Mixing – Analog or Digital?

A perennial topic among YouTube audio production channels is whether analog is better than digital and whether or not it even makes a difference. While I’m a video editor and not a mixer, the music projects that I have been involved with have all been recorded analog. Of course, in the past 20 years audio has been increasingly recorded and mixed purely in the digital realm. Although, sometimes analog pieces of gear were used for character and color.

Produce Like A Pro is a YouTube channel that I follow. Music producer Warren Huart frequently features videos by Grammy-nominated producer/engineer/mixer Marc Daniel Nelson. Many of these videos include downloadable session tracks that enable you to remix the songs in order to learn from the process.

I found this particular video (linked) of Nelson’s intriguing, because it tackled the analog/digital debate head-on. It’s from an older session of his in which he recorded and mixed the song “Traveling Light” by artist S. Joel Norman. As he explains in the video, most of the instrument tracks were “multed” – i.e. the mic signals were split and simultaneously recorded to 2″ analog multitrack tape, as well as directly into Pro Tools. Once the tape tracks were also ingested into Pro Tools, they could compare and pick whichever sounded the best. According to his commentary, the instrument tracks that were recorded to tape were preferred over those recorded directly to Pro Tools for this song. This is in keeping with the soul/gospel/RnB vibe of the song.

Doing my own remix

Since I like to mix some of these tunes (a hobby and to learn), I downloaded the tracks, dropped them into Logic Pro, and compared. As I first listened to the soloed tracks, the digital versions sounded better to me – louder and more open. My intent originally was to mix in Logic using mainly the built-in plug-ins. Unfortunately as I started to build the mix, I had trouble getting the right sound, especially with drums. Drums are often one of the hardest parts of the mix to get right. It’s usually the largest number of mics with the most leakage. Getting a drum kit to sound right and not like someone is pounding on cardboard boxes can take a mix engineer a lot of time.

I decided to change my approach and wherever possible, switch over to the tracks recorded to tape. Instantly the mix started to fall into line. This is a classic case of what sounds great in solo might not sound as good in combination with the rest of the mix. The whole is greater than the sum of the parts. This is why veteran mixers always caution beginners not to fixate too much on making each individual track sound perfect on its own.

Along with the decision to change my approach, I also abandoned the idea of doing the whole mix with Logic’s native plug-ins. Don’t get me wrong. The tools included with Logic Pro are quite good. Their compressor and vintage EQ options are designed to emulate certain models of sought-after, classic analog gear. They just don’t use the licensed branding. I did still use them, but more sparsely.

Tracks -> Stacks -> Submix -> Output

My standard track layout for these mixes is to combine each instrument group into a summing track stack (a bus) – drums, guitar, bass, keys, vocals, etc. I usually route all of these instrument stems (buses) to a submix bus, which in turn is sent to the output. This allows me to mix levels and add plug-ins/processing at three stages – the track, the track stack, and the final submix bus. I don’t add any processing to the output bus. Only metering plug-ins are applied there.

For this project, I decided to use a modified approach. All instrument stems were routed to a separate instruments bus (minus any vocals). Then the combination of instruments, vocals, and choir were routed to the submix bus. The advantage of this type of film/TV mixing style is that I could adjust all instruments as a group on a single channel and balance them as a unit against the vocals and choir.

In the past I used to rely on hardware faders, but I don’t own a control surface. I also used to write live automation passes with the mouse, but I’ve gone away from doing that, too. Instead, I surgically add and adjust keyframes throughout the individuals tracks, as well as the stems. Usually I will balance out the mix this way before ever adding plug-ins. Those are there to sweeten – not to do the heavy lifting.

Mixing with plug-ins and channel strips

My main effects tool for this mix was the Waves Scheps Omni Channel plug-in, which I applied to each track stack (instrument group). Andrew Scheps is a renowned mixer who has partnered with Waves to develop the Omni Channel. The advantage to a channel strip is that you have multiple effects tools (filters, compression, EQ, etc) at your fingertips all within a single interface. It mimics a channel strip on an analog console. No need to open multiple plug-in windows.

I also have both SSL and Focusrite channel strip plug-ins, but I prefer the Scheps version. Instead of simply designing just another SSL or Neve copy, Scheps was able to pick and choose the character of different products to create a channel strip that he would like to use himself. It sounds great, has a ton of presets, and unlike the name-brand emulations, the modules within the plug-in can be expanded and re-arranged. When applying it to instrument stacks, I can really develop the character that I want to hear.

No mix is ever finished after the first pass. When I compared my mix to the official mix that’s available on Spotify, I noticed some distinct differences. The artist’s version had some additional overdubbed instrumentation (strings and some embellishments) that I didn’t have in the download. They also chose to delay the start of the choir after the breakdown mid-song. These are all subjective choices based on taste. Of course, the release mix has also been professionally mastered, which can make a big difference.

What bothered me in my mix was the lack of a really present bottom end. This is often the difference in amateur versus pro mixes. A top-level mixer like Marc Daniel Nelson is certainly going to be way better at it than I am. In addition, he might be mixing in a hybrid fashion using Pro Tools along with key pieces of analog gear that really improve the sound and help to sculpt the sonic qualities of a song.

