Working with plug-ins is fun, but it gets complex when you want to be consistent across multiple hosts. The built-in effects can be quite good and if you only ever work in Media Composer, Resolve, Premiere Pro, or Final Cut Pro and are happy with what’s included, then nothing more is needed. But if you work in multiple applications, then what you like in one will be missing in the other. For example, the Logic compressor is available in FCP, but not Logic’s vintage EQ. If I use native effects in FCP, I have to use different effects to achieve the same results in Premiere Pro.
That problem can be solved by purchasing a plug-in bundle that is consistent across multiple hosts. If you install audio effects that support AU, VST, VST3, and AAX, then you are covered for Macs and PCs, and nearly all DAW and NLE brands. However, such bundles and/or individual plug-ins are typically authorized for a single machine at a time, via an activation code, a licensing portal, or a USB license key, like an iLok. If you operate a multi-seat shop, then it’s complicated juggling plug-in licensing across several machines. Hence, you have to purchase a plug-in set for each workstation, which can be costly. So free options become quite attractive. Install them on all the machines and never deal with the “missing plug-in” error message again.
I’ve run across two companies with free products that I find to be quite useful. The first is TBProAudio. They offer a range of audio plug-ins, including a couple of free products. The first is the sTilt v2, which is a linear phase equalizer, also known as a spectral-tilt or tilt-shift equalizer. Think of the frequency spread as a playground teeter-totter. The audio spectrum is on a “slope” that pivots on a center frequency. As you move the dial to the right, audio frequencies above the center frequency are boosted and audio below is cut or reduced. The result is a brighter sound. Move the dial to the left and upper frequencies are cut, while lower frequencies are boosted for a warmer sound. Adjust the center frequency value to move the “fulcrum” of the tilt-shift processing.
Another one of their free plug-ins is the mvMeter 2. This classic, analog-style meter array features several metering models, including, VU, RMS, EBUR128, and PPM. I started in radio, so working with VU meters is second nature to me. Since finding this plug-in I’ve used it on nearly every mix. I find that my mixes are now more standard with more consistent levels than simply judging by the built-in full scale dB meters.
Tokyo Dawn Records/Labs
As I searched for more useful plug-ins, I also ran across Tokyo Dawn Labs, a software offshoot of Tokyo Dawn Records in Germany. They offer a number of plug-ins, including four free products. Each of the free products includes a paid GE (“gentlemen’s edition”) version with additional features. The free products are not severely limited “lite” versions, but in fact, include 80-90% of the functionality of the GE products. These include two equalizers and two compressors, which are amazingly good – free or not.
TDR VOS Slick EQ is a mixing/mastering equalizer with several emulation models – American, British, German, and Soviet. Each model mimics certain gear or console characteristics. The American model is the most transparent. Slick EQ’s general operation is like most classic, three-band EQs with hi/low pass filtering and shelving controls.
TDR Kotelnikov is a dynamics processor, i.e a compressor/limiter. It has a very smooth and transparent sound with processing that’s affected by a stereo density control. Its transparency makes this tool ideal to apply to the final stereo output or master mix bus of any mix.
TDR Nova is a bit harder to describe. TDR calls it a parallel dynamic equalizer. It looks and acts a bit like a four-band parametric equalizer, however it also includes compression. So you can use it simply as an EQ, or you can combine that with compression to create a multi-band compressor.
TDR Molotok is another dynamics processor. I haven’t tested this one, but it definitely has the most old-school UI of the bunch. TDR states it doesn’t emulate any specific vintage device, but has what they describe as eleven flavor nuances. For me personally, Kotelnikov fits the bill for video project mastering, But If I were a music producer, then Molotok would hold some appeal.
An interesting aspect to these plug-ins is that default processing is stereo, but it can also be put into a sum or difference mode. Effectively this enables mid or side signal processing. For example, if you want to only process the middle portion of the stereo signal, set the filter to the sum mode. In addition, the filter can be switched from Precise to ECO (economy) in case you are working with an underpowered computer.
In wrapping up this series of posts, I want to point out that not all application hosts treat audio plug-ins equally. Typically DAWs generally do the best job of working seamlessly with third-party audio products. That’s less the case with NLEs.
If you use a Mac, you can install both AU and one of several VST versions of a plug-in. PCs only use VST varieties. However, in some cases, the AU version may have slightly different UI properties that the VST flavor. If you use Avid products, make sure to verify that a plug-in offers AAX and/or AudioSuite versions.
Finally, if you are a Final Cut Pro editor, tread lightly with plug-ins. FCP has increasingly become touchy with third-party audio plug-ins (under Big Sur), including many that play well with Logic Pro. And, of course, not all third-party plug-ins are yet fully compatible with the new Apple Silicon-based Macs. So make sure you test a trial version before you commit to a purchase.
