Don’t let the name fool you; Embody’s Immerse Virtual Studio plug-in is not a tool for creating immersive audio experiences. It is designed specifically to put you in the driver’s seat at some well-known studios to listen to your mixes in those rooms.
Immerse Virtual Studio sits on your master fader. When you’re done, you’ll either bypass or remove it before you export your mix. You get several studios to experience—Erik Reichers and Bob Horn’s Echo Bar, SAE’s Diamond Control Room, Warren Huart’s Spitfire Studio, and Carlos de la Garza’s Music Friends Studio. I have no first-hand experience with any of them, so I have no frame of reference as to how close they came…but each gets highlighted with photos and pages of information about their gear, engineers and artists that have used them. For this alone, the plug-in has educational value.
Embody has captured room responses to recreate the reflections and overall feel of each space. There seems to be a pretty wide range of styles in them, from what I consider a more traditional studio with soffited speakers to more personal spaces with speakers just piled up along the meter bridge and pretty close to each other. By selecting the headphones you’re using within the plug-in, the playing field is leveled via EQ, allowing the modeling of the speakers in each studio to translate more evenly. It takes some getting used to, and you’ll no doubt find your favorite space to be in.
Getting to work
To put the plug-in through its paces, I loaded up an old project that I’ve been aching to revisit. It was tracked on 2” tape on an SSL E Series with lots of external Neve mic pres, and later transferred to Pro Tools. It features a wide range of acoustic and electric instruments and vocals to test with. My hope was to close my eyes and be at that SSL again instead of my home studio.
When transitioning from a traditional headphone mix to this virtual room, there is a shift in how things are panned, moving to subtle and distant panning based on the position of the speakers in front of you. The acoustic vacuum of your headphones is given new life via replicated virtual acoustics (essentially a short reverb). While this helps to create the illusion, I found myself dialing down the ambience, especially when trying to listen for similar early reflections in the drum mics or even the lead vocal. Thankfully, Embody gives you control of that.
I want to pull back for a moment and manage expectations: You are not gaining a pair of Genelecs or Focals for your home studio. Instead, there is an EQ curve applied to represent the characteristics of those speakers in your headphones. I found the tonal differences when switching between virtual monitors a bit disconcerting at first, but got used to it. What really bothered me was trying to mix the lead vocal; the phase manipulation and other processing made it hard for me to cope. Admittedly annoyed and just not feeling it, I was beginning to dismiss the experience completely when I thought, “Let me bypass the thing and listen on my speakers for a minute.” And that’s when…
It worked! When I switched back to my real-world monitors, I had little expectation that anything would translate—but it did…and it did so beautifully! Generally, music is mixed in studios and translates well to headphones; this is the first time I had mixed in headphones and had it translate to the speakers.
There is a free trial available for Immerse Virtual Studio so you can poke around and explore on your own. It’s a nice way to hear a mix back on multiple speakers when you’re limited to one set. I would love to see them add some more ‘world-class’ studios to the mix, as well as a surround option. If you’re in a situation what you need to keep the noise down but want to continue working on a mix, this is a fun solve! Is it worth the price tag? You’ll have to decide for yourself.
While Minneapolis-based Alclair has been actively manufacturing its IEMs since 2010, the company has only recently gained the attention it deserves in pro-audio circles. No stranger to audiology, Alclair has been a primary player in that field for more than six decades, developing and manufacturing the material audiologists use to make ear impressions, and its Minneapolis retail shop also provides hearing aid fitting services. Alclair has a strong artist roster so I’ve been aware of the company for quite some time, but it was only with the release of its electrostatic driver-equipped ESM model that I knew I had to give its IEMs a try. A bit more research revealed that a handful of models are focused on studio mixing, which made me even more excited. After spending time at Alclair’s Nashville headquarters auditioning universal versions of their IEMs (a dozen models ranging from $349 to $2499) I opted to audition the ESM 13 and Studio4 IEMs for my review. Title
Alclair’s flagship ESM 13 ($2,499.00) incorporates 13 drivers and is the picture-perfect amalgamation of balanced armature and electrostatic drivers. The heart of the ESM is four proprietary balanced armature woofers, four balanced armature mid-range drivers, one balanced armature tweeter and four electrostatic drivers accompanied by a 4-way crossover. The four bore 30Ω IEMs include premium silver-plated copper cable, provide -26 dB of noise reduction and have a 110 dB SPL input sensitivity. Meanwhile, the Studio4 model ($949.00) incorporates four balanced armature drivers accompanied by a 3-way crossover. The three bore 32Ω IEMs provide -26 dB of noise reduction and have a 110 dB SPL Input Sensitivity.
To reduce distortion and increase clarity, Alclair employs a single tube and port for all of the drivers working in the same frequency range. This allows the sound to combine in your ear canal rather than the tubes making for a better resulting sound quality (this is true for all of Alclair’s IEM models).
All of the Alclair IEMs include a cleaning tool, ¼” adapter, and custom leather case by Haiti Made. It’s worth noting that besides being rugged and beautifully made, the cases support a noble cause as Haiti Made was born out of the desire to see the Haitian people (who typically live on less than $2.50/day) empowered by sustainable and dignified employment.
