches felt more clingy and the reissue Marconi knobs don’t feel like the real thing. Still, they do a good job at their function, it’s just hard to imagine them holding the same level of quality in 50 years time when they start out inferior.

One cost-cutting measure common to clones is to label the chassis instead of the face of the button. It’s understandable, but is a little problematic on the 273-EQ. There are six buttons for each channel, in two rows of three. For some reason Warm printed the legending for the top row above the buttons, and the rest below the second row. If you stick it in a rack, you have to bend right over below the buttons to read the bottom set of labels. If they’d simply printed the labels above both rows of buttons, it would be far less of an issue.

JUST ONE TRANSFORMER

Insofar as similarities go, the Warm Audio 273-EQ has every single feature of the original.

The only major circuit difference is the inclusion of only one mic input transformer — not separate stages for mic and line, with separate transformers to match. Instead, the signal is simply padded by up to -20dB for line level inputs. It does alter the gain staging and switch labelling a bit. Traditionally, the line level section would travel from -20dB to +10dB. It then has an off switch, so you don’t get a thump when switching between the two sources. Likewise, there’s another off position above +50dB to stop another thump occurring when the third amplifier is switched into the path.

Like most clones, the Warm Audio saves a bit of dosh by ditching the separate line input stage and simply switching in a pad. In this case, a rotary pad that extends past the 0 setting on the switch.

I doubt most people looking at buying the 273 EQ are going to miss the separate stage. Especially considering everything else is there: A full 80dB of gain for mic inputs, high pass filter with four selectable frequencies, three band switchable EQ, and a phase flip.

FREQUENT FIXTURES

There are plenty of features you don’t get on a vintage 80 series 1073 module that you do on the Warm Audio 273-EQ.  On the 1073 EQ, the low band has switchable frequencies of 35, 60, 110 and 220Hz, while the mid band switches between 360 and 700Hz, and 1.6, 3.2, 4.8 and 7.2kHz. The high band is typically set at a fixed frequency of 12kHz. The Warm Audio EQ adds some extra choice up here, allowing you to switch between 10, 12 and 16kHz.

While the original only gave you buttons for phase flip and switching the EQ in and out of the circuit, the Warm Audio version also allows you to instantiate an insert, switch a DI into the circuit and change the turns ratio on the transformer, which adjusts the input impedance. This function is marked as Tone. With the switch out, microphones will see an impedance of 1200Ω, for a more natural open sound. Pressing in Tone changes that impedance to 300Ω, which can sound thicker and punchier with dynamic and ribbon mics. Condenser microphones could care less.

The DI is a great advantage. It makes the 273EQ a really well-rounded channel strip. Especially for those looking to make a leap in quality from the front end of their audio interface.

HOW DOES IT SOUND?

We set up both the Neve 1073s and the Warm 273-EQ in a side-by-side test, and asked talented singer-songwriter Angus Legg to come down and play a tune.

Preamp shootouts are always tough. Unlike converter tests, it’s incredibly hard to conduct them exactly. Split the signal and there’s always a tradeoff; transformers added, loss of level, or even frying an input with phantom power. Rather than trying to make it overly scientific, we like to test preamps by taking a song and layering a bunch of tracks up so the character of the gear starts to shine through.

We did this with Angus — recording his acoustic guitar and vocals, then layering a bunch of harmonies, shaker, and trying to play around the out-of-tune octaves of an upright piano to get a stereo track.

We had both channels of each preamp ready to go into the line inputs of an Antelope Audio interface. After each take, we’d simply swap the mic leads over and go again. The gain matching between both was pretty spot on with the Warm at full fader.

COMPARE THE PAIRS

With everything tracked, Preshan did a rough mix and copied the processing between the tracks. He then did the rounds with a couple of unnamed stereo tracks to see what we all liked better. Unlike a high-end converter comparison, where the devil is in the infinitesimal, largely indistinguishable detail. The differences were darn obvious. It simply came down to a matter of, which would I choose?

On first listen, one was very forward. It had a lot more presence to it, and just generally sounded more punchy. The other undoubtedly felt more glued together, but the acoustic had a boominess I felt was distracting

You can probably tell where this is going. After a couple of listens I went with the first one, which turned out to be the Warm Audio 273-EQ. Quelle horreur!

While I couldn’t get the acoustic boom out of my head, others were far more entranced by the glue of the Neve and picked that.

