Acon Digital Restoration Suite 2

When it comes to audio plug-ins, video editors have different needs than audio mixers. Sure, you need EQ, compression, and limiting, but the plug-ins you reach for most often revolve around the clean-up and enhancement of dialogue.

There are a number of third-party plug-in solutions, which augment the built-in enhancement features of most editing applications. However, solutions like the full version of iZotope RX can be pricey, especially for upgrades. Accusonus, another alternative, has left the plug-in business. The tools in Premiere, Final Cut Pro, and DaVinci Resolve vary in effectiveness; however, they lack comprehensive user control, often presenting only an amount slider. So there’s room for innovation.

A friend recently pointed me to Acon Digital’s Restoration Suite 2. Acon Digital is a Norwegian software developer that offers a portfolio of audio plug-ins. The Restoration Suite includes four professional-grade plug-ins: DeNoise 2, DeHum 2, DeClick 2, and DeClip 2. Their larger Acoustica application (available in Standard and Premium versions) adds a few more tools. Yet, for most video editors Restoration Suite 2 fits the bill.

Version 2 was introduced in 2019 and added 64-bit OS support, improved algorithms, and Mid/Side processing functions. Acon Digital’s software supports Windows and macOS and runs natively on Apple Silicon, as well as Intel processors. The plug-ins install AU, AAX, VST, and VST3 versions.

DeNoise 2

Of the four plug-ins that constitute Restoration Suite 2, DeNoise 2 provides the most versatility. It’s designed to remove background noise, like wind or waves hitting the beach, but there are additional factory presets for voice and music. You can run it adaptively or with a noise profile (noise print). Adaptive processing can tackle broadband or combined noise. The difference is that combined processing takes into account noise with a tonal quality, like hum. Using the combined mode will affect the voice to a greater degree.

The noise profile works in a similar fashion to other tools. Run a piece of the audio that only has background noise for a few seconds and click Learn. Then click the “power” icon to apply that noise profile. According to Acon Digital, “Version 2 introduces the novel dynamic noise profiles that help reducing noise that varies randomly over time, such as wind noise or rustle from lavaliere microphones. Where the earlier versions merely captured a static noise print with time-constant noise levels, the dynamic noise profiles capture statistics from the noise to be reduced. The noise suppression algorithm then estimates the most suitable noise threshold curve for the noisy input signal using the measured statistics.”

Unlike most other noise reduction plug-ins, the DeNoise 2 interface includes controls for reduction, knee, attenuation, and reaction time. The Adaptation Time slider sets how long before the the plug-in responds to changes in the noise floor. A shorter time means that processing kicks in more quickly, but it can affect the desired signal.

A histogram dynamically displays the audio signal versus the processing curve. Click Emphasis and you now have control at multiple frequency ranges. Let’s say you want to remove background wind noise. That’s usually a higher frequency noise. Simply drag down that control point and adjust the curve. You can then raise the other control points if you like. While sounds like wind noise work well in the adaptive mode, I got the best result setting a noise profile and using Emphasis to tweak the control points. Finally, there’s a Mid/Side mode if you are working with stereo source material.

DeClick 2

DeClick filters are commonly used to remove clicks, scratches, and pops in recordings. This is typical with any music tracks that come from an older vinyl LP. Another source of clicks can be from digital recordings. This filter can be used to minimize them if minor, but if the audio is completely trashed, you are out of luck. However, DeClick 2 has other uses, such as the reduction of plosives – a voice-over announcer popping “p” consonants. There are separate factory preset groups of 78 RPM, Vinyl, and Voice. The later group includes presets for reducing mouth clicks and for plosives. So DeClick 2 covers more audio artifacts than the name might imply.

DeClip 2

This plug-in is designed to restore distorted, clipped recordings, such as an over-driven voice recording. DeClip 2 replaces distorted peaks with an estimate of the proper signal level. The histogram displays the signal with a positive and negative threshold slider.

The first step is to adjust the input gain so the signal is loud enough for the filter to make a proper correction. But, make sure some headroom is left. Pick the worst-sounding section and click Detect for an automatic selection. Then manually tweak the two threshold sliders to fine-tune the sound. Adjust the output gain if needed. This filter did a great job for me in recovering the transients in my test clips, thus repairing otherwise distorted voice recordings.