I’ve been living with the ESM 13 and STUDIO4 models for the past couple of months and during that time have been utilizing them daily. My critical evaluation listening was completed via Tidal, my streaming platform of choice, where I auditioned my staple reference albums, including Pink Floyd’s Dark Side of the Moon; Elton John’s Goodbye Yellow Brick Road; James Taylor’s Hourglass; The Beach Boys’ Pet Sounds; Fleetwood Mac’s Rumours, and Daft Punk’sRandom Access Memories. I also spent ample time mixing with both sets of IEMs as well as time referencing several of my own mixes from past projects. Both IEM models impressed me!
The electrostatic drivers in the ESM 13s work by applying a static electrical charge to a thin film floating between two perforated metal plates. As an audio signal is applied to the plates, the film membrane moves backward and forward because of electrical attraction and repulsion. To simplify, armature drivers work much like a dynamic microphone, while electrostatic drivers work like a condenser microphone. Electrostatic drivers are exceptionally fast, making them perfect for tweeters; the purpose of the electrostatic drivers in the ESMs is to emphasize the detail of the audio signal and add to the imaging.
The ESM 13s are a pleasure to listen to. Although they have a slight top-end and bottom-end boost, it’s just enough to make them fun without feeling overly hyped in those areas. The soundfield of the IEMs is impressively wide, laying out the perfect sonic space for the precise placement of every mix element to be clearly identified. The ESM 13’s electrostatic drivers provide amazing detail, allowing the most subtle mix elements to be heard. This was especially noticeable when listening to Hourglass, as I heard reverb trails on this album sink far deeper into the mix than I had ever noticed previously, and I’ve spent a lot of time with that album. The bottom end is full, tight and punchy. On some tracks, there is a perception of a slight bass boost, but never to the point of being overwhelming (drummers and bass players typically enjoy this type of performance in an IEM). The mid-range clarity is smooth, and the top-end is detailed and crystal clear. The headroom on the ESM13 seems nearly uncapped and there is no perceivable distortion, even at extremely loud listening levels.
Although perfectly suited for stage, the Studio4 is ideal for studio work. Think of it as a precision, uncolored pair of high-end studio monitors. Listening to the Studio4s is the closest I’ve felt to having ATC monitors in my ears. In my experience, it’s nearly impossible to mix an entire project solely with IEMs, but once I get 10% of the way into a mix, I’m completely fine moving to Studio4s and staying there until I’m ready for my final tweaking. With more and more people doing serious studio work in their homes, a flat IEM is the best solution for musicians and engineers needing to isolate from their housemates.
The Studio4 provides a tight, punchy bass with mid-range clarity and a smooth, natural top-end. Like the ESM 13, it is extremely detailed throughout, but in contrast, there isn’t quite as much headroom and there is no slight top or bottom boost (They’re not quite as fun to listen to but they are accurate as hell).
On an entirely different note, I’m a big vinyl fan and historically I haven’t been fond of listening to vinyl with IEMs. It has just never translated in a musical way, as any clicks or pops sucked me right out of the listening experience. The natural sound and flat response of the Studio4 has changed this completely. It is the ultimate IEM for vinyl listening and I can finally enjoy my vinyl collection when my wife and kids are sound asleep.
While all of the Alclair IEM’s have their place and purpose, the Electro 6 Driver Electrostatic is the perfect blend between the Studio4 Quad and ESM 13 models that I evaluated. Users attracted to Electrostatic drivers but lacking the funds for the ESM 13’s price-tag should give the $1,499 Electro 6 Driver Electrostatic consideration.
Mastering is one of those corners of pro audio that everyone knows about, but doesn’t necessarily know what it truly entails. Shedding some light on the subject is Evren Göknar’s new book, Major Label Mastering: Professional Mastering Process (Focal Press/Routledge; $42.95), which breaks the topic down into understandable concepts and actionable steps that can be grasped by everyone, whether they’re students, musicians or fellow pros.
Göknar knows from whence he speaks—a Grammy winner, he’s worked more than 25 years in the mastering field, spending much of that time at Capitol Studios where he mastered everyone from Mariah Carey to the Beastie Boys in addition to putting his talents to work for TV shows like The Voice. During that time, Göknar developed the centerpiece of his book, The Five Step Mastering Process—a thorough system of considerations and procedures for crafting and implementing a mastering game plan will best serve the music. Readily acknowledging that there can be as many subjective assessments to be made (“Does this approach fit the genre?”) as there are technical ones, Göknar finds ways to help readers determine what’s necessary and bring quantifiable logic to more nebulous parts of the process.
That said, there’s plenty of straight-forward ‘how-to’ content, from best practices for documentation, to equipment sequencing, to a go-to section on advanced mastering chain tools and techniques. The book is also filled with cool gear photos, informative screenshots, useful illustrations, documentation examples and more, providing additional clarity and insight. Whether a budding engineer or a seasoned pro, readers will come away from Major Label Mastering with far greater understanding and appreciation for the newly demystified process of mastering.
I’ve been a fan of JZ mics for some time now, having reviewed their Amethyst, Black Hole, Vintage series (V47, 67, 11, 12) and their small-diaphragm BT201s too, all with more than satisfactory results, all with plenty of color (except for the neutral Black Hole) and all with plenty of design style. The new BB29 from the JZ Signature Series looks to combine the overall structure of the Black Hole with the personality of the Vintage series, except sporting an output transformer—a first for JZ.
