Tutorial: Tristan Hoogland

Deplosive aims to remove plosives. While it’s effective, there can be a trade off in the body of the vocal if used inadequately. Settings vary, but I start with the frequency set to 150Hz, sensitivity to 5.0, strength to 6.0, and adjust to taste.

For both modules I duplicate the main playlist, process the entire audio in AudioSuite and fly up the processed region as a click appears. Avoid batch processing as it can affect diction (although I’m less cautious about that with doubles and BVs, as discussed later).

VOCAL PROCESSING CHAIN

My vocal processing chain is almost always the same:

1. De-Esser #1
2. De-Esser #2 
3. Corrective EQ
4. Flavour/Additive EQ
5. Compressor/Limiter #1
6. EQ
7. Smoothing Compressor #2
8. Tone
9. EQ

Sometimes I’ll use all of this, sometimes I’ll use none. I start with everything in bypass and, while the order rarely changes, I will engage whatever I feel needs attention first. For example, I may engage Compressor/Limiter #1 while getting a balance going, then some EQ, maybe some de-essing, then back to Compressor/Limiter #1, etc.

SSSIBILANCE

The best place for a de-esser is at the top of the chain. This allows us to control sibilant events and dial in top end afterwards without the vocal sounding harsh. I use two in split-band mode, each focusing on different areas of the vocal; the first targets higher sounds like ‘T’ and ‘S’, the second targets lower sounds like ‘Ch’, ‘Ka’ and ‘Sh’.

For male vocals I’ll typically set De-Esser #1 somewhere between 6kHz and 20kHz, and De-Esser #2 to somewhere between 2kHz and 5kHz.

For female vocals I’ll typically set De-Esser #1 to somewhere between 6kHz and 20kHz (same as male vocals), and De-Esser #2 to somewhere between 3kHz and 6kHz.

Ideally we want these working independent of each other, so that each is catching things the other isn’t. It’s not uncommon to see anywhere between 5dB to 10dB of gain reduction going on in each instance, but use your ears and aim to have them tonally sitting in alignment with the rest of the vocal passage.

CORRECTIVE EQ

The trick to having a vocal sound clear and present is less about cranking high end, and more about better management of low end and mid frequencies. With the vocal up, evaluate what state it’s in and what areas need attention first. I find it to be typically one of three things:

Murkiness/boxiness (150Hz to 400Hz)

Honkiness (500Hz to 1.5kHz)

Harshness (2kHz to 5kHz)

A good rule of thumb is “the lower the frequency the broader the Q, and the higher the frequency the narrower the Q.”

I’ll apply a high-pass filter to remove any rumble and other excessive non-musical energy. I’ll do this by sweeping it up to around 400Hz, and then bringing it back down until the integrity of the vocal is regained. This could be as low as 80Hz or as high as 300Hz. 

Low mids are the most delicate area to get right. Too little and things sound anemic, too much and they sound dull and unclear. Attenuating 200Hz to 400Hz with a moderate Q (around 1.0 to 3.0) reduces cloudiness and improves intelligibility, particularly if the singing is in a lower octave. Conversely, if there’s a lot of belting you’ll want to retain more of the low mids. Further up there can be some boxiness and honkiness that’s worth reining in; 500Hz to 1.5kHz can be a honky area that I’ll usually attenuate with a medium-tight Q. I’m always careful around this area, however, as I find it is where the character of most vocals are found.

High register sung vocals generally have a lot of energy in the upper mids, which can be fatiguing if not managed well. I rarely do broad cuts here, preferring tighter attenuation of harsh resonances. It’s not uncommon to find one or two hot frequencies that spike between 2kHz and 5kHz, and I’ll use FabFilter’s Pro-Q3 with a moderate to tight Q (3.0 to 10) and pull it down, anywhere from 4dB to 15dB depending on how bad the offending frequency is.

EQ, BUT MAKE IT DYNAMIC!