Comparisons are not without perils, and mixing one side first was probably an unfair play. There’s just no way to mix one and translate it to the other. EQ to compensate for a lack of body in one, would only accentuate the boominess in another recording.

Listening back to the original recordings, that acoustic boominess in the Neve mix translated to body that just wasn’t there in the Warm Audio tracks. In particular, when Angus occasionally plucked the lowest string on his guitar, it resonated satisfyingly deep and long in the Neve recording. On the Warm Audio recording, it didn’t have that resounding low end. The resonance of the low string matched the length of other strings, giving it a more compressed, controlled sound. In a mix, that can really help, but it also paints you into a bit more of a corner.

On the vocal side. It was hard to fault the solo vocal through the 273-EQ, which had more presence and air than the vintage Neve. It sounded pre-polished. However, when pushed, that presence could get a bit much and started to tax a little after layering three backing vocals. The Neve sounded more laid back on the solo vocal, and never felt overbearing when stacked.

RING TO IT

This is where the clone wars get interesting. Unlike the vintage Neve, the 273-EQ just sounds more modern. It was as if the harmonics were more obvious in the upper ranges than in the low end. Overall, it came across more modern mix ready, at the expense of extended low end and a darker tonality that can help to glue a track together.

Functionally, the 273-EQ far exceeds the vintage Neve. For one, you don’t have to carry it around in an 80 series rack. Everything you need for high quality stereo recording is in the one box.

If you’re desperate for the Warm Audio 273-EQ to be a dead ringer for a vintage Neve… well, it’s not. That’s not a bad thing. It’s just different; it’s a very capable, great-sounding preamp, that’s well-built, has a gorgeous EQ on it, the convenience of a built-in DI and 80dB of clean gain. All for a very keen price. If you’re looking for a different, Neve-ish sound to add to your arsenal, then this is a solid bet.

If you’re low on quality channel strips and were looking around for a classic design to kick off your setup, then this is a great place to start.

ARCHÉ

Early this year, Expressive E released Arché, a suite of three stringed instruments (violin, viola, cello) designed specifically for use with the Touché controller. When you load up any of the three instruments in Lié, you can assign Arché-specific control options to Touché’s physical movements — things like Bow, Vibrato Gain, Vibrato Frequency. I found Arché a great primer to get comfortable using the Touché as a performance controller.

Assigning the Bow control to the Touché heel while twiddling vibrato with left/right movement lets you rest one hand on the Touché and another on a keyboard, offering an enormous amount of dynamic expression in quite a natural-feeling manner. Tapping the surface mimics a bow striking the strings, or you can emulate long legato bowed notes by pumping the surface like it’s the bellows of an accordion. Keyswitching and button pushing becomes largely unnecessary when you have this much tonal control under your palm. The whole experience is both fun and functional.

TOUCHING

Yes, Touché is unique. But how much value does it have once the novelty wears off? A lot, I say. Especially for people who care about getting the most from their virtual instruments; either on stage or in the studio. If you’re the type who’d rather input some notes with a keyboard and then draw in MIDI control automation in a more surgical manner, after the fact, Touché probably isn’t for you. On the other hand if you care about having more control options at your disposal while performing or recording — perhaps you compose for film or TV or produce music on a regular basis — then Touché offers you an extra set of tools to lay down the perfect take first go.

WAVEFIELD SYNTHESIS

To make sense of the spatial system, you have to get a handle on what the end result will achieve. For this particular show, there was no overhead fandanglery, or even surround elements. The setup of MPavilion just wouldn’t allow it. Instead, the focus was on building an immersive Wavefield Synthesis rig that could give the illusion that the sound of the PA was matching up perfectly with the position of the musician.

Why not just pan it, you ask? Well, stereo panning doesn’t work like that. The image might be maintained for one specific point in the audience, but you can never maintain that perspective across all punters. Wavefield Synthesis is different, it helps your ears locate an object to that spot no matter where you’re standing.

The rub is that if everything goes according to plan, you shouldn’t notice it’s working. It should just sound incredibly natural; like the PA isn’t even working at all. That was exactly the impression I got when sitting there waiting for New Palm Court Orchestra to start. Tait was quickly sound checking the chamber ensemble, and as each instrument went from acoustic to amplified, the only difference was in level. There was no sensation of the sound coming from a speaker, let alone being farmed out to the left and right. Big success!