Out of the Box
The BB29 is a single-diaphragm pressure-gradient condenser microphone, utilizing JZ’s patented Golden Drop diaphragm-coating technology on a 1” membrane, with a Class-A discrete hand-built circuit. The mic is shaped like a rectangular cartridge, with a flat/squarish head basket—very much like the Black Hole except the hole area is solid. An elastic suspension shock mount ($99) grabs the BB29 via four grommets, holding the mic firmly and allowing close placement to things like speaker cabinet grills.
The provided frequency response graph shows 20 to 20k response, with a little 500 Hz bump and a gradual presence rise punctuated with a high-end boost of five dB that’s not rolling off until 15k. Output impedance is 150 ohms (1000 ohm ideal load), with a max SPL handling of 140 dB and A-rated self-noise of only 9 dB. There are no pads or switches on the cardioid-only mic, which carries a substantial five-year warranty, for $1,299 direct.
Being familiar with JZ mics, I threw the BB29 right in there as a third drum kit overhead, a “lower-head” about two feet off the floor and six feet out, via Manley TNT preamp, hoping to get a fairly balanced middle-of-the-kit image with plenty of cymbal; that’s exactly what I got. I could hear the room “air,” got largely uncompressed dynamics and a nice (mono) hi-fi picture.
The next day, I put the BB29 up there with two other mics in order to pick two favorites for a super-powerful alto female vocalist tracking a full-length. With amplification from an Avalon VT737 channel, my singer didn’t prefer the BB29 (I think the high-end emphasis bothered her), so I rolled off some top and thinned out the very-bottom, and we used it for most of the backing vocals.
Next, I had a drum session with a fairly light-handed drummer where my close mics and room ribbon needed some help in both punch and excitement. I placed the BB29 much like I did the first time but amped it up with an SSL VHD preamp and dialed in moderate levels of 2nd and 3rd order harmonics, followed by compression via ART VLA-modified with low-end enhancing Carnhill transformers. The result was a much bigger-than-life sound with gutsy punch, a fancy sheen and more raw power than my overheads or room ribbon.
I often use a JZ V12 on electric guitar, where its C12-like qualities make for one nicely articulated guitar sound, so I just had to compare the BB29 directly with a pair of AMS-Neve 4081 mic preamps which are super consistent and rather neutral. I jumped around various tones, but long story short, the BB29 seemed to have a little more up top, had flatter mids than the sculpted V12 and a little more bottom. Definitely a bright sound overall, but really nice and no trouble at all with SPL, even though the BB29 doesn’t have a pad.
I got good results with tambourine, shaker and handclaps with the BB29, too, even if the resulting tone was more crisp than full. The high-end emphasis was substantial but not overbearing, and the high-frequencies were clean, not at all distorted or crunchy.
Acoustic guitar—amped up with a Cranborne Audio Camden 500, chosen for a lack of color and consistent linearity—found the BB29 putting out a well-balanced picture if going a little heavy on very high-end, but in a way that most of you might just prefer (depends on the exact situation really). My Yamaha upright piano didn’t play well with the BB29 though, as the hammers hitting the strings created prominent clicks that just weren’t musical, lid open or closed. My personal vocal tests were revealing, too; the off-axis response was forgiving enough to allow a little roll to the side to ease off that very-high-end definition bump (I’d like to try that with the alto vocalist now) and the proximity effect was audible at 5”, substantial at 3” to 4” out and a bit overbearing any closer than that.
The Final Mix
All in all, the BB29 is another high-quality, colorful and high-fidelity mic from JZ; it looks like the company’s Black Hole, sounds more like the Vintage series but with the extra emphasis both down-low and up-high from a transformer (with a touch of low-end saturation, too). Oddly enough, my most universal comparison would be a bright vintage U67 in cardioid (they all sound a little different it seems), with ample midrange (that little 500 Hz bump?) and sculpted euphony at each end of the spectrum. As with most colorful offerings, the BB29 won’t be a perfect fit for everything and its simplicity limits the range of applications, but when it does fit, you’ll find it exceptional.
New York, NY (April 30, 2021)—The SPL Marc One Monitoring and Recording Controller is a beast, providing high-end quality whether you’re listening back or putting down tracks. It offers three monitoring modes, a 32-bit AD/DA converter and a smart, user-friendly design.
For this review, we connected it to our Kali IN-8s and Neumann KH 120s studio monitors via USB to the DAW, but it was impressing us before we even turned it on. First of all, its heavy-duty metal frame body and actual weight give you a hint of the quality you’re about to hear. All knobs and toggle switches are weighted, supporting the impression of a high-end product.
The first toggle switch on the left side of the device controls the two sets of studio monitors and has an off position in the middle. Position A on the toggle includes the subwoofer if you have one, while the off position is silent. There’s no bleed into your monitors in any switch position.
Next is the volume knob. It takes up a lot of real estate but it’s ultimately the reason I love this controller. It provides lots of ways to tweak your outputs that we will get into later. You’ll use this knob the most, and the even volume response sounds amazing.
The second toggle switch gives you the option to listen in mono, stereo or channel swap, which reverses the stereo image so you can hear everything that’s going on with your mix. There are light indicators for left and right monitor distortion to help protect your monitors.
When the monitor knob is centered, all input signals are equally loud; turn to the left and the analog stereo inputs get louder while the USB input signal gets quieter. Turning from the center to the right does the opfeposite, all helping to tweak your system to deliver the perfect volume for mixing. There’s also a headphone volume knob and a cross-feed knob so you can blind-hear the studio monitors and headphones together for more listening options. The amount of depth from the headphone amp to our studio headphones, Sennheiser HD 800s, opened my ears to new things in our mixes.