A useful feature available in many EQs is the Dynamic feature. This operates somewhat similarly to a multiband compressor, but it works particularly well on vocals as we have finer control over the Q compared to a multiband. I resort to a Dynamic EQ when the resonance of a vocal occurs infrequently, or when the vocal moves up and down through the register frequently.

Instead of continuously removing the frequency with a standard EQ, I’ll engage the dynamic function and bring down the threshold until I achieve uniformity. Ideally we want it to attenuate the target frequency only during heavy moments that bring out the resonance, and to relax at other times.

TO BOOST OR NOT TO BOOST?

While I prefer reductive EQ, it’s sometimes beneficial to have an additive approach. When boosting I’m looking to enhance areas that feel under-represented, typically lifting the top end to improve intelligibility. Furthermore, applying de-essing can result in a duller tone that we may want to compensate for. For this job I’ll reach for character equalizers like an SSL or Pultec. The SSL is a sharp and focused equalizer, whereas the Pultec is smoother and lighter.

On the SSL I’ll usually open up the vocal by adding some air around 8kHz to 14kHz, which helps it sit a little more up top of the mix. With the upper mid band, I’ll find a spot between 3kHz and 7kHz that excites the transients of the vocal, honing in on nicer qualities of those S’s and Ch’s; basically, anything that brings the voice forward. On the Pultec-style EQ I’ll often reach for something like 10kHz or 12kHz with a wide bandwidth and open it up. I’m more conservative when boosting, rarely pushing more than a few dB as I rely on reductive EQ to do the heavy lifting.

COMPRESS TO IMPRESS

When compressing it’s always best to focus on tone, not dynamic control. I typically like to use two compressors in series for vocals; the first is something fast that’s quick to grab the front of the vocal, the second is something slower that smooths out the overall performance.

The first compressor in my vocal processing chain is an 1176, which I have set up with the attack at 3, release at 7 and a ratio of 4:1. What I’m looking for here is to clamp down on the heavy moments in a vocal, particularly the loudest passages. This will, in turn, alleviate the amount of work the second compressor needs to do. I’ll increase the input until I start to hear the vocal thicken up. Depending on how dynamic the vocal is, the gain reduction meter could be working anywhere from 7dB to 20dB during the loudest passages! Having it operate between 3dB to 7dB most of the time isn’t unusual. The 1176 has a lot of mid range character and so I may pull out some mids afterwards.

The second compressor I reach for is RVox, but sometimes it can be an LA2A or TubeTech CL1B. RVox is a great vocal finalizer that, to my ears, compresses and limits the signal nicely. How hard I work RVox varies greatly depending on the state of the vocal; sometimes knocking back just a few dB is more than enough and sometimes slamming it 20dB works wonders. It often requires a bit of back and forth with the 1176 to get that relationship working.

WHY EQ BEFORE COMPRESSING?

It’s mostly about tuning the EQ to the compressor. The reason we EQ before compression is because we want to feed our compressor a signal that won’t be continually forced into gain reduction based on non-musical or undesirable excessive energy in a particular frequency range (e.g. usually too much low end). The other part of the equation is that after some time you learn what a compressor wants to see going into it. I like to think of this as tailoring the signal to the compressor so it can do its job more efficiently. That’s why it’s important to do these steps in conjunction with one another, and to go back and forth as you introduce a new processor down the line. You get a sense of what each device wants to see.

TONE/DRIVE

Tasteful compression usually brings all the drive and saturation I like in a vocal, but there are times where recordings come in that, even after processing, feel completely lacklustre. My favourite tool for this is Soundtoys Radiator. I set this up at the end of the chain (post compression) where I have a steady signal to feed it, because having it respond to a dynamic signal makes it harder to park. I’ll either drive it hard and mix in a small amount, or be more conservative and have the mix at 100%. In both cases I’m referring to a mild injection of tone rather than full over-the-cliff annihilation, but that can also be achieved.

SPLIT IT!

I like to split the vocal across two tracks, one for the verse and one for the chorus. This offers flexibility; EQ and compression settings can be better tailored to suit the vocal range and intensity of each part respectively.