In an effort to increase and improve the bottom end, I decided to swap the kick drum tracks recorded to tape for the digital versions. I also dropped the bass amp track in favor of only using the bass DI track. The second thing was to use Logic’s vintage graphic EQ to boost the kick drum and bass low frequencies. This particular plug-in emulates an API console EQ and is a good choice for the low end. 

In the modern era, live drum sounds are often replaced by drum samples. The samples are triggered by the live drums, so you still get the right feel and timing, but a better drum sound. Often a mixer will combine a bit of both. I don’t know whether or not that was done in the actual mix. I’m certainly not implying that it was. Nevertheless, this is a fairly common modern practice to get really killer drum kit mixes.

Dealing with recording reality

When you start playing with raw tracks, it’s inevitable that you’re going to listen to each in the solo mode. You quickly see that even the best recordings will have some wrinkles. For example, I don’t like when a singer or a voice-over artist takes huge breaths between phrases.  At first, I tried to mitigate these with De-Breath plug-ins – first Accusonus and later iZotope RX. Both introduced some annoying artifacts that I could hear in the mix. So I decided on the old-school approach, simply adding keyframes and ducking the vocal track at each breath. In doing so – and paying very close attention to the vocal, I also realized that some sort of gate must have been used during the recording. You could hear a track drop to silence as a last word faded between phrases. Riding levels helped to smooth these out, too.

Working with the bass track, I also noticed some “fizz” in the 3khz range. This appeared to be coming from the bass pick-ups. Noise reduction/restoration plug-in hurt the quality too much, so I used Logic’s parametric EQ to notch out this frequency.

Final thoughts

Circling back to the original analog versus digital debate, it simply comes down to preference and the genre of the music. If you grew up on the classic rock, country, or RnB/soul music of the 70s, 80s, and 90s, then you’ll probably prefer the sound of analog. After all, those recordings were usually made in the best studios, by mixers at the top of their game, and using the finest analog gear of the day. Can you reproduce those exact sounds on your own computer with bog standard plug-ins? Maybe, but unlikely. On the other hand, if your musical tastes go off in a different direction – electronica, hip hop, etc – then maybe digital will sound better to you. There is no right or wrong answer, since taste is personal.

The trick is starting with a great recording that gets you nearly there and then enhance it. To do that, learn the tools you already have. Every DAW comes with a great set of built-in plug-ins. There are also many free and/or inexpensive third-party plug-ins on the market. The upside is that you can apply multiple instances of a fancy name-brand emulation on each and every track of your mix, which would never be possible with the real hardware due to cost. The downside is that you have so many options out there, that a lot of users simply amass a collection of plug-ins that they have no idea how to use. This induces option-paralysis.

If you own a ton of plug-ins, it’s a good idea to ween yourself off of them. Focus on a select group and learn them well. Understand how they work and when to use them. As I’ve mentioned, I like Omni Channel, as well as the Logic plug-ins. If you are looking for a family of products, it’s hard to go wrong with any of the tools from iZotope, Sonible, and/or FabFilter. Music mixing is about taste and emotion. Be sure to preview your mixes for some trusted friends to get their feedback. After working for hours on a mix, you might be too close to it. Then refine as needed. In the end, if you are doing this for fun, then you have only yourself to please. Enjoy!

Click this link to listen to the remix on Vimeo.

©2023 Oliver Peters

Chasing Analog Character

I started in radio and at one point considered a career as a recording engineer. But the path took me to TV and then video post-production instead. I do mix simple projects as part of being a video editor, but complex mixes tend to go out to an experienced audio professional running Pro Tools. Nevertheless, I do keep my hand in mixing music just for fun. Thanks to the internet, even if you don’t know a band to record, you can download high-quality multitracks to mix. It’s a good way to improve your chops for other types of mixes.

I’ve paid close attention to the trends in audio plug-ins and some of the better YouTube channels related to audio topics. Naturally, the internet algorithms push more of this content my way. One trend in analog-style plug-ins for the past few years has been to emulate the channel strips of some of the top audio mixing consoles from past decades.

A trip down analog lane

Originally audio consoles for mixing were variations of radio broadcast consoles. Mono at first, since AM radio was mono, and later stereo. The typical AM radio console in the 1950s and 1960s was a unit that sat on the desk and featured rotary volume knobs, aka pots (potentiometers), for each input. Above the pot sat a switch for on/off, output, and cueing (audition a record without going out over the air). Inputs were set for various mics, turntables, tape decks, and cartridge players. The console’s signal passed through an outboard brick wall limiter and then on to the transmitter.

As recording technology became more “sophisticated” (think The Beatles), console designs changed. Rotary pots were turned sideways and adjusted with a lever-style volume control (fader). At most, each input might have basic filtering/EQ controls. Coincidentally, multi-track recording also came into its own, with recorders shifting from mono and stereo to 4 and 8-track configurations. Therefore, these consoles were designed to have direct outputs. Mic input 1 (fader 1) was sent out through the fader control directly to channel 1 of the recorder, mic 2 (fader 2) to channel 2, and so on. In the early days, having even an 8-track recorder was uncommon, so consoles were still relatively small. The classic example is the REDD console used by Abbey Road Studios.

