Tutorial: Tristan Hoogland

IT’S ALL ABOUT THE BALANCE

Start by pushing up the faders one by one, keeping a mental note of anything you might notice as you stack up the sounds. I’m usually keeping an eye out for anything that’s bloating the low end with the kicks, and anything that’s making the upper mids feel harsh with the snares, claps, percussion and hi-hats. Your first impression is usually right; if you feel like there’s anything you can address straight away that may be a problem later, I’d encourage you to deal with it then and there.

PHASE & TIMING

I spend a lot of time tinkering with the polarity and timing of the samples. Don’t just import stems then stack ‘em up and go, because it’s not uncommon for similar elements to audibly arrive later or earlier than one another. It might be that a sample has a 2ms gap at the start, or it was played in live with natural timing variations.

Start by identifying the main layer of each sound and get it in positive phase (meaning that the initial transient moves upwards/positively), then manually align the remaining tracks by hand. Next, establish a coherent phase relationship. One by one invert the phase of each track against the main layer, listening for the setting that provides the most focused sound.

For advanced polarity correction I use UAD Little Labs IBP Phase Alignment Tool, which continuously varies the phase between 0 and 180 degrees. Aligning the tracks, adjusting the phase and/or inverting the polarity can make for a night-and-day difference. Be sure to use your discretion when aligning tracks, however, as it may be the producer’s intention for them to land slightly differently.

EQ: THE ART OF CRAFT

EQ is a powerful crafting tool when attempting to make multiple sound elements work together as one. I tend to reach for subtractive EQ rather than additive EQ, with the goal of making space in one element for the other elements – especially when there are so many to fit in. Don’t be alarmed if you find yourself getting heavy handed; the more layers you have in a sound, the more EQ you’re likely to use to make them all fit. I’ll start with a medium to narrow Q and sweep through the areas of a track that I feel have excessive energy and are masking other tracks or distracting me from what I want to hear most in the sound, and reduce those areas accordingly.

KICKS: SCHOOL OF HARD KNOCKS

Kicks in synth pop music tend to favour ‘knock’ over click and sub, so I’m usually looking to reduce ‘pointiness’ (2kHz to 5kHz) and careful not to take away too much low mids (180Hz to 400Hz). In the example shown here I’m reducing 4.21kHz by -10.8dB and 715Hz by -4.3dB (bottom right), with a +4.2dB shelving boost at 98.7Hz on the kick buss (top left). Clickiness in the upper mids can usually be removed using a very tight Q without significantly altering the overall tone. You’ll notice I’m not adding any upper mids or top end on these EQs because I’m wanting to focus on the low end and low mids to get that ‘knock’.

I’ll do quite a lot of this work first on some smaller speakers, like Auratones or NS10s, and then switch over to the main monitors to make sure I’m not overlooking anything in the low end. Some samples will have a lot of resonance and bloom that will muddy things up or be out of tune with one another. I’ll experiment with high pass filters, sweeping all the way up to 200Hz on each sample just to see how much clearer I can get the low end, and then start parking it in a more reasonable spot (anywhere between 30Hz and 120Hz). Tip: FabFilter’s Q3 can be good for helping you ‘see’ the low end, and also for overlaying each track to see if you’re getting any excessive overlap between samples.

Another handy tool for shaping the low end is a transient designer. My favourite is iZotope’s Neutron Transient Shaper. The multiband option allows you to tighten the low end without affecting the rest of the spectrum. Setting the crossover between 80Hz and 180Hz can really help fine tune the ‘knock’ in the above band without adding a lot of crazy sub and bloating the low end.

SNARES, CLAPS & FINGERSNAPS

Snares and claps tend to build up in the mids and upper mids, resulting in unwanted honkiness and harshness. I spend a lot of time managing two areas in particular: 600Hz to 1kHz, and 2kHz to 5kHz. I rarely boost, except to add a top end bump if I feel like the cuts are creating a loss of overall presence.

One of my favourite techniques is using Lo-Fi (part of the D-Fi suite in Pro Tools) on uninspiring or ‘pointy’ percussion that lacks body and fullness, such as fingersnaps or claps. Simply wind up the distortion to taste and you’ll be surprised by how full the sound becomes without any other processing.