There are plenty of paid and free audio plug-ins on the market. They all fit into the good, the bad, or the ugly categories – some great, some not so much. One of the better developers of modern audio plug-in effects is FabFilter Software Instruments in the Netherlands. While FabFilter products are known and respected in the music recording industry, they are not as well known among video editors. Any of their plug-ins would provide you with a great software tool, but the plug-in that I felt was the best fit for a video editor was their Pro-L 2 limiter plug-in.
An audio limiter, just like a broadcast safe video limiter, is typically used as a mastering tool applied to the last stage of the audio chain. You can certainly use a limiter on an individual track, like drums in a recording session or a voice-over in a video mix. However, limiting is most often applied to the final output – the master or mix output bus. While a limiter is really just a variant of a regular compressor, it is optimized to catch and restrict all peak levels and transients in order to make sure that your mix is compliant with a given loudness target.
FabFilter Pro-L 2 Limiter
Like most third-party plug-ins, the Pro-L 2 limiter installs as an AAX, AU, and/or VST/VST3 plug-in and so is compatible with most DAWs and NLEs. FabFilter plug-ins use a license key activation code after installation, so no need to mess with separate license management applications or a physical iLok hardware key. I tested the Pro-L 2 limiter in various applications and performance and behavior was great, even in Final Cut Pro, which has lately been touchy for me when using some third-party audio effects.
At first glance, the Pro-L 2 limiter might seem like most other limiter filters, but looks can be deceiving. This plug-in is rather deep with many nuanced adjustments that are easy to overlook. The good news is that FabFilter has done a good job with video tutorials and both an online and PDF user guide.
There are three big selling points for me. First, Pro-L 2 supports various mix configurations – not just mono and stereo, but also surround, including Dolby Atmos. Second, there’s built-in loudness metering. This includes an earlier K-system metering method (developed in the late 90s by noted mastering engineer Bob Katz), as well as current ATSC and EBU loudness scales. Finally, it’s the sound. You can drive the input truly hard into gain reduction and the audio stays extremely smooth-sounding without coming across as heavily compressed or distorted.
The Pro-L2 user interface is well-designed with several size options, including full screen, as well as a compact mode that hides the audio waveform graph. Metering can be changed from standard (input, output, gain reduction meters) to full loudness. Several of the components, like the advanced control panel and output gain knob are fly-out panels that might not be readily obvious until you get used to the plug-in. As this is a minimalist UI design, there are other controls, like oversampling and true peak limiting, which are enabled by small control buttons along the bottom.
One UI tool that I really liked was the lock icon. When this is unlocked (disabled) then every time you switch between limiter algorithms or presets the input and output gain levels reset, which makes it harder to compare settings. However, when it’s enabled, the gain levels are “locked” as you toggle through the options.
One final UI feature to note is that you have control over the waveform scrolling method. The display represents audio levels, gain reduction, and peaks. There are four scrolling modes depending on how you prefer to see the waveform being drawn onto the screen.
The key to the FabFilter Pro-L limiter is how it handles sound. There are numerous presets and eight limiter algorithms designed with distinct character depending on the type of audio you are processing. The last four (Aggressive, Modern, Bus, and Safe) were newly added in version two of the limiter. So whether you want something with a little crunch or totally transparent, this limiter offers you choices.
The general operating controls are similar to other compressors and limiters. There are input and output gain controls – the combination of which determines the amount of gain reduction (limiting). Attack and release controls affect how quickly and how long afterwards limiting is taking place. In addition to lookahead (how far ahead the software is looking for predicted peaks), there is also an oversampling control, which may be CPU intensive. Sound is analog, so fast peaks can occur between the regular digital sampling intervals. These peaks can, therefore, be missed by a limiter. Oversampling is a technique to catch and process any inter-sample peaks.
Channel linking is another powerful tool. Generally, a plug-in is going to process the left and right sides of a stereo signal equally. But what if your track has harder peaks on one side or the other? That’s where the channel linking controls come into play. The Transient control knob alters the amount of linking on short transients. 100% is equal on both sides, but then you can dial down the percentage of linking from there. When working with surround, these control knobs change to add functionality for the C (center) and LFE (subwoofer) channels. When these buttons are engaged, the C and LFE channels are integrated into the linking process.
One feature that is supported by most DAWs, but not by most NLEs, is side-chaining. This is a method by which the dynamics of one track control the compression/limiting being applied to a different track. For example, you could apply the limiter to a music track, but use a voice-over track as the side-chain input. This technique can be used to duck the music under the voice every time the person speaks.
Honestly, I’m not a huge fan of music ducking in the first place, because I don’t think it sounds good compared to riding the levels manually. However, it is available. I tested this with the Pro-L 2 in Logic Pro. Quite frankly, using the same process and the native Logic Pro compressor yielded more pleasing results. That’s not surprising. Although compressors and limiters are audio cousins, they do process audio a bit differently. Since it’s not a method I use anyway, it wasn’t a big deal, but still worth noting.