As recorders advanced and track counts increased (16, 24, 32, and eventually synced 24-track machines for 48 tracks), so did the console sizes. Fader design also evolved to a flat slider, allowing for tighter spacing and more inputs. The mixing console sitting on the desk gave way to large mixing consoles that were the “desk.” While there were and are many different manufacturers, Neve, API (Automated Processes, Inc), and SSL (Solid State Logic) are ones that stand out. These were the consoles often used in studios where some of the iconic rock LPs of the 70s, 80s, and 90s were recorded.

Solid State Logic

The SSL 4000 E was introduced in 1979 as the first mixing desk with integrated dynamics processing on every channel. It also featured a master bus compressor in the center section. I’m familiar with the classic SSL 4000 E series desk, which was the Audio 1 mix console we installed in 1989 at Century III at Universal Studios Florida. The biggest characteristic of this mixing desk is the channel strip, which is what plug-in developers try to emulate today. The SSL 4000 E is quite possibly the most emulated channel strip plug-in of them all.

Vintage SSL analog consoles are still prized gear in many modern studios. The general signal flow of the console goes something like this. During live recording each input channel takes a mic feed, processes it, and sends it back out to the tape deck via a direct output. Then, in playback that same tape recorder channel feeds the line input of the same SSL channel for further processing. The mixed output passes through the center section of the console for the final stereo mix.

The key selling point was and is the integrated processing on each channel, including mic preamps, filters, gate/expander, EQ (equalizer), and compressor. A studio wouldn’t need to buy tons of outboard gear for EQ and compression, because most of what you needed was already in the console. In addition, classic SSL desks, like most mixing consoles, included a patch panel so that outboard gear could be patched and inserted into any of the channel strips. This is the origin for effects inserts common on the software mixer panels of most DAWs and NLEs.

More importantly for mixing engineers, all of the common processing controls are at your fingertips. No need to reach around to a rack to adjust an EQ or compressor, since you can dial in a knob right there on the strip. Only specialized items like a Lexicon reverb or Eventide Harmonizer require moving away from the desk.

Studios standardized on certain console brands, because of the sonic qualities characterized by the design of each manufacturer. To a critical ear, a Neve analog console sounds different than an SSL. Each has a different mojo, thanks to the electronics under the hood, curves selected for EQ and compression, and more. Aside from the physical layout of the strip itself, it’s this sonic mojo that developers like Waves are trying to emulate when they license and release a channel strip plug-in that models a classic brand and design. Even the original companies like SSL have their own flavor of these plug-ins.

Moving into the digital realm

Like all hardware manufacturers, modern SSL mixing products include digital, as well as hybrid console designs. The hybrid desks feature a combination of hardware surfaces and software effects processing. However, most users are running DAWs (digital audio workstation). These applications feature a user interface with mixing panels that mimic the fader array of a classic mix desk. But few include the full array of tools that a classic analog console offered as part of the built-in software channel strips. Apps like Logic Pro or Fairlight (within Resolve) do include in-line EQ and/or compression. But, the general approach has been to rely on native or third-party plug-ins inserted into the strip.

If you want a certain compressor, insert it into one of the available slots above the fader. Click the plug-in to open it and adjust the settings. Unfortunately, if you have both a third-party EQ and compressor applied, then you have to open two different plug-ins, many of which feature skeuomorphic designs to emulate the look of the real hardware – some larger, some smaller. Your screen starts to get rather cluttered, especially if you are doing this on several channels at the same time.

A new trend has been emerging, probably due in part to Universal Audio’s Luna DAW. This application is focused on mixing and takes more of an analog approach than other DAWs. Not only is the approach different, but it also strives to infuse the sonic qualities of analog gear. So now we are seeing a wide range of new third-party channel strip plug-ins, which each attempt to emulate the look and sound of the channel strip portions of classic analog mixing desks.

The truth about analog emulation

The color, warmth, or character associated with the analog sound is due to imperfections. An old chief engineer of mine referred to EQs, compressors, and similar devices as “controlled distortion devices.” Analog hardware uses components, like resistors and capacitors, which were and are all subject to a variance in tolerances, aging, and worse.

When a plug-in developer makes a licensed digital plug-in designed to emulate some piece of classic gear, they are often working from schematics of the design or maybe a working version of the actual piece of gear. However, if they are trying to emulate a console channel strip, odds are that they are modeling a single version or only a single input. In reality, the signal flowing through each channel of an SSL or Neve console is going to be slightly different from one to the next. That’s a result of the variances in the electronics for things like harmonic distortion, even though they may still be well within the design specs. Although extremely minor, channel 1 might sound different from channel 2 and so on.