HATS

Similar to snares, I find in order to get them all working well together I need to make cuts in the higher end of the spectrum, generally the 3kHz to 6kHz area. I’d also recommend experimenting with Lo-Fi, bringing down the sample rate and anti-aliasing sliders because they can help make the hats more present in the mids and far less thin and pokey.

VINYL SAMPLES, iPHONE RECORDINGS & LO-FI DRUMS

Flavour tracks such as iPhone recordings and vinyl rips of live-played drums often make appearances in electronic music, but they need a lot of work to sit. I aim to have them sit in the mid range and leave the top and low end for the samples. Therefore I’ll start by setting a high pass filter quite high up to remove any unwanted rumble; it may end up as high as 300Hz. Old vinyl and iPhone recordings usually have more noise rather than musical top end, so I’m not afraid to be low passing down to 6kHz or lower if need be. You can automate these settings depending on how dynamic the arrangement is; for example, open them up if the rest of the samples disappear for a few bars. If it’s a complicated stem with a full kit I might use a dynamic EQ or multiband compressor, mostly just to keep the hi hats and snares in check.

PARALLEL LINES

One of the struggles I first had mixing electronic drums was getting them to have impact. My tricks from the days of mixing rock bands, such as parallel compression, didn’t cut it – things would just get louder, not bigger. I discovered that introducing distortion in addition to compression would pay dividends. My philosophy is that the compression adds ‘smack’ and keeps the drums solid, while the distortion helps them cut and adds aggression. (As a preface to the following, I rarely use compression on individual tracks. If hits are seemingly inconsistent, and assuming they’re samples, I’ll manually correct them with clip gain or automation. Beyond that, I’ll resort to saturation tools such as Lo-Fi on a case-by-case basis, and achieve the rest with parallel compression.)

Here’s how to do it. Start by creating three auxiliaries. The first will be our Drums Submaster, which we’ll route all the drum elements through with zero processing. The second aux will be for parallel compression, and the third for parallel distortion.

Establish the required start balance by adjusting the amount of each drum element that is sent to the Drums Submaster. Now send each individual track to the second and third auxiliaries respectively and set them to post-fader. Sending them discretely like this offers the freedom to not only drive more or less of each element into the parallel processing, but to have control if any one element is being brought too far forward or not forward enough.

Getting the parallel compression to work is straight forward. I usually opt for a Distressor with a medium ratio (around 3:1) along with a fast attack and fast release so it’s working quite hard – I aim to see 10dB to 12dB of gain reduction. The Distressor’s meter is quite reliable, and I expect to see a similar amount of gain reduction on each major transient event. If one or two elements are driving it less or more than the others then I know that my fader balances and/or EQ need revisiting.

For the distortion I use Soundtoy’s Devil Loc, or sometimes UAD’s Thermionic Culture Vulture. I always set the Devil Loc to Crush = 3, Crunch = 3, Release = Fast, Darkness = 2.5 and Mix = 10. The Devil Loc is an abusive unit and the trick to getting it to work is to have control of the level you’re driving into it. I’ll either assign a master fader to the bus or place a trim plugin before the Devil Loc so the individual sends can always be parked at 0dB. Starting with the fader or trim control all the way down, slowly turn up the signal going into it until you find the sweet spot – which I usually find to be quite low!

Now blend the returns of the two auxiliaries (parallel compression and parallel distortion) in with the Drums Submaster to taste. If things are feeling too harsh back off the Devil Loc return. I typically end up with a fairly even amount of each, but if the source material is lacklustre I might end up with more Devil Loc. Once you’ve learned how to dial these in – particularly the Devil Loc – you won’t mix drums without them again!

EFFECTS

Unless effects are printed with the sample my advice is don’t bother adding them. With electronic music, and especially with drums, you’re going for immediacy. Adding reverb will smear the transients and set things back rather than forward. I can also guarantee that it’ll be the first thing your client asks you to remove!

UNTIL NEXT TIME…

Hopefully the techniques mentioned here have armed you with some ideas for your own projects. Try implementing one or two into your existing workflow, such as experimenting with parallel processing or parallel distortion instead of in-line processing, and see how it works for you. The palette is always changing, but developing some tools and techniques that create a level of predictability in your workflow is one of the key things in mixing. Besides that, anything goes!