DeHum 2

I didn’t have a real-world dialogue source with hum to test this last plug-in. Instead, I created my own, taking a VO track, mixing in 60Hz hum, and bouncing that out as my test clip. DeHum 2 completely removed the embedded hum without distorting the voice. There’s a preset for 50Hz and 60Hz sources, along with a variable frequency control. You can manually dial in the frequency and sensitivity or click the “target” icon to set the profile automatically. The hum in my test clip turned out to be 59.97 instead of a true 60Hz.

The number of harmonics can be selected, should the offending hum have those. This is displayed on the histogram. There are two modes selected by toggling the Notch Filter button. The Notch Filter mode reduces CPU load, but can impact the voice more. When it’s disabled, DeHum 2 subtracts a hum signal created through a re-synthesis technique in order to minimize signal distortions.

Conclusion

Acon Digital has developed a very useful audio enhancement/repair toolkit for video editors. It’s also handy for anyone producing podcasts – especially those recording interviews via Zoom. These four plug-ins are easy to set and adjust and give you plenty of control. Three include a solo function to monitor the noise being removed. In addition to the factory presets, you can save your own – tailored to your particular audio needs.

For complex challenges, stack more than a single instance of these filters onto a clip. For instance, you might apply DeClick 2 (remove plosives) plus DeNoise 2 (remove background noise) to the same on-camera presenter audio for the cleanest results. Each filter can tackle a range of similar audio artifacts.

When you compare that to competing products, others might require you to buy several plug-ins to tackle the same set of conditions. For instance, you might have to purchase separate filters for plosive removal and mouth clicks, rather than one filter that’s able to perform either task. With only four individual plug-ins contained in the Restoration Suite 2 bundle, you might mistakenly think another bundle with more plug-ins is also more comprehensive. That’s definitely not the case here.

Along with Acon Digital’s Restoration Suite 2, Acoustica, and a separate Mastering Suite, the individual plug-in products also include a handy, free reverb filter (Verberate Basic). You may run the software on as many computers as you personally own and have control of. All of the plug-ins that I tested worked well in the Apple and Adobe DAWs and NLEs that I use. Be sure to check out a trial version first if you have any questions about your particular kit.

This article was originally written for Pro Video Coalition.

©2023 Oliver Peters

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

The Color of Analog – FabFilter Saturn 2

In the beginning there was analog and it was good. Then audio engineers developed digital and it was better. Wait, not so fast! The more we heard pure digital production, the more we started to miss the character – and yes, flaws – of analog. After all, grit, saturation, and distortion are synonymous with rock ‘n’ roll. And so, since the dawn of the digital era, many audio software developers have designed products to bring back some of that missing analog sound.

The character or color of analog is generally a function of saturation, which originally came about because of the physical medium of audio tape and electronics used to record and mix songs. Tubes, transformers, and tape introduced internal compression and frequency roll-offs, but most importantly, harmonic overtones. You could push the signal hard and get a certain amount of “warmth.” Push it past the limit and you’d get distortion, but not harsh clipping in the same way as digital signals.

Plug-in developers have sought to emulate the characteristics of analog through various types of emulation. Generally these take the form of a simple saturation or overdrive plug-in. The bulk of these feature very simple controls with a range from subtle warmth to complete distortion. Unfortunately, you don’t have many options to tailor the effect with most of them.

Enter FabFilter Saturn 2

If you really want the ultimate in saturation control, then one of the best tools on the market is FabFilter Saturn 2 from FabFilter Software Instruments. I’ve reviewed their Pro-L2 limiter and it’s since become my go-to limiter for any music mix that I do. It’s often a good idea to stick within a family of products when you find a top-notch developer. There’s a reason that FabFilter plug-ins are the preferred choice by many mixers.

First of all, if you want a plug-in that mimics the appearance and operational controls of some vintage piece of gear, then this isn’t it. All of the FabFilter products feature a similar, modern design aesthetic with easy-to-use controls. At first glance, FabFilter Saturn 2 appears deceptively simple. But, once you dive into the presets, options, and various controls, you’ll find that it can work as a simple saturation plug-in all the way up to a multi-band audio processing device.