Listen to experienced mixers talking about their favorite studios and you’ll quickly learn they always sent drums through specific channels, guitars through others, etc – simply because of these variances. If you loaded up Pro Tools with a Waves SSL 4000 E plug-in on each channel of a mix, it would not sound identical to a mix done on an actual SSL console. Furthermore, a vintage analog console today that’s in good condition has often been recapped – meaning, capacitors and other aged electronic parts have been replaced. This affects the sound. A mix on a vintage SSL today might also sound different than on that same console when it was new 30 years ago. A lot of this chase for the ideal analog sound is rather Quixotic.

To emulate these minor variations, Brainworx integrates TMT (Tolerance Modeling Technology) into their channel strip plug-ins (AMEX 9099, SSL 9000 J, Focusrite SC, SSL 4000 E). They have modeled 72 slight differences, intended to reproduce the channel-to-channel sonic variations of a real console. If you apply one of these channel strip plug-ins to multiple inputs, you can opt to set the TMT setting to all be the same number (1-72), be sequential, or be random.

When you click the TMT button to random, then each plug-in uses a different model from the 72 choices. You can opt to re-randomize the order and in doing so, get a slightly different sound to the mix. Do this a number of times until you get the magical combination that you like. While the differences may be trivial, I can attest that the changes are real. Of course, you are applying this to different sounding instruments in a mix anyway, so does it really matter? You decide.

Turning your DAW into an analog desk

We are so enamored with the analog sound, that this has taken many different turns. For  example, Pro Tools now includes a plug-in/feature called HEAT (Harmonically Enhanced Algorithm Technology). Quoting from the Avid website: “HEAT does more than just warm-up your sound – it actually fuses the color characteristics of vintage analog consoles, vacuum tube circuits, and analog tape into your mix using high-quality, sophisticated algorithms. In the analog world, euphonic characteristics are introduced across individual audio tracks when mixing on an analog console or tracking to analog tape. HEAT works similarly, processing all audio tracks individually. But it also gives you the power to tweak its Drive and Tone controls globally to get the sound you want, whether that means something richer, brighter, smoother, or livelier. You can also A/B individual tracks or the entire mix to compare your handiwork, choose a pre or post insert state, or bypass HEAT altogether.”

A similar approach using the channel strip interface is featured in the Waves CLA Mixhub. Noted recording engineer Chris Lord-Alge has partnered with Waves to produced a number of CLA-branded plugins. CLA Mixhub is his variation on an SSL-style channel strip. You can apply the plug-in to up to 64 tracks. By assigning channels to “buckets” – 8 channels per bucket, 8 buckets total – it enables you to work more like you would on a traditional console. Click an instance of the plug-in the single view and you see the traditional SSL-style adjustments: EQ, filters, dynamics, etc. Click to bucket view and you’ll see a group of eight inputs at a time. Now you can select between each of the sections. This enables you to work with eight EQs or eight compressors all at once, much like you would on a real console. To my knowledge, no other plug-in works this way… yet.

Of course, Waves is a popular plug-in developer and they offer many other choices for channel strips. One of my favorites is the Andrew Scheps Omni Channel. Scheps is also a top mixer who has lent his name to several plug-ins. Rather than do his version of an SSL or Neve channel strip, Scheps had Waves combine the tools he likes best, taking a little bit from a variety of analog tools. Not only does it include many useful tools in a single plug-in, but you can re-arrange the signal flow order. Want to swap the compressor before the EQ, or Gate before the de-esser? No problem. There’s also an insert slot to add in other available Waves plug-ins on your system.

I’ve spent a lot of this post talking about plug-ins that look, feel, and sound like vintage, analog hardware. Yet, there are modern approaches to a channel strip as well. iZotope’s Nectar, Neutron, and Ozone are exactly that. In the end, the appeal to a channel strip is ergonomics. All of the important processing is there close at hand without the need to open multiple plug-ins each with different interfaces. Not only should they sound great, but they should be easy to use and help you get to a great mix quickly.

As I’ve stated before, these plug-ins are all designed first and foremost for audio applications. Most will work within editing applications, too, although with some exceptions. Test a trial version before you commit. But, if you’re chasing an analog sound and mixing experience, many of these tools are worth your experimentation.

Addendum: During the last week of March, Waves abruptly changed its business policy from perpetual licensing to a pure subscription model. Later in the week they backtracked and announced that both perpetual and subscription options would be available after all. More on that in the next blog post.

©2023 Oliver Peters

Loud, but not TOO Loud!

Clients always want a mix with impact. In their minds, impact equals loud, but that’s not really true. What you really want is a dynamic mix that fits within a comfortable range. The listener should hear some variance without the broadcasters and/or streaming platforms affecting the mix too badly through loudness normalization. Hence, the battle over loudness and in recent times, the various ways to measure it.

In the early 2000s the so-called loudness wars came to the attention of legislators who, through the FCC, eventually codified US standards with the CALM Act of 2010. This resulted in loudness targets measured in LUFS (loudness units full scale). It has become the standard for broadcast mixers ever since. However, years before the CALM Act, noted mastering engineer Bob Katz offered a solution in a white paper to the AES. His proposal was the K-scale, which is still a valid way to mix and is an available feature in a number of metering plug-ins. This method is known to recording engineers, but probably something that most video editors never knew existed.