Now to the back of the unit. First is a heavy-duty on/off switch that looks like it can stand up to years of use. Next to it are two dip switches: 1 provides a 10 dB pad on your studio monitor sources, while 2 is a Rec 1+2, which mixes together both line inputs and allows you to record them together via USB in mono. Both dip switches are easy to flip without requiring a tool or a long fingernail to use.
All of the TS and TRS inputs and outputs on the back of the unit stick out for better grounding. The USB connection is class-compliant, which means that all Mac computers and iOS devices like iPads and iPhones can use the full performance bandwidth of the 32-bit AD/DA converter without driver installation. (iOS devices will need the camera adapter to connect, however.) For Windows, you’ll only have to install a driver (downloadable on the SPL site) if you need higher sample rates. The USB connection will connect to your DAW for superb studio monitor playback at very high resolutions— for example, DSD4 and DSD256 (11.2 MHz) are supported.
Over the course of our review period, we found the amazing resolution and high-quality build combined perfectly in use as we toggled seamlessly between studio monitors and headphones to listen to mixes. Used as a monitor controller, SPL got the layout perfect for muscle-memory mixing, with the monitor toggle switch and volume knob ideally located for quick adjustments while listening. When you consider that it’s also an interface for recording, able to record high-resolution audio from 10 Hz to 200 kHz, the $799 price tag is worth it. The SPL Marc One is now a staple in our studio and workflow.
Fela Davis is a 2019 Hall of Fame inductee at Full Sail University. She owns 23dB Productions and One of One Productions Studio, which specializes in podcasting, video, and music production. Clients include the Holding Court with Eboni K. Williams podcast, SiriusXM, Atlantic Records, iHeartRadio and numerous Grammy Award-winning musicians. www.oneofoneproductions.com.
How do you capture the essence of a legendary engineer/producer with 23 Grammys, 160 gold and platinum albums and a ‘who’s who’ resume, and put that into a piece of software? Well, that’s just what Leapwig and the iconic Al Schmitt went for with the new Al Schmitt signature plug-in. The team literally encapsulated his gear, mixes’ textures and workflow to come up with something that ambitious.
When first opening the plug-in, you select from a Source dropdown menu that offers up Vocal, Bass, Brass, Mix, Piano or Strings. These are referred to as ‘profiles’ and each of them are tuned differently with their own character and tone. Each profile also features a different ‘tuned’ amount of harmonic distortion. Within each profile, there are a number of options as well—for example, Vocal features Body Level, Air Level, Echo Level, Compression, Air Type and Echo Type.
This approach to plug-in design has led Al and the team at Leapwig to create something that operates in a unique fashion. When audio is played, rings that represent loudness are played around the relevant icon in the center in real time. If there is something like gain reduction happening, the outer rings tighten up accordingly at ½ dB per ring. For instance, if there are four rings happening, you’ve got 2 dBs of reduction. It’s something your eyes have to get used to because it’s simply a new way of operating.
Since each source features its own customized parameters to tweak, you quickly adjust to how to get around. For example, Mix features Sub Boost, Low, Mid and High Level, Low, Mid and High Comp, a compressor link and Air Boost. Bass is nothing but Compression, Body Level and Air Level, but it includes additional harmonic distortion within those parameters. Piano, which is one of my favorites, features Compression, Echo Level and A/B/C Echo Type. Note that “echo” is actually a reverb, a name that was chosen since that is what Al calls it. Aside from that, there’s In and Out Meters with up to 12 dBs of gain.
What I like about this plug-in is that you can dial in some taste very quickly. When first listening, it helps to run through each source to understand what the parameters do. To hear the echoes clearly, I would simply put on audio with attack and stop the transport, listening to what sound is created afterwards. The others, such as Sub, Air bands and EQs are easy to hear. Compression is subtle yet clearly audible. I found it useful to also mix and match—for example, using the compression in the bass source on something like a piano. Then, if I wanted more, I put another instance after in the DAW and used the EQ in Echo in the Strings source, or the EQ and Air settings in the Mix source. Once you have a feel for it, your instinct knows where to go. I saved a number of presets for easy recall: I like Echo Type C on the Strings source, so that’s now my “RT Echo 1 Strings’ preset, and I also captured a nice Mix bus preset with Sub Boost, Air Boost a few dBs of Gain and a touch of Highs as “AS Master 1.”
Aside from being easy, this plug is fun to use. You can get to a sound with just a few quick fader slides and most importantly, it works as advertised. It’s not big, bold and aggressive, but subtle and tasty, especially in the reverb/echo fields. Most importantly, all of these sounds are clean, clear and tasty. I would also use the word “refined,” which is a testament to the team making it. Since you probably can’t get him to your session, now you can bring a little of Al Schmitt’s magic sonic touch to your own tracks.
My favorite part of tracking a band is matching my collection of mic pres to my carefully curated mics for euphonic results, and 500-series mic amps have made this fascination more affordable and convenient. The newest addition to my 500 rig is the VHD Pre from SSL and it has affordably given me that classic SSL sound, along with a number of creative options.