The main set of controls are accessed from the bottom control bar. The style selector button covers a range of emulations, including vacuum tube, audio tape, guitar amps, saturation, transformers, and special effects (smudge, breakdown, foldback, rectify, and destroy). Each of these categories includes numerous options within. As you can see from these groups, the options cover not only a wide range of musical production possibilities, but also broadens into sound design. The central drive dial controls the strength of the saturation. Controls to the left and right let you further adjust the sound. At the top right, you can also select from a wide range of presets.

Click into the top area of the interactive display to change the plug-in into a multi-band processor. Each band has its own set of controls. You can also adjust the crossover position and slope between the bands. Each band can make use of different styles of emulation, along with different control adjustments.

Modulation

In the default mode, Saturn 2 looks pretty simple. But you can open the area below the control bar to further modulate any of the effects. There are numerous ways to modulate these and the interface uses a handy color scheme and visualization to give you a better idea of what’s being done.

There are also drag points. For example, click on the circle above an envelope setting and drag it to one of the controls on the control bar. In FabFilter’s description this now connects a modulation source to a modulation target. These are now linked and the action of one impacts the other.

Trying to explain how this works for a blog post would really get us in the weeds. It’s better handled by FabFilter’s user guide or their comprehensive YouTube tutorial.

In use

Saturn 2 installs on both Windows and macOS (Intel and Apple Silicon) systems in AU, VST/VST3, and AAX formats. Therefore, these plug-ins will work with most video editing and audio mixing applications. There is no right or wrong way to use Saturn 2. For example, you could apply it with a default setting to a vocal track, pick a warm tube selection, and dial in a bit of drive for character. Since it has four tone sliders, you can also use it for a bit of equalization.

Or add Saturn 2 to a guitar track and use one of the guitar amp emulations. If you’ve ever used Apple Logic Pro, then the options will be similar. The choices are listed by generic names, since actual brands haven’t been licensed. However, the selections are designed to sound similar to recognizable amp manufacturers, such as Fender, Vox, Marshall, and others.

You can certainly use Saturn 2 on an instrument stem or the final mix bus and dial in a more aggressive setting. This is where the presets help you to learn how the controls function. Apply the plug-in to the mix bus and you can radically add character to the complete mix. Of course, there’s no reason you couldn’t apply multiple instances of Saturn 2 at various stages to get the analog color you are looking for.

Aside from analog character for music, Saturn 2 can also be used for some pretty extreme effects. Need to pitch down a voice-over to sound like a monster? No sweat – Saturn 2 can do this. Special effects settings work for vocal processing, sci-fi sounds, and musical processing, like ping-pong and tremolo filtering.

I’m personally running this on mixes on a 2020 iMac in Logic Pro. Like FabFilter’s other filters, it performs well. Even when I have Saturn 2 applied to numerous tracks, there’s really no drag on the application. The filter has two high quality settings – good and superb. Even with all instances set to superb, my iMac had no issues. While you can definitely go crazy with the possibilities, a little goes a long way. When using Saturn 2, it’s best to start out with a subtle setting and dial in the various adjustments by ear.

There are plenty of other saturation filters on the market, but I have yet to find one with such a wide range of filtering options. This is especially helpful when your audio production needs cover more than just music mixes. It’s hard to tell from demos and blog posts whether a plug-in fits your needs, so be sure to check out FabFilter’s 30-day trial for this or any of their other outstanding plug-ins.

©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

NLE Tips – Audio Track Mixing in Final Cut Pro

In the past I’ve explained how audio in routed through the Final Cut Pro architecture. I’ve also discussed track-based audio mixing, predominantly based on the workflow in Premiere Pro. Today I’d like to extend that workflow into the realm of Final Cut Pro.

Everyone knows that FCP is not track-based. The timeline consists of a string of audio/video clips called the primary storyline, which empowers its magnetic feature. Additional audio and video clips can be attached to the clips on the primary storyline as connected clips – video above, audio below. At this level the software is indeed trackless. (Click on any image to see an enlarged view.)

Understanding audio roles and lanes

Several years ago, Apple added the “roles” feature. Audio and video clips can be assigned default and/or custom role designations, which can be used for visual organization and other functions. For example, do you want to export a “textless” ProRes file from your timeline? Then simply disable the Titles video role in the export dialogue.