The K-scale is based on the concept that mixing should always be done at a reference speaker volume. Based on research done by Tomlinson Holman (the TH in Lucasfilm THX) and Dolby for theater sound calibration, that level is 85 dB SPL (sound pressure level) in movie theaters. The K-scale weights a VU meter’s scale according to three levels of headroom. These are labelled as K-12, K-14, and K-20 for 12, 14, and 20 dB of headroom, respectively. K-20 is intended for mixes with a wide dynamic range like classical music, K-14 is good for pop music, and K-12 for broadcast. Of course, those genres are only suggestions.

The point is to preserve dynamic range in the mix based on the type of material. The key to the weighting of the scale is that 0 VU always equates to a speaker volume of 85 dB SPL. It’s intended for professional music mixing and mastering studios. However, don’t discount its value for modern video post. K-scale metering has two advantages. First, you are mixing with a VU style meter, which has an averaged response. That is more akin to how humans perceive loudness than fast-responding full scale meters common to most NLEs and DAWs. Second, it encourages consistent monitoring levels when you mix audio.

How do you implement this in a small edit suite?

First, set up the room properly if possible – depending on available space. Place your speakers at the front. Your seating location should be about 1/3 of the way from the front with 2/3 of the room space behind you. Sound absorption panels are highly recommended. The speakers should be about five feet from you. Calibrate your speaker volume using a sound pressure level meter. The simplest and cheapest option is a phone app. I use the free NIOSH SLM app on my iPhone set to C weighing. It’s fine for casual monitoring like this, but understand that a phone app is not accurate in critical environments due to how the built-in mic is calibrated.

There are several meter plug-ins that feature K-scale presets. I prefer the free mvMeter2 from TBProAudio, which emulates an analog VU meter. It includes numerous VU and PPM presets along with the three K-scale settings. Place this on your final master or mix bus without any other effects or level changes after the meter. Play a pink noise file in your timeline and set the output level so that it reads 0 VU on the meter scale that you’ve picked – for instance, 0 VU on the K-14 scale.

There are plenty of home theater guides about how far away you should be when measuring SPL. The official distance would be one meter from the speaker, but it’s also valid to measure from where you would normally sit. A comfortable listening volume is going to be somewhere between 70 and 85 dB SPL. Start at 85dB. Test your mix playback. If this sounds too loud, you can lower the level and retest with the pink noise file. You might not operate at that level the entire time that you are editing. However, keeping this consistent level each time you mix will give you more predictable and translatable results.

Using this system to mix

Now, let’s put this into practice. Most of the mixes I do myself are for broadcast TV or some social media outlet, like YouTube. Each uses different loudness targets. I find that the K-12 or K-14 setting is best for my internet work and K-20 is best for broadcast. Note that Katz recommended the K-12 scale for broadcast; however, that was before the industry standardized at -23 or -24 LUFS. I find that if you use the K-20 scale (with 20 dB of headroom) that the result is closer to this spec. That difference is probably because Katz was talking about music broadcast, as in radio airplay of songs, rather than TV programs.

The reading on the meter will react like a typical, analog-style VU meter. If you’ve mixed with VU meters in the past, then you’re probably used to the needle hovering around 0 VU and safely (and often) bouncing up into the red overload zone up to +3 dB or more. When using the K-scale method, you’ll find that a lower meter reading will actually match what sounds right to you, assuming you followed the speaker calibration described above. Most of the time my levels are somewhere in the middle of the VU meter, well under 0 VU. It only bounces above 0 on a few peaks. If you compare this to Premiere Pro’s loudness meter, you’ll find that this equates to around -20 to -14 LUFS, which is a good target for platforms like YouTube.

Remember that a VU-style meter is an average. Be sure to limit your signal for peaks. A limiter plug-in should be placed before the meter. In my work that’s usually set to a brick wall level of -3 or -6 dB true peak depending on the program. This is lower than the -1 dB limit often recommended for music mixes. This approach to mixing should give you a decent amount of dynamic range without the need for severe normalization by the platforms.

It’s important to understand that many of these recommendations for streaming service targets are based on music mixing and not TV shows and films, where dialogue is usually dominant. In 2021 the AES posted an updated technical document (linked here) that spells out recommendations for dialogue-based content. Their suggested target is -18 LUFS based on measuring the dialogue tracks rather than the full mix.

Alternatives

The K-scale system is old, though still valid. Of course, NLEs like Premiere Pro and Resolve include their own loudness meters and there are numerous third-party plug-ins using traditional and modernized ways to measure loudness. One modern metering technique is to measure the difference between peaks and short term loudness, aka PSR, and/or peaks versus long-term loudness, aka PLR. A plug-in like Meterplugs’ Dynameter provides you with this feedback. The reading will tell you how the mix level will be handled by the loudness management of the top streamers. Loudness Penalty is an online resource that will tell you this, as well.