EFFECTS

Producers typically provide you with effects of their own and in most cases I’ll make those work, but I may expand upon what’s already there or recreate where appropriate. Each song is prescribed a unique dosage of effects and it’s important to know what the producer has in mind when including them. I like to use a combination of effects and always have the following ready at my disposal:

Pitch-Shifting/Chorus: Microshift, Dimension D

Reverbs: Small Room, EMT Plate #1, EMT Plate #2, Spring, Long Hall

Delays: Slapback, Medium Delay, Long Delay

PITCH-SHIFTING/CHORUSING

Pitch-shifting can really help in achieving apparent width and size. My favourite tool for this is Soundtoys Microshift. My default is setting II with the focus knob at 160Hz and I’ll typically have the send set anywhere between -10dB and -20dB depending on how much I’d like. Exaggerating the Detune and Delay knobs can help in achieving even more ambiguity and width.

ROOM

Adding a room reverb is a nice way to add a bit of diffusion to the vocal. I like using it if the lead is sounding too stark. I don’t use room reverb’s too often when mixing pop, perhaps 30% of mixes where the arrangement is sparse. I generally prefer it to be very short and unnoticeable; my favourite plugin for this is the Medium Room setting in ValhallaRoom. It tends to be quite spitty and bright so I’ll usually bring the hi cut way down with a shortened decay (60ms to 90ms).

EMT 140 PLATES

Plate reverbs are complimentary to vocals due to their clear and rich character. I like UAD’s EMT 140 and I’ll toggle between plate A and B depending on the colour I’m going for (A is brighter, B is more neutral). I’ll set the decay to taste, but two seconds is a decent starting point. I enjoy introducing a moderate pre-delay of around 30ms to 80ms depending on the song, but extending the pre-delay to longer extremes like 200ms can be fun. Ultimately I tune this by the feeling, and by the phrasing of the vocal.

LONG HALL

My favourite hall for this is the Concert Hall algorithm in Valhalla VintageVerb. It has a very nice, hazy furriness to it that works well at creating a large backdrop without drawing too much attention to itself. The default setting is surprisingly effective in most scenarios, though I may extend the decay time if the song calls for it or I am doing a large throw. It is heavy on the low end so I’ll usually roll off a lot both before and after the effect so it doesn’t take up as much headroom. I like to try to get this to work as an extension of the plate.

SLAPBACK

I like UAD’s RE-201 for this duty. It’s got a thin ‘mid-rangey’ sound to it and distorts on the front end nicely. I use it in multi-mono mode starting with both channels linked to set the rough delay amount I’m after – anywhere between 80ms to 130ms depending on the song. I’ll unlink the channels and offset each side slightly (e.g. 10ms to 15ms up and down respectively). Driving the input fuzzes up the delay nicely and also helps get a more consistent, compressed like level coming out.

MEDIUM & LONG DELAYS

Delay’s are generally more effective than reverb in dense pop arrangements as they take up less room in a mix. I have two delay channels set up in my arsenal, a medium and a long delay.

I like using Soundtoys EchoBoy. My usual aim for the medium delay is to feel like a tucked in reverb – if I can hear the repeats clearly, it’s too loud. First, find a subdivision that works well for the feeling you’re going for. Sometimes this ends up being 1/4th or 1/8th, but it surprises me how often I’ll end up with something slightly off beat. I’ll set the feedback to anywhere between 10 and 2 o’clock depending how deep I want it to feel. I’ll adjust the hi and low cuts, focusing on aspects of the vocal that every so often make the delay jump out. I’ll also remove quite a lot of low end, focusing on the dry vocal while making this adjustment and find the spot where the dry is clearly dominating the low mid frequencies without clutter.

I’ll reserve the long delay for throws, which conversely to the medium delay are to be noticeable. We can do this off a send, but I duplicate the lead vocal track and its processing and throw the delay on as the last insert. This allows me to control, with great precision, the words I’m wanting to throw off as I can trim the region, fade in or out, etc.