3 KEY VARIABLES

There are three key variables to think about when designing a Wavefield Synthesis rig: The size of the area you’re trying to cover, the number of speakers you have at your disposal, and the distance the listener is going to be from the array.

The rule of thumb is is that the minimum distance between the punter and the closest speaker should be no less than the distance between two adjacent speakers. The closer the speakers are together, the higher the frequency you can render objects at.

In a situation where the audience member might be a metre from the closest loudspeaker, you need to therefore make the interval between speakers less than a metre. In an arena, they can be spaced further apart because the audience is likewise further away.

The speaker arrays can simply be arranged in a line across a stage, or completely surrounding the audience for a fully immersive experience. In this case, Yamaha set up a relaxed L-shape with the apex at the centre of stage. Tait used the Yamaha emblem to space each speaker 600mm apart. The reason for the tight spacing was that the array was doubling as foldback for the artists, and some were right up against it.

At that proximity, objects were being rendered accurately up to around 1.2kHz. “As you walk back, two things happen,” explained Tait, digging into the science of perception. “The ability for you to perceive frequencies higher than that increases, but the vector-based panning element becomes the localisation. At 4kHz, you’ve got all these speakers producing 4kHz, but your ears perceive the front of the speaker that hits you first with that sound because the other ones are lesser in amplitude.

“It turns out your ears are quite forgiving of that phenomenon. If you can properly render a wavefront within those critical speech frequencies, like 1kHz, then beyond that your ears perceive the wavefront at that frequency but also an amplitude-related directionality from that sound source. The associated delays are also in-phase, so your ear forgives it and locates the sound.

“Overall, with more loudspeakers, you’ll get a stronger image, but there comes a point where it kind of doesn’t matter.” What Tait it saying is that he could have used far less speakers if the double bass player wasn’t literally standing a metre from the array.

THE SYSTEM: LATENCY AS RELIGION

You can follow along in Figure 1, for a break down of the system in a conceptual manner, and compare it with the actual system in Figure 2.

All the analogue inputs were routed into the Yamaha CL1 via the Rio racks, while any digital signals from the artist’s DAW were tapped using Dante Virtual Soundcard.

Tait mixed on the CL1 in a similar way to a monitor console, creating 18 mixes that were then sent to a PC via Dante. They used a PC tower because the entire ‘religion’ with this system was latency. While Dante PCIe-R card drivers are available for Windows and Mac, PCIe slots aren’t a feature of Macs any longer. “DVS imposes a latency of 4ms, whereas the PCIe card allows you to get the latency down to 0.25ms,” explained Tait as to why the PCIe pathway was important. “The whole roundtrip latency had to be low because artists were performing live with it. It depends on the size of your stage — an arena stage might have the guitar amps eight metres from the drum kit, accumulating 20ms in time of flight — but here, anything above 8-10ms would have started to be noticeable. We got the entire roundtrip system latency down to 6ms in the end, and with audio stability.”

MORE OBJECTS, LESS SPACE

The PC was running Soundscape Renderer, open source software from GitHub. “It allowed us to automate the movements of objects over time. We synced up timecode with the performers’ kit over MIDI, then used a VST plug-in written for Soundscape Renderer inside Nuendo to read the timecode and run automation in time with it.”

While it’s tempting to turn every possible source into an object, Tait said it becomes pretty obvious that’s a bad idea. “The number of sound objects is limited to how cluttered the GUI gets,” he explained. “You could conceivably get 32 objects out of the CL1; 24 mixes and another eight matrixes. Early on, I loaded up every instance of a 32-channel live recording I had as objects. It becomes unmanageable. Especially if you want to move and place things.

“You’re better off busing the drums and everything that has a built-in spatial element to stereo. A guitar amp should be a single object. A larger instrument like a grand piano should be two objects at the most, spaced by a small amount. Backing vocals, stereo at most.

“The renderer we were using has an Android app, so if you want, you can bring a tablet around into the WiFi zone, and touch and move objects around. In much the same way the VST plug-in tells objects to move over TCP.

SPEAKS DANTE

Out of Soundscape Renderer, the final step was a Yamaha MRX7-D (a 64 x 64 Dante matrix) as master EQ, gain and limiting for overall system management. The outputs were sent back out to the 24 Yamaha DZR10 top boxes via the same Rio racks via analogue. Tait also stacked a pair of DXS18XLF subs at the array apex to provide the bottom octave below 50Hz.