Next issue:

LADY LUCK

“It wasn’t until many years later that I realised we got a little lucky, especially with the ceiling,” said Storyk. “I went back and found out why the room worked so well, and it’s because of the ceiling, which looks like a flat propeller. It was partly aesthetics, and partly a half-baked notion that I shouldn’t have a parallel ceiling. I also remembered reading a little bit about membrane absorbers. They’d been used at RCA Studios and the great Kultura Studios in Moscow.  I didn’t completely understand the science and took a stab at how to build it. We used very lightweight plaster and the ceiling turned out to be a giant membrane absorber, which keeps the low frequency reverb time down in the room.”

Storyk was frank: “I’m not going to lie. I had very little science to back up what I was conceiving, but Eddie and I took a shot at it and it worked out.” It wasn’t until many years later that he realised it gave Electric Lady a desirable large studio characteristic of tailing down the low frequency reverb time. It tightens up the sound, as opposed to concert halls, where you want it to tail up and bloom.

“That’s what’s doing it,” said Storyk about the studio’s ‘sound’. “It’s not Jimi Hendrix’s spirit in the walls, the coloured lighting, the fact you have to slide in a tiny door on Eighth Street, or the river that flows under the studio. All interesting stories, but the science is in the ceiling… although we all still wonder if Jimi is still with us sometimes”

STUDIO PODS

These days, Storyk’s company, WSDG, that he started with his wife Beth Walters (the W) is averaging about 40-50 studios a year. In its lifetime, WSDG has worked on thousands, along with halls, theatres, AV design, conference rooms and structural acoustics, “but the studios are still what I love”.

Lately, those studios have been getting smaller and many are specifically geared towards podcasting. ‘Small format streaming facilities’ have become more and more common.

WSDG has been designing spaces for all the big players; Audible (owned by Amazon), studios in LA and NY for Midroll (owned by Scripps), and a huge facility for Gimlet, which got bought out by Spotify for US$200m this year.

Every radio station is essentially in the streaming game now, as podcasting on demand has turned breakfast and drive-time shows into ‘anytime shows’.  The variety of content and production styles could mean anything from a one-person/no-engineer facility to a four-person, fully-staffed interview setup. “In the case of Gimlet, a 45-minute Gimlet podcast could take months to make,” said Storyk. “It could have been done in 20 different segments with live people, phone interviews, and purchased material woven into the product.” Often the equipment is kluged together from off-the-shelf products not specifically tailored to podcasting, and part of WSDG’s job is to piece together high-end audio systems that can integrate phones — “not as easy as it sounds”— and be operated by a podcast producer with no engineering experience. When he saw Rode’s latest product, the Rodecaster Pro, Storyk was impressed and wondered why it had taken so long for a manufacturer to pull all those pieces together into a podcasting-specific console.

A lot of the studios also have intense isolation requirements. Gimlet’s facility features 10 tightly-packed studios, and because most of these top end facilities are in Brooklyn, Manhattan or Los Angeles, no space goes to waste. Admittedly, says Storyk, the majority of podcasting is centred around the voice, so the low frequency isolation requirements are often reduced. However, the smaller the room, the higher the frequency of the first order mode, which can often land in the speech range. Small rooms are often more complex to deal with than larger ones”.“

While you’d be forgiven for thinking a company like WSDG would only spec, bespoke, custom-designed acoustic treatment, Storyk is a fan of off-the-shelf products [see Beating the Low Frequency Trap]. He also lauds the ability that today’s modern iPhone can make 90% of the acoustic measurements we need. This is amazing! That means thousands of people can make measurements, which takes the voodoo out, and puts a little more science back into it.” Unlike some acousticians who would be happy to maintain the perception of their practice being a dark art, Storyk is pleased about the dissemination of measuring tools and the information available about acoustics. “People are getting smarter,” and Storyk applauds that. He would have killed for it when he was designing Electric Lady.