PRE- & POST-EFFECT PROCESSING

Implementing EQ before and after your discrete effects returns pays dividends. This offers the flexibility to change the character of the effects by altering the sound going into them and also processing the sound coming out of them. I’d also encourage experimentation around compression and distortion, both pre- and post-effects. For example, compressing the effects return can almost act like a sidechain from the dry signal, pulling back the effect to allow the dry signal to creep through and return in the gaps between phrases.

BACKING VOCALS

There are numerous types of backing vocals that may exist in an arrangement, but for the purpose of this segment we’ll focus on the most common variant: harmonies.

Where possible I prefer to work with groups rather than individual harmony parts, primarily because the balance of the group is what the artist and producer are used to and it’s easier to work from a place of agreement. If more freedom is needed further down the track, then we can resort to the individual layers.

I’ll approach things like de-essing, mouth declicking and so on similarly to how I described earlier for lead vocals. Unlike lead vocals, however, my treatment on harmonies is quite broad and aggressive because I aim for them to surround the lead vocal when they appear. I like rich sounding harmonies and this often means attenuating low mids and mids, along with some heavy compression, all in the interest of keeping the BVs clear of the body and focus of the lead vocal.

As can be seen in the example, that is exactly what I’ve done to this stack of harmonies: a high pass filter at 220Hz, a broad 9dB cut around 250Hz and another cut around 900Hz. I also added some air around 14kHz on the SSL. Following the EQ is the TLA-100, which is just doing a little bit of musical compression (around 3dB max) followed by RVox, which is doing most of the heavy lifting – slammed quite aggressively (averaging 12dB to 15dB of GR) to bring out those upper harmonics and keep them solid. It’s important to note that these vocals were recorded very well and adjustments were made based on how they related to the lead.

In regards to effects I’ll first consider what’s on the lead. If the lead has a significant amount of ambience I may keep the BVs mostly dry, which in context is hard to detect if there’s a lot of reverb already present. More commonly I will send them to UAD Dimension/Studio D to add a bit of density and richness. If I feel like they need some reverb I’ll usually send them to a plate – typically Valhalla’s Plate, which I like because it’s quite ‘vanilla’ and thin sounding.

In stark comparison, remember those $20k DSP-assisted Pro Tools systems I mentioned earlier? There’s no latency roadblock. What enters the mic is heard via the system less than an imperceivable millisecond later – and less at higher sample-rates. All the time, every time, regardless of the speed of the CPU, and you can load that input track with plugins should you wish. That’s why those interfaces go for the big bucks. Dedicated DSP ain’t cheap.

As I outlined initially, the entire concept of the audio interface has evolved from basic I/O into the ‘nerve-centre’ paradigm. Now a single unit typically takes up those multiple outboard duties. Alongside a plethora of I/O you’ll find multiple headphone outputs, switching between different monitor pairs, dedicated talkback microphones (or the option to add talkback via a mic input), metering, high impedance instrument inputs and more. Which brings us to the pinnacle of the concept: Avid’s latest all-in-one ‘production centre’ interface: Pro Tools | Carbon.

CARBON AVB

Unlike many interface choices, Carbon doesn’t connect to your host computer via USB or Thunderbolt. Avid has taken the Audio Video Bridge (AVB) route, as MOTU has with its professional line of interfaces and as Presonus with its live audio products. Connection is via a Cat-6 ethernet cable, or if you’re using a Thunderbolt-equipped Mac you can use Apple’s Thunderbolt-to-Ethernet adaptor. Avid has limited Carbon’s compatibility to Thunderbolt equipped Macs only and macOS 10.15.6 (Catalina). I assume, as is the Avid way, the company is treading the compatibility line with an abundance of caution. While AVB is an open standard that can function in Windows environments, and via USB 2 and 3, not all ethernet chipsets support AVB, and USB2 connection would possibly not play nicely with Carbon. As it happens, the ethernet chipset used by Apple supports AVB so this is where the compatibility cards fall. Equally, Carbon runs Pro Tools only. Why’s that? Oh do read on…