Out of the Box
Occupying only one 500-slot, the VHD Pre packs in plenty of features without feeling too cramped or crowded. It starts with a gain control ranging from +20 to a whopping +75 dB of gain—enough to amplify quiet sources and passive ribbons. There is an input pad of -20 dB, which is enough to accept hot mics and line level sources, as well as a defeatable -18 dB/octave high-pass filter that ranges from a nearly-subsonic 15 Hz up to a truly-midrange 500 Hz for anything from rumble removal to a complete removal of all bass.
You’ll find the requisite phantom power and polarity switches, but most important is SSL’s VHD (Variable Harmonic Drive) circuit as taken from the Duality line of consoles (which actually have two sets of preamps—VHD and SuperAnalogue, hence Duality). A switch engages the circuitry, and the Drive control allows 2nd-order, 3rd-order, or a blend of both, harmonics.
There is a 1/4” direct input for electronic instruments, along with a Hi-Z switch that changes input impedance from 1.2 K ohms to 10 K ohms for tonal and sensitivity variability. A single tri-color LED indicates signal presence with the familiar green/yellow/red scheme. Finally, an output trim control (ranges from -20 to +20 dB) is provided for dialing back level on all that high-passed VHD-goosed signal you’ve created.
It should come as no surprise that the VHD Pre has a clean, largely neutral sound that is high on headroom, wide in frequency response and particularly sweetly defined in the top-end—the classic SSL sound if you will. Some call it “glass,” others call it “pure clean gain,” some call it “sterile,” and a few call it “thin.” I call it time-tested, versatile and familiar. On vocals, acoustic guitars, classical instruments and the like, the VHD Pre strikes a chord you’ve heard a million times and can immediately recognize. I’m not saying that this SSL tone is shockingly different, just simply that the numerous subtleties add up to create something very familiar and even nostalgic.
Let’s quickly cover the FAQs before getting to the VHD details: The gain is clean all the way to +75 (there is no sudden jump in noise or distortion at end of travel); the HPF is accurate, smooth and musical (ranging up to 500 Hz is brilliant for committing to wildly filtered sounds); the switchable impedance is a “must have” if you like to tweak your tones on the way-in without EQ (10 Kohms gives you more of that SSL air); and the output trim is essential to have for precise level setting and attenuating an overdriven circuit.
The VHD section is a little tricky, and solving its mysteries is the key-to-the-kingdom. VHD is simply on or off; you can’t select the “amount” of it (not directly, at least), although the knob controls the blend of second- and third-order harmonics. Fully second-order multiples yield a warmer, congealing tone that is dark-ish, smooth, sort of scratchy and “tubey,” while fully third-order is a bit crispier, more sizzly, fuzzy and more transistor-like. Both are quite subtle without a lot of signal, although rather useful for their subtlety, especially when blended.
If you hone-in on today’s pop music, you’ll notice subtle saturation on almost any kind of track—vocals, keys, basses, drums, even handclaps—and the VHD Pre delivers those tones all day long. I found VHD often working best at juicing up detail and immediacy without being obvious or even noticeable (until bypassed, at least). Unruly tambourines, anemic vocals, boring bass, stock synths and “meh” guitars all take the heat well. I almost always had a 60/40 or a 40/60 harmonic blend, and you might be surprised how often 3rd-order is useful.
With ample gain, VHD jumps into distortion and you have to contain it deliberately for musical results. The key here is to carefully balance input gain with output trim, driving the input just hard enough to get the dirt/grit that you do want, re-balancing the odds and evens in the VHD, tweaking the HPF (maybe even the impedance) and then attenuating output until you’ve got that elusively desirable gritty growl that is manageable and sounds cool. Don’t be surprised if you find heavy VHD best combined with clean signal in parallel. All things considered, VHD really does the trick for harmonic dusting and moderate grit, but heavy distortion and manglings are more hit or miss. Basses and electronic drums? Oh yeah! Vocals? Not so much.
The Final Mix
At $579 (street), I couldn’t resist getting a VHD Pre for that widely popular vocal sound it so easily achieves. The versatility of the HPF and the variable impedance have made it an easy pre to plug-in when I’m not sure what a client is going to deliver. And now that I’ve mastered the use of this VHD section, it looks like I’ll be needing another one, so my stereo keys and other dual-input sources can get the benefit of a little SSL harmonic massaging, too.
As the debut product from Tula Mics, the appropriately named Tula Microphone is pretty unique, and not just because of its lustrous exterior. Instead of being another pocket-sized recorder that can double as a USB mic, the Tula is a pocket-sized USB mic that can double as a recorder. That may sound like splitting hairs, but it’s indicative of where the mic and the company behind it are coming from, rethinking the familiar from a different vantage point.
Roughly the size of a deck of cards, the Tula Mic is a stylish prosumer microphone designed for use in podcasting, content creation, the work-from-home world and so on, and it has a price tag to match at $199. Housed in the Tula’s solid metal/plastic case—available in black, red and cream—are cardioid and omnidirectional capsules, Burr Brown op amps, a Texas Instruments audio codec and a custom iteration of Swedish music software company Klevgrand’s Brusfri noise reduction algorithm. The mic connects to computers and devices via a USB-C port on back, and comes packaged with a USB-C to USB-A cable, a built-in (but removable) stand, and a universal threaded mic stand adaptor.