MARKETABLE ACOUSTICS

With over 3000 studio rooms under his belt, Storyk admits there have been times over the years when he has worried his interest in the subject would wane. Always one for the science as well as the artistry, the arrival of Time Delay Spectrometry (TDS) was one of those developments that piqued his interest along the way. TDS helped birth some of the industry’s most ‘marketable’ studio designs, and unravel others. “Live End Dead End came about when we had TDS and specifically when we had measurement devices that for the first time gave us data that mirrored what we were hearing. “Everybody realised that certain treatments were at the wrong end of the room. Chips Davis said, ‘Why don’t we put the dead end in the front, and the live end in the back?’ Peter D’Antonio then said, ‘We should put diffusion in the back.’ Which is a sharp idea. Then the acronym came about and it got marketed.” Nothing wrong so far, except that it makes “little sense for the low end. It made no sense to put all your low end absorption in the front, and low end diffusion or reflection in the back.” Storyk reckons it should have been called Frequency Dependent Live End Dead End, “but six-letter acronyms don’t roll off the tongue.”

Similarly, Storyk says the Westlake design developed by Tom Hidley “had some fallacies”. Again, the TDS analysis proved revealing. “We were now able to see that the compression ceilings were a mistake.”

The original intention of coupling a hard ceiling to a baffled hard front with big Augspurger-style speakers was to elicit more low end. Great news on one hand. The bad news was you would hear a first order reflection off the ceiling, creating disturbing comb filtering as energy bounced off the consoles. “It often occurred to me that you could have gotten more low end by just adding a sub,” said Storyk. “But people weren’t up to that.”

This isn’t criticism from the outside. In the early ’70s, Storyk and Hidley designed a room together for Stevie Wonder, which had a compression ceiling. It was Studio B, a small Quad room at Record Plant. “Stevie lived in it for two years with his engineers/producers Bobby Margouleff and Malcolm Cecil. One day, I walked in to one of the sessions, and there was a packing blanket covering the entire console. Bob had his hands underneath it while he mixed on the knobs and faders. “Bob said, ‘I don’t know what it is, but it’s better when I put the blanket over the console.’ Of course, the speakers up high were bouncing off the console, then off the ceiling, back off the console and back up again. He was essentially hearing a comb filter. He didn’t put the science to it, neither did I at the time, but his ears knew it.”

These days, one of the technological developments that most excites Storyk is Auralization. In short, the ability to listen to architectural drawings. “It’s pretty common now,” said Storyk. “Most acousticians will use it, though there’s only a handful of software manufacturers, and it’s not easy to use. We looked at the most popular ones (CATT, EASE, Odeon) and at the moment are still most comfortable with CATT-Acoustic.”  WSDG has Auralization rooms in both its Swiss, Berlin and New York offices. It allows designers to model acoustic environments and then listen to them in a calibrated room. “It’s one thing to read graphs, it’s another to listen.”

WHITE, WITH COLOUR

Storyk feels indebted to Electric Lady Studios. “Career tip: Make your first project famous!” He said. “Before it opened I had four more studios to design.”

At the time, Storyk could never have known the impact Electric Lady would have. Not just on the New York recording scene, but on the entire recording industry.

It was one of, if not, the first artist-owned studio. Jimi’s plan was unheard of; he would dictate the record making cycle to the label. “He was going to make his own records, at his own pace, then when he was done, he would hand them to the label. Few artists were doing that.” The studio was designed for artists, who were often seen as “the unnecessary people on that side of the glass,” said Storyk. “It didn’t matter how they felt; the quicker they got out, the better.” It meant he designed a larger control room, to make more room for artists on both sides of the recording process.

Jimi also wanted the furniture to be part of the mood, to make the live room feel like a living room. Coming from Cerebrum and then the shelved Generation re-design, Storyk simply figured recording studios should be architecturally interesting, too. That too was not the norm.

One of Jimi’s biggest takeaways from Cerebrum were the ability to shift the mood with colour-changing lights projected onto white walls. Back then, it wasn’t easy. At Cerebrum it was accomplished with 20 Kodak 35mm slide projectors. At Electric Lady, lighting designer Bob Walsch had to buy quantities of bulbs in three different colours, and attach them to giant dimmers to mix the colours. Four years ago, things came full circle. Paul Epworth had bought The Church Studio in London and wanted to overhaul the design. He’d worked at Ann Mincieli’s Jungle Studios in New York, a WSDG job, and wanted that specific Augspurger monitor system. It turned out one of the rooms at the The Church was almost the same size as the Jungle Studios control room, which worked out nicely. “In one of the conversations with Paul, we asked what he felt it should look and feel like,” said Storyk. “He said, ‘I’ve got this idea where everything’s white, and we just change the colours and the mood with lighting.’ I thought to myself, ‘I haven’t gone very far in 45 years!’