Ideally, they would have used all Dante-equipped versions of the speaker, but the Dante versions are only a few months old and Yamaha only had a dozen that weren’t already sold off. “Rather than mix analogue and Dante top boxes — the small difference in conversion might affect the rendering,” figured Tait, “it had to be either all analogue, or all Dante. So we ran into the analogue inputs of the Dante-equipped boxes.”

SPACE VS MOVEMENT

Yamaha specifically chose a variety of artists for the show — starting with the drum, bass and guitar avant-garde of Turret Truck, followed by the New Palm Court Orchestra acoustic chamber ensemble. OK EG’s electronic set and LAPKAT’s art DJ-ing opened up the mix for plenty of movement, Cookin’ on 3 Burners mixed robust live performance with pre-recorded stems, and the event was capped off by a bit of Spoonbill eccentricity.

Throughout the show, artists used the system to varying effect. Some kept it simply as a localisation tool, others went wild auto-panning objects throughout the field.

“What we found in this whole discourse is that space is more important than movement,” said Tait. “The most important achievement these types of Wavefield Synthesis rigs gives us is not the ability to swing the hi-hat around at a million miles an hour. That overuse does no service to the artistic intent. The more important element is localisation. You have an amplified product a large audience can experience, but not restricted to hearing it through a left/right window. It’s a more aesthetically believable and comfortable listening experience.”

THE GLUE

Another tip Tait had was to create a ‘Glue’ stem; basically ‘everything everywhere’. “Spoonbill [the last performer] was throwing stereo stems hard left and hard right, and it worked a treat. Balancing it all, he had his groove stems like kicks and bass running through the ‘Glue’ stem.

“You had to have some things everywhere. Otherwise you get a bass groove coming from an object. Time alignment is so important to groove. If that object is physically displaced from other things, for some audience members it’s going to be ahead, for others, it’s going to be behind. With that established, then the ornaments become the objects.”

TIMING ISSUES & PHANTOM SOURCES

As a system engineer, designing a spatial audio system presents a new set of challenges. The first is changing the way you think about sub timing. In short, says Tait, “there’s literally no point in timing your subs. It’s not unimportant, it’s just impossible. What are you going to time them to? You’re moving sources around in space, so the alignment with the sub becomes irrelevant.

“The answer to that is using top boxes with the lowest possible crossover point. If you can use top boxes that are good down to about 50 or 60Hz, that’s your best bet. We were crossing over those subs at 50Hz, because the DZR10s were pretty good down to there.”

Tait says that while you can do a simple frontal system, it’s preferential to have the array wrap around the audience, and even better if it remains as one unbroken ring.

Tait: “You need to render the side, just to try and complete the wavefield synthesis wave formation. When you break the array — either to the extreme left or right — you introduce artefacts, which is just inescapable physics. The reason is, if you want to synthesise a wave coming from a point just to the left of the end of the array. Some loudspeakers within the array have to fire first, then the ones at the very end of the array fire a little bit later. If you’re standing inside the array, instead of perceiving the properly synthesised wave, your ears might localise the speaker that fires first.”

UPSIDES TO BE GAINED

On the upside, there’s generally more gain before feedback available in a system like this explained Tait: “It becomes a totally different formula, evidenced by the fact that for Cookin’ on 3 Burners we had a full band with drum kit, guitar amps, and a screaming Leslie cabinet, and a singer who could hear herself perfectly. And that was all running through the PA coming from behind them.

“If you have a gain before feedback issue, it’s just a matter of moving the object a little bit away, so you’ve still got the same SPL reaching the audience, and the performer still gets everything they need.”

AT A VENUE NEAR YOU

Tait says the learning curve shouldn’t be a turn off: “I’ve immersed myself in it for a few months so I’ve become familiar with it; but the amount of skill required is akin to setting up a Q-Sys, or Crestron, or big concert PA. Anyone that can do that, can do this. The time has come where the cost of everything is low enough where you should be able to get someone clue-y to get the software working and get a proper spatial system.”

With readily available tools, Yamaha proved it doesn’t take a team of acoustical engineers and software programmers to build a spatial audio system; and that it doesn’t have to be bankrolled by an arena tour budget. “We want to see venues and rehearsal studios that are putting a left/right line array — that probably shouldn’t anyway — cost up a spatial system,” said Tait. “It’s going to use many more, but less expensive drivers and amp channels. You can do a lot more with it, and even simulate a stereo PA system… if you want to.”