Apple engineers have done more with audio roles, which can be further grouped into audio “lanes” through the timeline index window. If you’ve assigned the correct audio roles to each clip, then all dialogue clips are grouped into the dialogue lane, all music clips in the music lane, and so on. If you exported an FCPXML file for an outside mixer, then audio roles help to organize the track layout in other audio software.

At this point the clips are still individual. However, once you combine all clips in the sequence into a single compound clip, then the audio for all clips within an audio lane are summed together. This is similar to a group or submix bus in a DAW. The combination of lanes are in turn summed together and sent to the mix output. In essence, each audio lane within the compound clip is similar to a summing track stack in Logic Pro. You can adjust volume and apply effects to the entire lane, on top of anything done to individual clips contained inside of that lane.

Mixing in FCP on real-world projects

I’m working on an Alaska travelogue series – on-camera host on location, voice-overs, voice-over pick-ups, and music. The host stand-ups were recorded in two environments – close to the shoreline and in a quiet wooded area.

The location sound mixer recorded both a Lavaliere mic and a boom mic on separate channels. My personal preference is the boom, but sometimes the waves on the beach created too much background noise. In those cases, it’s the lav mic, but then I have to contend with the duller sound of the mic under the clothing, along with some rustle.

The next challenge is getting the voice-overs to sound close to the on-camera audio. These were recorded on location, but in a quiet room. The final challenge is to match the sonic quality of the voice-over pick-ups (done by the host at his home) to the original voice-overs.

Step One

The first step in this process is to assign the proper audio roles before clips are edited into the FCP sequence. Roles are quite versatile. If you had multiple speakers, each one could be assigned a separate role. In this project, my audio roles are Dialogue, VO, VO2, and Music. Once clips are imported and roles assigned, I can edit as I normally would in Final Cut. I personally add very few audio effects at this point to the individual clips, because I will do that later. In addition, certain effects, like noise reduction simply don’t work very well with short clips (more on that in a minute). So I only add what I need to “sell” the cut.

Step Two

Once the cut is approved and locked, I can move on to a final mix. To start, I’ll remove any audio effects that I’ve added to individual clips. Then, I meticulously go through and even out any level imbalances. Final Cut Pro features multiple gain stages. You have the clip volume control, but if you expand the audio, you see the individual channels, which each have volume controls, as well. Each of these can be raised by up to 12dB. So if you’ve applied 12dB to the clip and it’s still too quiet, expand the audio and bump up the channel volume. Or work this process in reverse. My objective is to end up with a clip volume that’s a bit hot in the peaks and then use the range tool to highlight the larger peaks and duck them down a bit.

Expand audio and make sure you have overlaps with fade handles between all clips. This is somewhat time-consuming. It’s far simpler in Premiere Pro to add audio dissolves (crossfades) across all audio edits in the timeline in a single step. But it’s a necessary step, including the addition or room tone/ambience to fill any gaps in the speech.

Finally, check the music. Make sure the edits work musically. Overall, the music volume can be a bit loud at this stage, but you want to make sure the balance is right for the entire sequence. So pay attention to the proper and graceful ducking of music around spoken audio.

Step Three

After you’ve made everything as uniform as possible, compound the sequence. Open the timeline index, enable “show audio lanes” which expands the audio of the compound. You’ll now see a “track” or summing bus for each audio role – Dialogue, VO, VO2, and Music. When you select an audio lane, you can adjust its volume and apply audio effects to only that lane. That lane’s audio parameters are shown in the inspector pane.

Selecting the topmost level of the clip, displays the output (i..e mix) bus parameters. Additional effects can be added here. It’s fine to apply and adjust such “master” effects, but I recommend that you do not make any changes to the volume. That’s because the volume control comes after any effects, which would include a meter plug-in, such as the built-in multimeter plug-in. Leave the volume slider alone if you want to see accurate volume levels.

Aside from mixing in tracks/busses, audio roles add another value at the time of export. My deliverables include a ProRes file without titles, as well as audio that’s split into separate tracks. In Final Cut Pro’s export setting, I can select the Multitrack Quicktime and then arrange the combination and order of roles. For this project, it’s a ProRes file with four stereo tracks corresponding to the four roles that I’m working with.