Lastly, what about headphone mixing? There are pros and cons, but in order to trust your ears and know whether your mix will translate to various speaker systems, Waves has a solution. Their Nx technology emulates the three-dimensional sound and space of several famed studio control rooms, including those of Chris Lord-Alge, Ocean Way, and Abbey Road.

Place the plug-in last in your mix bus, select the correct headphone or ear bud model EQ curve (270 presets are available), and you’ll experience your mix based on an emulation of those speaker systems and rooms. Each plug-in includes head tracking and Waves sells a separate Nx tracking sensor. However, it can also use your front-facing computer camera.

With the plug-in enabled, you’ll hear a proper sound image as if listening to those speakers (Abbey Road, Ocean Way, etc.) instead of the usual left/right split of headphones. With head tracking, the spatial image shifts as you move your head. If you disable head tracking, you can still manually pan the image around the room and hear the result in your headphones. Since the plug-in is only designed for headphone monitoring, disable it when listening on external speakers and when you output the mix. Since the plug-in is manipulating phase to create these emulations, material that is truly just mono might sound more odd than full stereo mixes.

Granted, you’ll probably never own the kind of speaker systems these studios use, but the real point is that these plug-ins give you a transportable reference based on some of the best control room environments. As you move between different systems with different monitor set-ups, you can maintain a common monitoring reference for all of your mixes.

Happy mixing!

©2023 Oliver Peters

Could Fairlight be your next DAW?

When I review audio plug-ins and software, it’s from my perspective as a video editor. I’m not a recording engineer or mixer; however, I do dabble with music mixes as a hobbyist and to improve my audio chops. As such, I occasionally delve into digital audio workstation software, such as Sound Forge, Audio Design Desk, and others. My favorite is Apple Logic Pro, but as a DaVinci Resolve and Adobe user, I also have Fairlight (part of Resolve) and Adobe Audition. I touched on the Fairlight page in some detail as part of my Resolve Studio 18 review, but in this post I want to focus on it purely from the perspective of a DAW user on music projects.

Blackmagic’s reimagining

When Blackmagic Design acquired the assets of Fairlight, the software was refreshed and developed into the Fairlight page within DaVinci Resolve. Even though it’s nested inside of a video editing and grading tool, Fairlight is capable of being a standalone audio application. No need to ever have video enter into the equation.

Fairlight is integrated into both DaVinci Resolve (free) and Resolve Studio ($295). The Studio version can be activated on two computers at the same time. Nearly all Fairlight features and effects are the same in both versions, with the exception of ATMOS and spatial audio mixing/monitoring, which requires the Studio version. If your only interest is stereo recording and mixing, then Resolve is one of the only, truly free DAWs on the market. No significant feature restrictions and no Blackmagic hardware required. Plus, it works in Windows, Linux, and macOS.

Along with this software development, Blackmagic Design has expanded the ecosystem of companion Fairlight products. These include an accelerator card, a modular chassis, control surfaces, controllers, and an audio interface. The Fairlight page also supports Blackmagic’s two editor keyboards. You can run Fairlight without any external hardware, yet it’s scalable up to a complete recording studio rig. On a Mac, any Core Audio device will do, so recording into Fairlight and monitoring the output is compatible with simple USB audio interfaces, like Focusrite, PreSonus and others.

Understanding the interface

The Fairlight interface is compatible with single and dual-display set-ups and uses UI panels that can be turned on and off or slid onto the screen as needed. You can show or hide individual pieces of the mixer, as well. Unfortunately in a single display system, like an iMac, you cannot display the mixer panel full-screen. A project with 20 to 30 or more source tracks, requires left to right scrolling. However, since the 18.1 update, the meter bridge panel allows for two rows of meters.

The mixer uses a channel strip format for each track, which includes input/output/send routing, effects, and a built-in parametric equalizer and compressor. This is much like the channel strip of a traditional analog studio console, like an SSL or Neve. Unlike some other DAWs, you can also change the signal order of effects, EQ, and dynamics (compression) within each channel strip.

Modern plug-ins

Resolve includes Fairlight FX audio plug-ins that cover most common needs. But since this software is targeted towards the film and TV customer, it doesn’t include music-centric plug-ins, like the guitar amp and pedal emulations offered in Logic Pro. That focus is true of the plug-in presets, as well. For example, the factory preset choices in the compressor will be for dialogue and not musical instruments, like a drum kit or guitar. That doesn’t mean you can’t do music with these plug-ins. Presets are just suggestions anyway, so you should tweak based on what sounds right to you.

Fairlight does not color the sound. The sonic character, interface, and plug-in design take a clean, modern approach. There are no vintage options and none of the plug-ins are designed as skeuomorphic emulations of studio gear synonymous with classic recordings from the 70s. After all, film re-recording mixers have never been particularly precious about certain consoles or outboard gear from ages ago. Other than maybe a love for old Nagras, I doubt there’s much fondness for old audio gear like mag dubbers. At least not in the same way that music recording engineers still like to use analog recorders in the signal chain.