“I didn’t tell him that though, I just said, ‘That’s a good idea… I think we know how to do that.’”

THREE MICS AT ONCE

How about laying two of the microphones side-by-side, laying the third along the top of them like stacking logs, and taping them together? They all have tubular bodies of similar diameters, and when bundled this way each diaphragm would be as close as possible to each other diaphragm. There are two problems with this approach.

First, these are all end-address cardioids with ports behind the diaphragms. Bundling them together as described here means each microphone will have its ports obscured at two points (one point from each of the other microphones). Whether or not that will affect the tonality of any of the microphones is debatable, but as long as it is debatable the comparison recordings are meaningless.

Second, each of the three microphones has an outer diameter of around 20mm, so the closest any two diaphragms can get to each other is around 20mm centre to centre. That distance is unavoidable, of course, but could result in each microphone capturing a different tonality from the instrument if the relative orientation of the microphones is not considered. Which orientation provides the least tonal differences on the instrument being reviewed: vertical, horizontal or diagonal? Get it wrong and the comparison recordings become meaningless.

UNMATCHED PAIRS

I decided to focus on two microphones at a time: the TF-5 and the one I was comparing it against. I’m not comparing the KM184 against the 4011, so I don’t need them in the same test.

The next decision was placement. As any experienced engineer knows, the best spot for one microphone may not be the best spot for another. To avoid playing favourites, I used a consistent placement of 30cm from the guitar, perpendicular to the soundboard, with both microphones aimed at the point where the guitar’s neck joins the body – my preferred starting point when recording acoustic guitar. For the distant miking tests I pulled the microphones back while keeping them focused on the same spot.

For orientation, putting one microphone above the other (rather than side by side) yielded better tonal consistency during subtle performance movements of the guitar – the tonal differences between microphone positions was significantly smaller than the tonal differences between the microphones themselves, and therefore irrelevant. I left a small gap between the microphones to avoid any possibility that one microphone was blocking the ports of the other.

To ensure consistency between comparisons, the guitarist composed four short pieces to reveal different aspects of each microphone’s performance: long slow strums, fingerpicking, muted strums and full-bodied strums. Finally, I sent a blend of the two microphone signals to the headphones so the guitarist could ‘work’ the microphones; staying on-axis and subtly moving closer and further back as desired, just like any normal recording session.

MATCHING PERCEIVED VOLUMES

The final step in preparing the comparison files was to carefully match the perceived volumes of each pair of recordings. This was essential because the TF-5’s frequency response includes a shallow dip from 2kHz to 7kHz, falling to almost -1dB from 4kHz to 5kHz. This dip includes the range of frequencies that human hearing is most sensitive to, meaning the TF-5 will probably sound lower in perceived volume when compared to a microphone without such a dip. In contrast, the KM184 and 4011 both have a subtle boost in the upper part of that same range of frequencies. Any attempt at simply matching the peak levels or matching the waveforms consistently resulted in the TF-5 sounding dull and uninteresting in comparison to the others when, in fact, it was being reproduced at a slightly lower perceived volume. As any experienced engineer or hi-fi salesperson knows, if you want someone to prefer one sound over another make it louder. Without carefully matching the perceived volumes, the comparison recordings would always favour the louder of the two files and would therefore be meaningless.

The User mode is of the greatest interest. The first of its 16 slots is preloaded with Waves by Korg. At the time of reviewing the Prologue, third-party Multi Engine oscillators were yet to emerge. Since then, a small but growing core of developers has stepped up to the plate. Thankfully oscillators developed for the Prologue will import to the Minilogue xd. [Refer to ‘Third-Party Multi Engines & Effects’ for links.]

For the sake of the review, I installed the Freeware oscillators offered by Peter Allwin and Jason Long. Exploration of these oscillators immediately showed how much the Multi Engine expands the sound palette, with many being rich in harmonic content. These additional oscillators inspired many new patches, and in several cases I even found it unnecessary to have the analogue VCOs layered to get satisfying tones with plenty of movement.