Here’s the big deal. Carbon has in-built DSP. Carbon is touted by Avid as a hybrid device in the sense that it can use native CPU-based plugins, just like any other native-based interface, but it can also utilise DSP-assisted plugins courtesy of its eight built in HDX DSP chips. These are the very same DSP chips you’ll find on an HDX core card, albeit eight rather than 18. What this means is that Carbon can provide sub-millisecond latency when recording, even when using plugins instantiated on record tracks, into Pro Tools – precisely like those expensive systems I mentioned earlier.

Now I hear the shouting. Yes, I know there are other audio interfaces on the market offering onboard DSP. Universal Audio’s Apollo range immediately springs to mind but there are others – they’re all at it – but those systems still require a separate mixer/monitoring application to monitor audio directly from the interface to alleviate latency. Carbon, on the other hand, combined with Pro Tools, avoids this conundrum completely by virtue of Avid’s AAX plugin format. Because AAX plugins run in both the native CPU arena and on Avid’s DSP processors, it’s transparently simple to swap from one processing base to the other (CPU or DSP) and maintain the exact same sonic effect. Everything is done within Pro Tools. Conversely, with the Universal Audio example, UA’s plugins run on UA’s DSP – and nowhere else.

How’s all this integrate? Within Pro Tools the switch between DSP and native AAX is virtually seamless. Each channel strip within the Pro Tools mixer has a small lightning bolt icon. Clicking the lightning bolt icon to ‘on’ switches the channel and and any instantiated AAX plugins over to Carbon’s DSP mode for latency-free monitoring, then switching it off flips the channel and plugins back into native CPU processing. It couldn’t be any simpler. Absolutely no need to flip over to that third-party mixer app for latency-free monitoring. Pro Tools also takes care of any plugins that may be instantiated downstream that are associated with the DSP-assisted record channel, such as effects returns and subgroups. If there happens to be plugins on those returns and subgroups that can’t be converted to DSP-assisted processors, you can switch those channels to DSP safe mode – in which case they’ll remain in the native DSP domain. In some cases this may be perfectly acceptable, for example, a reverb send isn’t going to suffer from a few milliseconds of pre-delay on an auxiliary track. It really is the best of both worlds. Super low latency when you need it, alongside a plethora of native processing and seamless switching between each.

CARBON LOADING

As for the mechanicals of Carbon, there’s a bunch going on. The single rack unit Carbon houses eight of Avid’s 126dB (A-weighted) THD+N 105dB (0.00056%) 20Hz–20kHz mic preamps using combo Amphenol (!) connectors. Spec-wise these are decidedly respectable and are in a similar ballpark to the Pro Tools | HD Omni mic preamps. What’s unique to the Carbon mic pre lineup is that preamps five to eight offer variable impedance switching, as do the two front panel instrument inputs. The mic preamps offer three input impedances (1kΩ, 5kΩ and 50kΩ) while the instrument inputs provide five options (32kΩ, 72kΩ, 90kΩ, 230kΩ and 1MΩ). Line level I/O is via DB25 connectors, and their 125dB (A-weighted) dynamic range surpasses the HD Omni’s line input specification of 118dB greatly. Next up are dedicated 6.5mm TRS jack monitoring outputs with 120dB (A-weighted) dynamic range and THD+N of -110dB (0.0003%) – again, virtually identical to the HD Omni specification give or take some inconsequential decimal places. An additional two monitor pairs can be fed from the output D-subs, and assigned within the Hardware Setup window within Pro Tools. These can be switched via dedicated buttons on the front panel. I’d have liked to see a digital coax output for some dedicated upmarket D-to-A monitoring, but I guess you could do this via the ADAT ports if you’re that keen. Bear in mind this is a production station, not a mastering rig. ADAT expansion follows the standard S/MUX layout, so your sample-rate I/O mileage will vary accordingly. BNC wordclock ports are available, then there’s two ‘NET’ ethernet ports. One for connection to the host computer, with the other I assume for expansion options in the future. AVB is, of course, an infinitely expandable platform – so expansion might include things like external fader controllers, additional mic preamps, bridging to Dante networks, or more. Have a look at what Presonus and MOTU are creating using the AVB protocol and you’ll see the potential.