Note that when you export a multitrack file, each lane output also has any master output effects added to them. For example, if your mix uses a compressor and a limiter on the main output of the compound clip, then each lane/bus/track of the multitrack will now also have the added effect of that compression and limiting. If you don’t want this, then make sure to disable these effects prior to exporting a Multitrack Quicktime file.

Which effects should you use?

I’ve now discussed how the process works, but what combination of effects should you be using? Obviously that’s a question of style and personal taste. The type of effects for me will be similar to my description in the Premiere Pro article. I tend to stick with native Final Cut Pro effects, so that I don’t have to worry about what’s installed if I move to another Mac or a different editor has to step in. Also, Final Cut Pro is often a poor host for some third party audio plug-ins. I don’t know the reason, but have been told it’s up to those developers to optimize their tools for FCP. In most cases these same plug-ins work well in Logic Pro, not to mention other non-Apple applications. Go figure!

I’m happy with most of the built-in Apple audio plug-ins, with the exception of noise reduction and other audio repair tasks. The Accusonus tools are my go-to, but they are sadly no longer available. After that it’s the RX package from iZotope. If you have a really challenging piece of audio, then use the standalone RX package on that clip and re-import. If you don’t own either of these, then the newly added voice isolation feature in Resolve is pretty sweet (and better than what’s in FCP). Another impressive contender is Adobe’s Podcast beta. The AI-powered voice enhancement feature is available for free use through their web portal. I’ve used it for some really poor Zoom interview audio and it did an outstanding job of cleaning up all manner of audio defects.

Where this explanation is most pertinent is on location-based dialogue recordings. These are the ones that often benefit from noise removal/repair. These tools require consistency and some lead-in to the first audio, so they are best applied to full tracks and not individual clips. That’s why I make sure I have overlaps and fill in gaps and do all of this processing on the lanes of the compound and not on individual clips. If you have different dialogue sections – some noisy and some clean – then it’s best to organize these into separate audio roles, so that they are sorted out correctly once you compound the clip.

My typical processing chain

My FCP effects layout is similar to the description in the Premiere Pro post. Dialogue and VO tracks get some noise reduction, EQ, and compression. Voice-overs are particularly susceptible to plosives (popping “p” consonants) and sibilance, so plosive and de-essing filters are useful. For music, I usually spread the stereo image more and dip the EQ in the midrange. Plus some compression. All of this is designed to allow the dialogue to sit better in the mix. 

The last level of processing is what you do to the top level of the compound clip itself. That’s a bit like mastering in audio production. Applying effects to the compound clips is analogous to applying effects to a mix or output bus in the DAW world. On this particular chain, it’s EQ, exciter, compressor, adaptive limiter, and the multimeter. The effects stack is processed before the volume slider. Since I’m judging peak and loudness levels with the multimeter plug-in, I don’t want to make any volume slider changes on the compound clip, because those would be applied after the reading on the multimeter.

You’ll notice from my screen grabs that different compressor models have been used. These are all from the same Logic Pro compressor in FCP. This single plug-in features various presets designed to emulate tried-and-true analog compressors favored by top recording engineers/mixers.

Final thoughts 

As with my other Final Cut Pro audio articles and posts, I can already hear some screaming that this is just a workaround for the fact that Final Cut Pro has no “true” audio mixing panel. While that may be true, it’s also irrelevant. Until such time as Apple’s ProApps engineers redesign the audio section or add a “roles-based mixer” to the tool set, this is the software you have. If you want to mix in Final Cut Pro and deliver a properly mixed master file without using specialized audio software, then it’s best to understand how to achieve the required results.

If you step into the compound clip to make any editorial changes to the sequence or to individual clips, then you will not hear the results of the top-level mixing and effects. The proper mix is only heard when you step back out. This is a short-coming compared with this same process in Premiere Pro. Therefore, when you are editing in Final Cut Pro, it’s best to leave all of the final mixing until the end. In Premiere Pro, I tend to mix as I go.

Hopefully this post gives you some insight into the “guts” of the software. If you can’t send the audio to a mix engineer and don’t want to bounce over to Logic Pro, Pro Tools, or Resolve (Fairlight) yourself, then there’s no reason Final Cut Pro can’t be made to work for you.

©2023 Oliver Peters