The highlight for me was Jason Long’s developments; in particular, his Chips 2.0 oscillator which brings vintage digital game console sounds into play. He has been quite prolific of late, having released an additional two effect algorithms, Bucket and Hera, and also a preview video of a custom wavetable/sample import tool he has in development with full key-range mapping and more: https://tinyurl.com/yxbw3h6k

With the sheer number of new oscillators I’ve discovered so far, it was apparent that having 16 slots for third-party oscillators may soon become a limitation. It’s important to note that swapping algorithms in and out of the synth will break patches unless the same oscillator types are loaded back into the correct slot number, because patches reference oscillators based on the slot they’re loaded into rather than by name.

FILTER SECTION

The VCF is a two-pole (12dB/octave) low pass filter making use of Operational Transconductance Amplifiers (OTA). It has a rich, powerful and consistent sound when swept across the audible frequency spectrum, and is capable of self-oscillation. The cutoff frequency can be modulated in a bipolar manner from the EG or LFO. New for the xd (and just like the Prologue), the filter now features a drive stage offering two preset amounts of drive: 50% and 100%. VCF keyboard tracking can be engaged at 50% or 100%. With resonance at self-oscillation, you’ll find the key tracking is scaled to be in tune (great for high resonance white-noise leads).

The four-pole mode of the original Minilogue is absent for the xd, fueling speculation that Korg has repurposed the filter circuit from the Prologue for the Minilogue xd. Notwithstanding, to my ears it sounds slightly different.

MODULATION MATRIX/SOURCES

The modulation matrix is basic, nevertheless the essentials are there to create ample movement. The xd has two envelopes: a traditional ADSR (hardwired to the VCA), and a simple AD routed to either VCF Cutoff or the pitches of VCO1 or VCO2. A single LFO is provided capable targetting one of either VCO Pitch, VCO Shape or VCF Cutoff at a time. There are additional LFO destination options in the menu, where it can target specific oscillators. (The multi-engine shape parameter has its own dedicated envelope also hidden in a menu.)

On the back panel, you’ll find dual CV inputs designed to interface with Eurorack hardware. For those seeking more expansive modulation options than those available internally, you can easily integrate all manner of exotic modulation sources from your Eurorack modular synth. There is a total of 29 different modulation targets available for the CV1 and CV2 inputs. Additionally, the upward and downward travel range of the Joystick’s Y-axis may be independently mapped to the same list of 29 modulation targets.

EXPANDED (& EXPANDABLE) EFFECTS

The original Minilogue’s effect engine was a mono bucket-brigade analogue delay that was a touch primitive and noisy. The effects section has been vastly overhauled for the xd, inheriting the same lush-sounding stereo DSP Effects Engine of the Prologue series.

The first slot provides Modulation effects: Phaser, Flanger, Chorus, and Ensemble. The second and third slots offer Reverb and Delay respectively. Only the Rate and Depth parameters are available to tweak – all other parameters are hardwired in various factory presets. While each algorithm sounds excellent and the presets are well-appointed, like anything preset, I did crave the ability to tweak further.

Similar to the Multi Engine oscillator, additional effect algorithms can be installed with 16 slots for the Modulation FX and eight slots each for Reverb/Delay effects. At this point, I can’t help but recall the excitement of loading third-party effects into my Ensoniq EPS16+ and ASR10 samplers back in the ‘90s. A brief run-down on what is available is covered in ‘Third-Party Multi Engines & Effects’.

ARPEGGIATOR & SEQUENCER

Like its predecessor, the Arpeggiator is implemented as a feature within the Voice Modes, rather than being an independent entity, and now features a Latch function. Thirteen arpeggiator modes are available on the Voice Depth control with some even providing polyphonic modes, which is a nice touch. Rather than a dedicated octave span setting, you’re limited to what is predetermined within these modes. The decision to implement the Arpeggiator within the Voice Modes means it’s not possible to have arpeggiated patches with the unison engaged simultaneously.

A polyphonic Sequence can be attached to each patch memory independently and may have up to 16 steps. Sequences can easily be entered in step mode or recorded and overdubbed in real-time alongside a metronome. A powerful Motion sequence feature allows capture of knob movements for up to four parameters simply by recording movements during playback. It’s also possible to have Motion sequences defined without having any note data in the sequencer, effectively providing four beat-synced, 16-stage looping envelopes – a useful means to extend the modulation facilities of the synth.