Out front there are two instrument inputs, mic/instrument gain pots, buttons for impedance switching, phantom power, polarity invert, eight I/O LED bar-graph meters, left/right monitoring meters, monitor switching and level control, mute and dimming buttons, and (wait for it) four headphone outputs. Four! Enough for your smaller ensembles. These are rated at 100mW into 300Ω and 200mW into 32Ω – so plenty loud enough.

SOFTWARE

Software supplied includes a one-year subscription to Pro Tools with a perpetual license and 116 AAX plugins (including about 70 AAX DSP plugins). These include UVI Falcon, Celemony Melodyne 5 Essential, the Avid Complete Plugin Bundle, and HEAT (ya gotta love HEAT!), three free months of Antares Auto-Tune Unlimited, including Auto-Tune Hybrid, and a 5.4GB sound library. Plus you get Arturia Rev PLATE-140, Purple Audio MC77, Embody Immerse Virtual Studio, McDSP 6050 Ultimate Channel Strip HD, Native Instruments Vintage Organs, UVI Model D, along with Brainworx bx_console N, bx–rockrack and bx_masterdesk.

CONCLUSION

Overall, it’s a killer package to my mind. Until now there’s been nothing like Carbon, and certainly not at this price. I must admit I’ve been a little spoilt over the years as I’ve predominantly owned and worked with Digidesign HD and TDM DSP assisted systems, so on the few occasions that I’ve been forced to record in native systems I’ve been left disappointed with the direct monitoring fiasco. For me it’s a big fat horse-fly in the ointment – I don’t know how you guys with native systems put up with it. So if your DAW of choice is Pro Tools and you’re still in the Apple world, six grand is a no-brainer compared with a base level HDX system at 14 grand. Everything’s built into the one box: monitoring, mic pre’s, instrument inputs, headphone outs, the fabulously creative option to play with impedance loadings, and no latency issues ever. Plus you can run this from a qualified MacBook; no aging cheese-grater or ten grand Mac Pro tower required. It’s the best thing since sliced bread – go get one.

ON THE RECORD

Firing up Presonus’ own Studio One DAW, I plugged in a guitar and a mic, to lay down a track. It was a remarkably intuitive process. Studio One steadily wins more friends and part of its appeal is its deep integration with companion Presonus hardware.

There’s something quite refreshing about the layout of the controls. The desktop format means that key controls, such as Gain, are easily to hand… much more like an actual mixing console —  a godsend when you’re huddled behind a boom stand. Switching between mix channels is a breeze. I only wish there was a button to toggle between the Mix and Session/Timeline windows in the DAW.

If ioStation is your introduction to recording, then you’re in safe hands. Studio One is a pro piece of software and is especially suited to singer/songwriter producers. Features, such as the pre-built FX chain presets (for example, ‘Basic Acoustic’) are a welcome ‘cheat code’ for beginners and more experienced producers alike. You are immediately in the ‘ballpark’, with a polished sound. With a little tweaking, a respectable Soundcloud-worthy mix is achievable, with the onboard software doing most of the legwork, leaving just the muso’s capable ears to decide on what to tweak, and which presets sound best.

ioStation will happily converse with other DAWs via the venerable HUI protocol but if you’re still open-minded about which DAW horse to back then I’d encourage you take StudioOne for a spin.

AUTOMATION STATION

Gen-Z has been born into a world of automation but riding actual faders remains a foreign concept.

Riding virtual faders is not the same. In fact, a mouse or a touchscreen discourage such activity and the resulting ‘set/forget’ mixes are the worse for it.