If you do want vintage tools, then Fairlight supports third-party AU and VST plug-ins. However, as with other video applications, I’ve found that some of the skeuomorphic effects don’t always work or look right. For example, I often use the free VU meter from TBProAudio. In Fairlight, only the AU version will appear as intended. And if you own an M1 or M2 Mac, then double-check that your favorite third-party plug-in is natively supported.

Fairlight isn’t just for audio post

Avid’s Pro Tools is the 800-pound gorilla. But, many Pro Tools users are often frustrated with the cost of staying current and dealing with Avid as a company. While such concerns may or may not be justified, Pro Tools isn’t the only game in town. Unless you need to interchange Pro Tools projects, there are plenty of alternatives. And that’s where Fairlight comes in. First of all, if audio post for film and TV is your primary focus, then Fairlight is up to the task. Resolve will import XML, FCPXML, and AAF files for both color and sound finishing. Fairlight includes an ADR recording routine, a free sound effects library, and a foley sampler plug-in. But let me focus on Fairlight as a music DAW.

I started with multitracks of song covers available from Warren Huart’s “Produce Like A Pro” YouTube channel. I didn’t record my own tracks, other than to test how recording might work. I’m a big believer that a great mix is achieved by doing 90% of the work at the time of the studio recording. It’s not about building the sound through plug-ins and tricks, but getting the right blend of gear, mics, and performance from the players. That was already there in the multitracks, so the mix was more about the right balance of these elements.

Achieving a successful mix

Fairlight works with as many tracks and busses as are created in your timeline. My standard layout for mixing is to use summing busses. You can create as many as you need. The 35 tracks for this song include drums, percussion, bass, piano, electric and acoustic guitars. I route each set of instrument tracks to a buss dedicated to that group, even if there’s only one instrument track in that group. These six busses are then routed to a submix buss, which in turn is routed to the master buss for output. This allows for gain staging and quickly balancing  levels. The default Fairlight layout automatically routes the first buss (drums in my case) as the output to the speakers and on the Deliver page. Be sure to change each of these to your master buss for the proper intended output.

My goal was to come out with a result that hit desired loudness targets and sounded good to me, mainly using the stock plug-ins. You’re going to adjust levels, but most of the effects center around EQ, compression, and reverb. Each of these is adequately covered by the complement of Fairlight FX. If you have singers, then there are also vocal processing effects, like de-essing. However, an investment in iZotope RX is certainly a useful add-on. For example, RX includes a specific tool to remove or reduce guitar squeaks and string noise. The Resolve 18.1 update added many audio-centric features, including a new voice isolation feature. It works well for any vocal situation and in my opinion has fewer negative artifacts than most of the competing options.

In my test mix, I adjusted level, panning, EQ, and compression on each channel strip. At the buss level, I added more EQ and compression, plus some reverb. The last stage was a multiband compressor and a brick wall limiter on the submix buss. Only meter plug-ins were added to the master buss. Of course, Fairlight includes its own useful set of meters for level and loudness.

Fairlight is actually quite good for music production, editing, and mixing. Since it’s built into an NLE, the project supports multiple mixes. You can have bins and timelines to organize the tracks and mixes for various different songs, as well as different versions of each mix. Resolve 18 added new cloud collaboration tools, however, you can easily collaborate on mixes by exporting a timeline file to send to a colleague. Assuming the other system has access to the same audio files and third-party plug-ins (if used), then it’s simply a matter of importing that timeline file.

Processing for this number of tracks and effects was easily handled by my iMac. It could have handled more, including more intense third-party plug-ins, like Gullfoss, Ozone, FabFilter, or Sonible. If you really need to go BIG, then Blackmagic Design promises up to 2,000 real-time tracks for the full Fairlight hardware installation! So if Pro Tools isn’t in the cards for you, then look over Fairlight and Resolve. It might just be right for your music mixing needs.

Additional thoughts

Some of the comments I received on the PVC version of this article (see link below) pointed out that Fairlight does not include such music-centric tools as MIDI and a piano roll, like some other DAWs do. While this is true, these are tools used by music creators working with synthetic instruments, like software samples for guitar, strings, drums, etc. That’s not a universal requirement, especially if you record and mix live performers using real instruments. Certainly if you need those specialized features, then other DAWs are a better fit for you.

It’s important to remember that digital audio workstation (DAW) software is used for a wide variety of audio production tasks. Such productions are often recorded and edited with tools that do not include some of these music features either. For example, Adobe Audition is widely used in the production of podcasts and radio commercials. So while Fairlight might not fit all needs, there’s little harm in trying a free application and then seeing where that leads.

Want to try mixing in Fairlight for yourself, but don’t have the tracks? Check out these 50 free, downloadable multitrack song sets from Warren Huart. I’ve only scratched the surface, so be sure to check out Blackmagic’s Fairlight training series.

This review also appears at Pro Video Coalition.

©2023 Oliver Peters

Sonible smart:comp 2

Audio software plug-ins (effects and filters) come in two forms. On one hand, you have a wide range of products that emulate vintage analog hardware, often showcasing a skeuomorphic interface design. If you know how the original hardware version worked and sounded, then that will inform your expectations for the software equivalent. The other approach is to eschew the sonic and visual approach of analog emulation and build a plug-in with a modern look and sound. Increasingly this second group of plug-ins employ intelligent profiles and “assistants” to analyze your track and provide you with automatic settings that form a good starting point.