Because the VCO levels can be automated, it’s possible to simulate complex multitimbral sequences by bringing different VCOs and/or the Multi Engine in and out of the mix on different steps. This is further enhanced by the fact the Multi Engine can be set to bypass the VCF within the menus. The possibilities are capably demonstrated in a video tutorial by BoBeats, here: https://tinyurl.com/y3lgnx55​.

My only real disappointment is that Sequences can’t be transposed in real-time by striking notes on the keyboard as part of a live performance; which is a missed opportunity because that’s the essence of making step sequences really come to life.

I had issues with the Sync Source setting in the Global Edit menu providing only ‘Auto’-based settings for slaving to external clocks, either via MIDI or USB. For example, if you are playing the Arpeggiator (Latched) whilst slaved to MIDI clock and you stop the transport, the unit assumes you no longer want to use an external clock and, after a brief moment, resumes playback. Hopefully, Korg considers adding non-auto sync options.

CONCLUSION

The Minilogue xd packs a punch in terms of features, considering its size and price. Far from being a mere incremental feature bump, the Minilogue xd is a substantial refresh of the original concept – so much so that it really is a different synth: sounds created on the original Minilogue cannot be imported into the Minilogue xd because the architecture differs too much.

Despite its limited modulation options, after exploring what the Multi Engine oscillator adds it’s clear that the Minilogue xd is capable of a sound palette far beyond a dual VCO synth. Once you program your own patches to taste and think inventively to achieve movement within sounds, it really sings. The icing on the cake is the inclusion of the Prologue’s Effects engine; on the original Minilogue it was difficult to evoke really big lush sounds without using an external effect unit, but with a few knob twists and switch flips in the Effects section of the Minilogue xd you’ll quickly find yourself in sonic nirvana.

It was effortless to coax everything from angular aggressive digital basses and snarling dirty leads through to floaty sublime keys and pads. I found myself inspired to create new patches every time I had a session with it. If you’re in the market for a modern four-voice polyphonic analogue synth with the extra fruit that only hybrid digital synths can provide, the Minilogue xd is definitely worth checking out.

SOFTWARE

At first glance the Orion’s software panel is sleek, modern, well designed and, most importantly, scalable!

There is nothing worse than a minuscule panel on a 27-inch screen that you cannot see, let alone operate; so, thank you Antelope! Tabbing through the different sections while making selections is seamless and intuitive. The Mixer is laid out over five windows to represent the seemingly endless channels, and cycling through them is a breeze.

The wheels began to come off the cart for me in the Matrix section. Let me be very clear that this is an evaluation from an ordinary person trying to make sense of an extraordinary routing system, and the fault here does not lay entirely with Antelope. The Matrix section is, in a word, brilliant; however, some of the labelling decisions are questionable. Though the Matrix is colour-coded and it is easy enough to assign I/O, I got completely confused trying to assign outputs to my monitoring section. The loose cog in the machine is in the naming of the I/Os. ‘TB Play 1-32’? If you are like me and did not initially watch the videos or read the instructions, ‘TB Play 1-32’ could be anything! What I later discovered is that it’s the assignment for ‘Thunder Bolt Playback Channels 1-32’ – the digital returns from your DAW via Thunderbolt. Once this got cleared up, the ability to route DAW outs to the Orion’s analogue outputs became blatantly obvious and actually quite easy. Maybe Antelope should re-label it as ‘DAW Returns 1-32’ or similar for those of us who are keen to get started without watching videos or reading instructions!

RECORDING

Low latency is one of the big talking points of the Orion series, and for good reason. The ‘sub-millisecond’ (ie. less than one thousandth of a second) latency is not long enough to throw off any musician, and I would call out those who say they are affected by it. For me and the artists I worked with – and this included Midi-based musicians – the Orion performed outstandingly. With an electronic drum kit triggering Kontakt’s ‘Abbey Road’ series of drum kit plug-ins, the response and feel of the audio coming back through the monitoring section was smooth and felt to be in perfect time. Recording audio was obviously no drama due to the vast number of channels; the only limit was how many preamps I was able to stick in front of it!