For those who use the Tula as a recorder, there’s 8 GB of internal memory (no SD or MicroSD cards here) which can hold up to 14 hours of recordings captured in .WAV format. When used on its own without a computer, the Tula is powered by a rechargable internal 3.7 V 700 mAh lithium ion battery that can hold enough power to record continuously for 10-12 hours with noise cancelation on, and 14 hours without. The Tula recharges via the USB-C cable, and when plugged into a computer, it appears on the desktop as a USB drive, allowing users to copy audio files to their machine.
Sporting a retro-futuristic look that vaguely recalls the Star Trek communicators of yore, the Tula has a minimalist design that underscores the usually intuitive controls on the mic. Aside from the detachable built-in stand, there are no moving parts on the Tula. All the control buttons run up each side of the mic and are under pressable mesh; notably, there is no screen on the Tula to convey information like settings, gain and so on, so crucial info is instead provided through two LEDs on the front face. Thanks to that minimalism, the mic may have a timeless look but there’s also far fewer parts to potentially break—a crucial factor for a mic that is likely to get tossed in backpacks and the like.
When used strictly as a USB mic, the Tula is pretty straightforward; it gets power from the USB-C cable in the back, but still requires the user to hit the On/Off button to activate it. The Tula defaults to the cardioid capsule, but a short tap of the Mic Select button switches to the Omni, and a long tap activates the Tula’s 3.5 mm lav mic input, which doubles as a headphone jack for playback.
Perhaps the Tula’s strongest selling point is its noise cancellation, because the onboard Klevgrand Brusfri algorithm gets the job done. In testing, I unfairly placed the Tula just six inches from a loud space-heater blasting right at the mic, started talking and hit the NC button halfway through recording. Upon playback, I found the algorithm had ripped that noisy heater out of the recording, leaving my voice very clear and usable, if unsurprisingly missing some low end. Lest that scare you off using the NC button, don’t fret; the Tula automatically records two copies of your audio file—one with noise cancellation and one without—so that you have options come edit time. Still, the noise cancellation is a real problem-solver, if not a miracle worker. It’s not supposed to offer the scalpel-like precision of your favorite audio repair plug-in, but it does an impressive job on the fly of creating more than passable audio in less than ideal circumstances; in those instances, the Tula’s noise cancellation really shines.
Used as a stand-alone recorder, the Tula is slightly less impressive—it records well, but is somewhat hindered by the device’s sleek minimalism. Most of the buttons’ functions are relatively clear, labeled with familiar universal icons for ‘record,’ ‘stop’ and the like. Confusingly, however, there are two Playback Volume buttons and two Gain Level buttons, and both sets are labeled with identical +/- symbols. That aggravation aside, the Gain Level buttons work well (once you remember which are which); adjusting in 5 dB increments, they affect an LED light on the front that alternately flashes green, yellow and red to help gauge the right level.
In all, the Tula offers a unique sense of style and design for its intended audience of content-creators—a market where, once video comes into play, a mic’s looks can be as important as its sound. The cleverly designed controls can be a little too clever at times, but the surprisingly robust on-board noise cancellation is impressive and will come in handy, especially for users who take the Tula out into the real world. The Tula Mic marks a solid debut for its namesake company.
With hosts, guests and talent comfortably seated in your purpose-built and acoustically-treated isolation room, the discussions are easily recorded with minimal bleed, defined clarity and no intrusion or distraction from unwanted sounds and noise. Yeah right! The truth is that for most of us voice recordists producing podcasts, audio books and interviews our audio is polluted with unwanted sounds of numerous varieties that must be prevented or removed if we expect rapt attention to the content. It can be hard to stay focused on dialogue when competing voices, air conditioning, ground hums, noisy appliances, passing trucks, airplanes, sibilance, plosives, ticks, clicks and massive vortexes (actually breaths, amplified to ridiculous levels) are stealing our attention. Luckily today’s digital, analytical, often aided by machine-learning software programs (and certain hardware pieces) are capable of not only mitigating, but downright removing extraneous noise. I’ll be using iZotope Rx 8 Advanced premium software in my examples, but there a number of competing programs that accomplish the same goals, in sometimes similar manners.
Before we delve into fixes, some effort should be spent making sure we are capturing the best audio we can before applying processing. The cleanest and purest signal possible will ensure less severe processing, more successful processing and undetectability.
The Room Is Key
Start with a room that allows spacing between the persons, with each speaker in the null(s) of the other’s mic(s) and seated in a circular pattern for large groups, which encourages interaction, allows more visual communication and rejects mic bleed. Make sure the room is devoid of anything that creates sound, including fridges, air purifiers, computers, cell phones or even hummy DC adaptor “wall warts”. Seal up windows, cover them with blankets, seal up doorways with weather-proofing (whether an interior or exterior door), dampen air ducts, reduce unwanted room ambience with acoustic treatments (ie. foam/fiberglass/cloth absorbent panels) and place diffusors on the closest walls to scatter sound waves and stop the dull muddiness that results from too much absorptive treatment in a small space.
All speaking talent will need closed-back headphones that don’t leak sound excessively, a pop filter on their mic to reduce plosives and enough room to back-off of the mic when needed. It is wise to have water available in glasses (plastic bottles can be noisy), a notepad (to aid memory and reduce unwanted interjections) and clean cloths (to dampen sneezing/coughing or other personal disasters).
Some basic mixing technique may be corrective enough to reduce any noise problems to the inconsequential. The big three tools here are equalization, compression/limiting and automation.