As an introduction to hands-on mixing with automation, Studio One in combination with the ioStation is a marriage made in heaven. It was dead easy to ride my acoustic track throughout the song to balance it against the vocal, and then to add in some ‘Touch’ button edits where the level needed a bump. The experience left me hungry to mix some more.

Those requiring more ins and outs will look elsewhere, and those producing electronic music will probably have already bailed out of this review. But if you’re a solo artist looking to do some recording and to up your mix game, ioStation24c (and the included Studio One) will be a cost-effective and potent weapon in your studio.

And why wouldn’t you? It’s built beautifully. Everything about the Cobalt 8 feels solid, just like the previous Argon 8, which has the same form factor.

I enjoyed my experience playing live with the Cobalt 8. The single most important feature of this synth, in this respect, is the ability to store presets to one of the eight buttons just above the keyboard manual on the left. There’s no other way of quickly navigating to a preset — otherwise you simply have to scroll, scroll, scroll, using the encoder to find the sound you’re after. 

The Fatar keybed plays well and the aftertouch is satisfying. The four-axis joystick, looks incredible but for performance I would have preferred a bog standard mod wheel that stays in the position you left it rather than snapping back into neutral — this is not a monosynth, after all, and with eight-note polyphony, two hands on the ‘steering wheel’ isn’t uncommon.

Another feather in the Cobalt 8 cap is MPE support (where you can apply polyphonic control and expression to individual notes), which will please performers with an MPE-compatible master keyboard. 

THE SOUNDS

Cobalt 8 sounds great. Just before I had to return the unit, I downloaded 100 free sounds from Baltica, which confirmed my sense of what a master programmer is capable of with this synth. Everything sounds… produced… easy to mix, it sits well… I’m not saying the sounds are predictable or anodyne — Cobalt 8 has plenty of surprises up its sleeve — but the building-block algorithms just seem so solid, well designed and selected.

My main gripe feels a bit churlish, but I really wish the sounds were categorised. When you scroll through the presets it really is like Forrest Gump’s box of chocolates — you don’t know what you’re going to get next.

If I was a Cobalt 8 owner, in for the long haul, I’d spend more time in the app and get my patches organised — basses, sequences, leads etc. The app is another Modal Electronics ace in the hole — a real asset for sound designers.

WORKING WITH COBALT

Working in Cobalt 8 is satisfying. There’s always a modifier that helps to take your sound places. The 4-pole morphable ladder filter with resonance takes centre stage in the layout. The little monochrome OLED screen is sufficient in almost all cases and a little frustrating on the odd occasion. The plethora of buttons and encoders is welcome. Still, it’s remarkable how often I was accessing ‘shift key’ secondary functions, but this isn’t a complaint. There’s logic and consideration in the layout of the UI.

The sequencer is another welcome inclusion as is the arpeggiator, which is a cut above your regular ‘up/down/random’ and ‘octave 1, 2, 3’ variety.

MODAL ORDER

Modal’s snappy tag is ‘beyond analogue’ and I’m becoming a convert. As much as I’m seduced by the mystique and cachet of true analogue with its attendant names like Dave Smith, Bob Moog, Tom Oberheim and others, that we invoke reverentially like high priests of a religious order, in truth, I want up-to-date MIDI specs, playability, reliability, ease of editing, presets, and regularly updated software. Modal Electronics does all this in addition to dishing up sounds that are modern, snappy, have depth and are easy to make your own. 

CONNECTED

One thing I love about Wireless Go II is it’ll plug into pretty much anything. In a short time I’d used Wireless Go II to record into a smartphone, computer, and a camera, while using the internal microphone, a Sennheiser lav, and a Røde shotgun as the source.

The receiver has both USB-C and 3.5mm TRS outputs. The USB-C connector allows it to interface with all manner of digital devices: iPhones or iPads with the SC15 Lightning cable, Android smartphones with the SC16 USB-C cable, or even straight into a computer with the included USB-C to USB-A cable. Plugging the receiver into my MacBook instantly created a new input in System Preferences which can be selected as a recording source in a DAW or QuickTime. 