The Orion’s software comes provided with a number of DSP plug-ins from their FPGA FX platform, including their Master De-Esser, Power Ex expander, Power Gate, Power FFC compressor, Clear Q five-band parametric EQ, and Aura Verb reverb. Additional plug-ins can be purchased and added later, with over 30 to choose from. I must admit that I only gave the installed plug-ins a cursory glance because I already have an exhaustive library of top-quality plug-ins, and I assume anyone in the market for the Orion would have their own plug-ins as well. Having said that, the Orion’s built-in reverb in the monitoring section is a stroke of genius. The vocalists I recorded with the Orion were so grateful and happy to have reverb quickly added to their headphone mix without the time-sucking task of patching in a unit or dealing with troublesome DAW latency issues that often cause unnatural artefacts. By quickly dialling in a bit of reverb, I kept the workflow uninterrupted and ultimately more fruitful than it would have been otherwise. The reverb has a comprehensive feature set, sounds pretty good and serves its purpose brilliantly.

SOUND

I am not a fancy man when it comes to the monitors I use for mixing records. I have a pair of Event 20/20s that I’ve made many records with; I know them well and I trust them. After installing the Orion I pulled up a current track that I’d been previously mixing through my Apogee converters. The sound through the Events came alive in a way I had not previously heard. The soundstage widened, there was greater separation between instruments, and the plane of sound felt as if it had jumped towards me by another six inches or so. The instruments that were compressed sat back in the way you would expect while the more dynamic elements leapt from the speakers, creating a thoroughly pleasing three-dimensional space. These improvements in soundstaging and separation were even more evident when I switched to my AKG K712 open-back headphones.

In addition to being an insufferable nerd about recording and mixing, I lead a double life as an insufferable audiophile. Perhaps the two are not mutually exclusive? After the Orion’s success in the studio, I decided to take it into my living room, connect it to my hifi and see how it faired in the hifi context of listening for enjoyment rather than making critical mix decisions. I have an extensive range of hifi equipment but my main system is built around a Cambridge Audio DacMagic Plus driving a pair of B&W 606 speakers through a Harman Kardon HK980 amplifier. I connected the Orion’s Monitor outputs to the HK980 with a pair of 6.35mm jack to RCA leads. I don’t think it will shock anyone who has heard Antelope’s conversion to find out that, yet again, the sound quality was peerless. The same soundstaging mentioned above came into sharp relief as the speakers breathed out Nick Cave and the Bad Seeds. It was an enveloping aural experience, but buying the Orion 32+ as a stereo hifi DAC would be a ridiculous notion to most rational audiophiles. It was fun though, and it certainly showed the capabilities of what the Orion can do. I fear my Apogee days may be limited…

GRIPES

I have a couple of gripes about the Orion 32+ Gen 3, and they are to do with its monitoring section and front panel. Whilst I love the idea of having routing presets, access to these from the front panel seems superfluous. I would’ve preferred a dedicated Monitor volume pot so I did not have to rely on software to control the level to my main speakers, which is not ideal and hinders workflow. It’s not a problem if you have a dedicated monitor controller, but I would prefer a front panel volume control for this purpose.

A headphone socket on the front panel would also be desirable. At only 1RU high the Orion has the potential to be a semi-portable device but offers no way of independently hearing anything coming out of it without, at the very least, connecting a headphone amplifier. I know this is nitpicking and the Orion is not marketed as a stand-alone interface, but having buttons to select routing presets seems like a waste of precious front panel real-estate when a dedicated Monitor out volume control and headphone socket would be far more useful. If Antelope is reading this, please consider these suggestions for Gen 4.

WRAP UP

The Orion 32+ Gen 3 provides a perfect bridge between a console and DAW, with a near-infinite amount of routing possibilities and enough channels to handle most large format consoles. It’s poised to compete with top level studio-grade interfaces like Avid’s HD Systems, albeit with more of a ‘free market’ mindset. It’s also very well suited to live recording applications – nothing could be simpler than a MADI connection into the Orion to create a bridge from a live console to a DAW-based recording system. Those with home studios would consider it overkill for their needs; for that market Antelope offers the Zen Tour and Orion Studio HD, and they make this distinction obvious through their marketing.
At $5499, the Orion 32+ Gen 3 is a considerable investment but if you’re looking for an ingeniously-designed and spectacular-sounding alternative to the status quo of studio grade interfaces, you will be sated and inspired.