Assuming you’ve captured clean mic signal, sometimes a little filtering is all we need for high fidelity. If you used a bright condenser mic realize that too much crispy, treble-y, high-end definition can be irritating (if nearly painful). If so, employ either a high-frequency shelf somewhere around 8 kHz (reducing/attenuating three or four dB, or to taste) or engage a low-sloped low-pass filter somewhere between 12 to 18 kHz to remove the really high stuff.
Conversely, much of the environmental noise causing us trouble is found down in the low frequencies (ie. heating/cooling rumble, passing jets, appliance rumble, foot steps, traffic noise, the mic placed too close etc.) so a high-pass filter is essential. Engaging one around 80 Hz will remove noise without reducing “chesty-ness” but don’t be afraid to filter up to almost 200 Hz if severe low-frequency noise is persistent. The cumulative positive effects across multiple mics can be amazingly effective!
Many trained speakers are quite talented at maintaining ideal levels; knowing how to “stage whisper” or “half yell” for maximum effect without issue. Most people are terrible at such skills, so try to capture voice with a touch of compression to smooth out levels as you record. If your minimal set-up doesn’t allow this consider getting a mic preamp and a compressor (or all in one), or the purchase of a modern recording interface that allows software-based compression as you record. Then, still compress the track liberally in your DAW and apply limiting (severe compression) when stray peaks are stubbornly popping into the red.
Perhaps most importantly, you should automate levels in key moments to maintain consistent thematic focus. Like riding a physical fader for volume, automation can truly feature the right persons at the right moments; I aim to reduce volume on any given track that is not currently in use (or is relegated to only “ok’s” and “uh-huh’s) by about -4 to -8 dB. This maintains a consistent “air” and “presence” even as speakers take turns, but can still focus attention where you want it. If your noise problems were minimal to start, with only occasional major disruptions (ie. sirens or sneezes), such automation might be enough to fore-go the use of any corrective software.
Software Handles the Rest
If problems still persist, there is more we could do with surgical editing, extreme EQ and extreme automation, but why dig that deep when intelligent software can do the job more quickly, with less effort and with likely better results.
The HVAC of summer and winter makes more noise than is acceptable to reach Amazon/Audible’s audio book technical standards, so I clean each voice track with iZotope Rx 8 Advanced using the Voice De-noise module. This process requires a brief sample of the noise problem, so I use few seconds of pre-roll audio for analysis. Once De-noise has “learned” both the noise and the vocal timbre it can neatly remove noise for the entirety of the track. I often find -12 dB of reduction (set for Dialogue and Gentle) to be sufficient to achieve noise-floor standards, but a second less-intense pass can be done for bad problems. Leave a few seconds of each track’s pre-roll noisy so you can clearly verify the improvement the processing has made and check for “liquid” artifacts (if so, you’ll need to undo and process less severely). Remember to clean each track with its own uniquely learned noise profile, as there can are significant differences from track to track.
Popping P plosives and prominent breaths are horrible when monitored via today’s subwoofers and earbuds, so make sure you’ve removed all of them … or should I say reduced them to proper levels, as P’s, B’s and W’s require a little burst of air to function. The De-plosive module sure is quick and effective. Simply highlight the offensive plosives only (they are usually easily seen, with a big ole wave of low-frequency energy in the waveform) and adjust Sensitivity and Strength parameters dependent on severity of the issue. A second pass can be done, but you’ll seldom need it. De-breath is similarly useful in that it does a great job of moderating the problem without the stark removing of breath(ing). Beats the heck out of severe editing and placing “room tone” in the gaps!
The opposite of plosives, excessive sibilance can make S’s, C’s and K’s sound like little knives in your ears and they only get worse with bad earbuds and distorted playback devices. Treble reduction won’t cut it and side-chained compression is too complicated, but many de-esser plug-in’s will quite effectively manage sibilance. However, since Rx8 has a De-ess module it’s faster to make the fixes there. You’ll likely only need adjustments to Threshold and Cut-Off Frequency to get great results. Like the De-plosive module, only process the moments with problems, not the whole track top-to-tail like with Voice De-noise.
If you just couldn’t get the room right, or recorded in a big reverby space, the Dialogue De-reverb module works miracles. More importantly, I have found that this module works quite well at reducing short ambiences that are far quicker than reverb tails, in places like bedrooms, offices and meeting rooms. Some expertise and experimenting will be in order here with adjustments to Reduction, Sensitivity and Ambience Preservation and stereo “linkage”. Fear not, with a little practice it works way better than it ought to.
De-hum and De-rustle are both effective at their respective eponymous functions and easy enough to use, but Mouth De-click deserves the MVP award. My bane are those nasty, irritating mouth clicks/snaps that permeate vocal tracks from under-hydrated talent, so I’m elated that these modules actually work! You won’t need to adjust the Sensitivity or Click Widening that much, but do expect to require two, even three passes, on the worst offenders.
I routinely use all of the above modules to achieve compliance, but sometimes situations call for even more aggressive processing. Spectral Repair allows the targeted removal and replacement of unwanted sonic events (closing doors, dropped objects, sneezes etc.) in accordance with the time domain and the frequency domain. That is, you can select a lasso tool and neatly draw around the problem noise using a waveform and a spectrograph to clearly see the issue, then either simply attenuate it or replace the problem with audio from before or after the disturbance. Once you’ve experimented with Strength and Bands you’ll find this easier than my descriptions and miraculous.