Wireless Go II’s ability to transform into a tiny audio interface sets it apart from the pack. Now you can easily record two wireless channels straight into Pro Tools with no power cables, XLR leads or other paraphernalia. It’s ideal for content creators in a studio-based multi-cam setup to capture audio in your computer and sync it up later in your NLE. Or it makes a quick podcast recording a breeze.

Alternatively you can bypass the digital interfacing and use the receiver’s TRS output into any device that’ll accept analogue inputs. But the upside to using the USB-C output is it frees up the analogue TRS output as a realtime headphone monitor.

While the Wireless Go II’s built-in omnidirectional mic sounds fine, don’t forget the 3.5mm input jack makes any audio source with a line output fair game for the Wireless Go II system, like a nice lavalier or headset mic. I even plugged my Røde VideoMic NTG straight in with a 3.5mm TRS cable for an instant wireless shotgun rig. Interview audio has never been easier!

COMMAND CENTRAL

To unlock its full array of features, Wireless Go II needs to be paired up with Røde Central software – a free download off Røde’s website. Connecting the receiver offers you various extended functionality including a 10-stage pad for finer gain setting (in 3dB increments), easy firmware updates, and a mute lock. The Ø button function can be toggled from a backlight on/off switch to a marker button, which will drop a marker on the internal recordings whenever it’s pressed.

HIT RECORD

Internal recording happens pre-airwaves right in the transmitters themselves. The feature is activated inside the Røde Central companion app – there’s no way to do this on the units themselves. Once activated, the transmitters will begin recording the moment they form a link with the receiver.

Depending on your needs, recordings can be in a compressed format (around 40 hours) or uncompressed WAV 24-bit/48kHz (around 7 hours). All your recordings show up as a list in Røde Central where each can be previewed and scrubbed. Signal dropouts are signposted with markers, as are marker points from the receiver’s Ø button. Recordings are time stamped which many editors will greatly appreciate. You can export a full recording or just a selection, with options of WAV/MP3, 44.1/48kHz, and 24-bit/32-bit floating point. 

At time of writing, Røde released a firmware update which allows a WG II transmitter to record uncompressed audio without being connected to the receiver. In other words, a transmitter can also be used as a standalone recorder, meaning you don’t have to worry about staying within a safe link distance. The applications for this are endless: from a personal dictaphone to a Zoom-like portable handy recorder, to an ‘attach to anything’ audio capture device on set.

MAKE IT TWO

To state the obvious, the standout distinguisher of Wireless Go II is how effortless it makes dual-channel recording in a single camera setup. Forget camera-top mics and the background noise they introduce, or field recorders and their bulkiness; Wireless Go II provides all the benefits of clean, crisp spoken word, times two. One can only imagine the appeal this has to doco filmmakers, wedding videographers, vloggers, journos and podcasters.

A word of advice, though. When it comes to multi-mic setups, don’t forget the 3:1 rule – the distance between microphones should be at least three times the distance from the microphone to the source. Not adhering to this usually results in all manner of phasing weirdness, as audio bleeds in and out of both mics at slightly different times. Facing the mics away from each other helps lessen the effect, as does using cardioid lav or lapel mics, but still do your best to keep a fair distance between them when recording with both transmitters or you’ll end up with the painful task of gating or automating levels for each one in post.

One minor operational complaint is when using the 3.5mm TRS jack as a headphone output. If I’m recording two channels in Split mode, the headphone output reflects the USB-C output by hard panning the channels. Naturally it can feel a bit odd monitoring with a different voice in each ear and a welcome feature would be the ability to collapse the headphone feed to mono while splitting the channels on the USB-C output.

All up, my experience reviewing the Røde Wireless Go II was filled with pleasant surprises and regular exclamations of “wow, that was easy”. Not everyone has time for complicated wireless mic systems with scrollable menus and a hundred and one frequency options. For the run ’n’ gun types, Wireless Go II is for you. Turn it on and it just works, in various ways and with various devices, and now with two channels.