Austria has a long and proud musical history and heritage of developing leading audio products. There are many high-end Austrian audio manufacturers. One of those companies is Sonible, which develops both hardware and software products. The Sonible software falls into that second camp of plug-ins, with clean sonic qualities and a modern interface design. Of key interest is the “smart:” category, including smart:comp 2, smart:limit, smart:EQ 3, smart:reverb, and smart:EQ live. The first four of these are also available as the smart:bundle.

Taking a spin with Sonible’s spectro-dynamic compressor

I tested out smart:comp 2, which is billed as a spectro-dynamic compressor. It’s compatible with Windows and macOS and installs AU, VST, VST3, and AAX (Avid) versions. Licensing uses an iLok or is registered to your computer (up to two computers at a time). Let’s start with why these are “smart.” In a similar fashion to iZotope’s Ozone and others, smart:comp 2 can automatically analyze your track and assign compressor settings based on different profiles. The settings may be perfect out of the gate or form a starting point for additional adjustments. Of course, you can also just start by making manual adjustments.

Spectro-dynamic is a bit of a marketing term, but in essence, smart:comp 2 works like a highly sophisticated multiband compressor. The compression ranges are based on the sonic spectrum of the track. Instead of the four basic bands of most multiband compressors, smart:comp 2 carves up the signal into 2,000 slices to which compression is dynamically applied. As a compressor, this plug-in is equally useful on individual tracks or on the full mix as a mastering plug-in.

In addition, I would characterize the interface design as “discoverable.” When you first open the plug-in, you see a clean user interface with simple adjustments for level and ratio. However, you can click certain disclosure triangles to open other parts of the interface, such as control of attack and release timing, as well as side-chain filtering. There are three unique sound shaping controls at the bottom. Style controls the character of the compressor between “clean” (transparent) and “dirty” (warm and punchy). The Spectral Compression control dials in the amount of spectral (multiband) compression being applied. At zero, smart:comp 2 will act as an ordinary broadband compressor. The Color control lets you emphasis “darker” or “brighter” ranges within the spectral compression.

Simple, yet powerful functions

Start by selecting a profile (or leave on “Universal”). Play a louder section of your mix and let smart:comp 2 “learn” the track. Once learning is done and a profile established, you may be done. Or you may want to make further adjustments to taste. For example, the plug-in features automatic input riding along with automatic output (make-up gain). I found that for my mixes, input riding worked well, but I preferred a fixed output gain, which can be set manually.

There’s a “limit” function, which is always set to 0dBFS. When enabled, the limit option becomes a soft clipper. All peaks exceeding 0dBFS will be tamed to avoid hard clipping. It’s like a smooth limiter set to 0dBFS after the compression stage. However, if your intended use is broadcast production, rather than music mixes, you may still need to add a separate limiter plug-in (such as Sonible’s smart:limit) in the mastering chain after smart:comp 2. Especially if your target is lower, such as true peaks at -3dB or -6dB.

smart:comp2 did a wonderful job as a master bus compressor on my music mixes. I tested it against other built-in and third-party compressors within Logic Pro and DaVinci Resolve Fairlight. First, smart:comp 2 is very clean when you press it hard. There’s always a pleasing sound. However, the biggest characteristic is that the mixes sound more open with better clarity.

smart:comp 2 for mixing video projects

I’m a video editor and most of my mixes are more basic than multitrack music mixes with large track counts. Just a few dialogue, music, and sound effects tracks and that’s it. So the next test was applying smart:comp 2 on Premiere Pro’s mix bus. When I originally mixed this particular project, I used Adobe’s built-in tube-modeled compression on the dialogue tracks and then Adobe’s  multiband compressor and limiter of the mix buss. For this test, I stripped all of those out and only added smart:comp 2 to the mix output buss.

I noticed the same openness as in the music mixes, but input riding was even more evident. My sequence started with a 15 second musical lead-in. Then the music ducks under the dialogue as the presenter appears. I had mixed this level change manually for a good-sounding balance. When I applied smart:comp 2, I noticed that the opening music was louder than with the plug-in bypassed. Yet, this automatic loudness level change felt right and the transition to the ducked music was properly handled by smart:comp 2. Although the unprocessed mix initially sounded fine to me, I would have to say that using smart:comp 2 made it a better-sounding mix overall. It was also better than when I used the built-in options.

How you use plug-ins is a matter of taste and talent. Some pros may look at automatic functions as some sort of cheat. I think that’s wrong. Software analysis can give you a good starting point in less time, allowing more time for creativity. You aren’t getting bogged down twirling knobs. That’s a good thing. I realize vintage plug-ins often look cool, but if you don’t know the result you’ll get, they can be a waste of time and money. This is where plug-ins like the smart: series from Sonible will enhance to your daily mixing workflow, regardless of whether you are a seasoned recording engineer or a video editor.

©2022 Oliver Peters