There are three other highly specialized modules that use machine-learning for the odd difficult tasks that you may rarely if ever, encounter. For multi-location productions or (especially) interviews where the locations sound distractingly different, Ambience Match is a life-saver. You’ll need to teach the track the ambience you desire from another track, but it’s worth the effort to reduce such issues. If you receive audio from another location that is digitally ruined (ie. lossy encoding or low sample rate) Spectral Recovery can actually replace lost high-end information intelligently. Finally, Dialogue Isolate can save really poor audio (surely not recorded by you!) that suffers too much background noise. Careful adjustments must be made to Sensitivity and Ambience Preservation and multiple passes may be required, but this was your last stop … if your voice audio isn’t clean enough after all this you might need to refocus on better tracking.
One For the Road
It takes a lot of effort to ensure voice audio good enough that the audio itself becomes a non-factor, invisible and not even thought of. It is only then that we can truly maximize the goal of enabling communication and conveying ideas. You know that you’ve run your sessions right and used iZotope Rx 8 Advanced to edit/process/clean properly when all anyone can talk about is the actual content itself.
At first, I wanted to be sure I had the basics covered with premium plug-ins for all my requisite EQ, dynamics and time-based effect work. Now that those bases are covered, all I seem to desire anymore are quirky, non-traditional and lightning-fast workflow enhancers—plug-ins like the new Parallel Aggressor from BABY Audio.
Out of the Box
Parallel Aggressor (PA) is basically a saturation/distortion plug-in that accomplishes parallel processing with two different processors, Spank and Heat.
If Spank’s processing sounds familiar, there’s a good reason, as it’s actually the sound of the classic dbx 163 single-slider, super-budget compressor as digitally virtualized by BABY Audio’s I Heart NY parallel-compression plug-in. The 163 was my very first compressor, so many moons ago…. (sigh.)
If Heat’s processing sounds familiar, you’re likely a long-in-the-tooth veteran of the not-so-good-old analog days when we would print stereo audio to cassette or VHS tape for their “high fidelity,” only to find that it distorted in a surprisingly juicy manner when overdriven; this sound is digitally virtualized by the sound of BABY Audio’s Super VHS plug-in.
PA has a fader and solo button for dry signal, Spank and Heat as well as Auto Gain, and an output level control to achieve balance and enable wise comparisons. Both Spank and Heat each have an intensity control that decides their tone even more than the four Style button controls per processor.
The Spank side has an Extra Punch button (adds some attack), Extra Smack (brings up some high-mids), Sidechain Filter (filters out some bottom from the detector) and Mono. The Heat side offers Extra Hot (adds a little more distortion), Tone (adds some mids), HP Filter (filters some bottom off the actual signal, not the side chain) and LP Filter (which can take some nasty high-end off of buzzy distortion).
Spank’s 163-inspired response ranges from a nice, warm congealing and smoothing, to a bold grabby and chesty thing (characterized by that trademark 163 attack), all the way to a pumpy, growly super-squash that is best suited to parallel uses for sure. Heat ranges from a slight increase in detail edge and definition to a sizzly solid-state-ish buzz, all the way to an over-the-top, full-on distortion that is quite recognizable as the tape-slammed-all-to-hell variety.
The metering for process intensity presented as an orange ring around each control; the amounts of each process are the big hemispheres far left and right.
There are a lot of different ways to use saturators in general and PA, with its dual-processors and abundant control, offers tons of flexibility beyond the norm, so pardon me if I only summarize operations. Suffice to say, you can go easy on both Spank and Heat intensity, and blend them in subtly for an increase in size, detail and apparent “quality” that is perfect for mixes without being at all self-evident. I used light Spank, with Extra Punch (optional) and Sidechain Filter mixed about -12 dB under my dry signal, and with very-light Heat (no filters) mixed at least -12 dB down if not more.
Sub-groups like drums and basses benefit from moderately aggressive use of both Spank and Heat, whereas guitars, keys and vocals require a little more restraint. I found myself typically using Extra Punch and Extra Smack for the aggression and bite, with Sidechain Filter and Mono both in to increase bottom-end response and avoid losing any thump. Extra Hot, Tone and HP Filter all helped increase detail and aggression on the Heat side, with the LP Filter proving essential at taking enough spiky aggression off to allow bolder usage. Guitars and vocals required the most sensitive touch, drums allow the most flexibility and the only question is which subgroups don’t get PA’d (don’t saturate everything in a mix, tempting though it may be).
Individual tracks allow the most fun with PA, especially the Heat side, which can be “too much” pretty easily. Vocals can take the Heat, as can guitar solos, gnarly bass monsters (of the stringed or synth variety), oversized synths and industrial-ized snares. It’s all great, addictive fun; just make sure to experiment with those Style buttons when listening to your tracks in the whole mix, not in solo. I liked just about all the options PA gave me when soloed, but the question is what works in the big picture.
The Final Mix
Parallel Aggressor offers plenty of versatility to avoid any “one trick pony” issues and still manages to be a “quick fix” with a limited number of powerful controls. There are a lot of effective plug-ins available today for saturation and distortion tasks, but few offer the convenient use of each, with a full complement of variables, in such a convenient package and full parallel mixing abilities on-board. At this price of $49, I can think of no other, actually.