FabFilter Pro-L 2 offers a lot of depth and you really need to go through the user guide to fully appreciate its intricacies. That being said, it’s super easy to use. But for me, the quality of the sound is the key. I was impressed with how hard I could drive it when I needed to and still maintain good sound and proper loudness levels. That makes it worth the price of admission.
As a developer, FabFilter Software Instruments seems to be on top of things. If you are a Mac user, these plug-ins are already Apple Silicon-compatible. That not true of every audio plug-in maker. If, like me, you work across multiple NLEs, then it’s nice to have a consistent set of plug-ins that work and sound the same regardless of which NLE I’m working in. FabFilter Pro-L 2 definitely fits that bill.
In the previous post I presented an overview of common plug-ins. Two types used in nearly every project are equalization and compression to tame volume levels and sculpt the sound. Let’s take a closer look into how each operates.
All equalizer plug-ins work with several common controls. Some EQ models have more features, but the general concepts are the same. An equalizer will boost (raise) or cut (lower) the volume of a specific frequency within the sound spectrum of the track. Some EQs feature only a single control point, while others include more – three, four, or even unlimited.
As you boost or cut volume, the audio frequencies around the control frequency are also progressively raised or lowered in what is presented as a bell-shaped curve on a frequency graph. The width of this bell is called the Q value. As you widen the curve – the Q value – more of the surrounding frequencies are also affected. A smaller Q – a tighter curve – results in a more surgical adjustment. An extremely tight Q value is often referred to as a notch, because you are only affecting that frequency and very little else. Notch settings and separate notch filters are often used to remove or reduce specific annoying background sounds in the audio.
Most multi-point EQs are designed so that the lowest and highest frequency control points are shelf controls. When you adjust the high frequency and it operates as a shelf, then everything above that frequency is rolled off. The same for a low shelf, except that the roll-off is in the other direction – lower frequencies. The slope of this roll-off can be gradual or sharp, depending on the features on the plug-in.
An extremely sharp slope at the low-end creates a high-pass filter (higher frequencies are allowed, lower frequencies are cut). An extremely sharp slope at the top is a low-pass filter. Depending on the equalizer model, a high-pass filter control may also be referred to as a low-cut control. High-pass versus low-cut is purely a semantics difference as the controls work the same. Some EQs allow the slope of the low-cut to be adjusted along with the frequency, while others leave the slope at a fixed amount.
Compressors come in more varieties and with a wider range of features than EQ plug-ins. For most, the core operation is the same. The intent is to squash audio peak levels to reduce the overall dynamic range of a track between low and high volume levels. The smoother a compressor works, the more natural and unobtrusive the compression effect is.
The threshold control determines the volume level at which the compressor starts to bite into the signal. As you lower the threshold, more of the signal is impacted. Some compressors also include an input gain control to raise the audio coming into the filter ahead of the threshold control.
The ratio control determines the amount of signal compression, i.e. gain reduction above the threshold. A 2:1 ratio means that 2dB of gain over the threshold would be reduced by half to 1dB. A 4:1 ratio would be a reduction from 4dB down to 1dB for any audio peaks that exceed the threshold.
The make-up gain control (when available) is often used to compensate for the gain reduction. When you apply a heavy amount of compression, affecting a larger range of the signal, the overall output will sound lower. Increasing the make-up gain compensates for this volume loss. However, this risks also bringing up the noise floor, since quiet portions of the tracks have been attenuated, too.
When you see the compressor settings displayed graphically, the adjustment appears as a hockey stick standing on its end. The threshold point is displayed on the graph as the point where the line bends. The angle of this bend is the ratio. The higher the ratio, the flatter the bent section of the line. A slightly curved bend is referred to as a soft knee, meaning that compression kicks in more gradually.
The response of the compressor to peaks is controlled by the attack and release adjustments. Set a fast attack time and the compressor will react quickly to peaks. A slow release time means that bite of the gain reduction holds on longer before the compressor returns to a neutral effect. The attack and release times determine the characteristics of how that compressor sounds. As an example, your adjustments would be different for speech than if you were recording drums. The impact of the compressor would sound different in each case.
The lookahead setting determines how far ahead the compressor plug-in is analyzing the track in order to respond to future peak levels. But, you are balancing performance versus precision. Long lookahead times require more processing power for the computer. A very short lookahead value means that some peaks will get through. Lookahead only works when you are working with a recorded track and isn’t applicable to compression on live sources, such as in a recording session.
Multi-band compressors and limiters
There are two special types of compressors. The multi-band compressor divides the sound spectrum into several frequency ranges. This enables the user to control the amount of compression applied to different parts of the signal, such as low versus mid versus high frequencies. As we will see in Part 4, some equalizers can be paired with compression controls to create a combo plug-in of EQ coupled with multi-band compression.
Another variation is the limiter. This is a compressor that’s designed to block all volume peaks above a determined threshold. Limiters are important if you have to deliver files for broadcast or streaming services in order to stay within loudness parameters. Some editors and mix engineers will place a multi-band compressor followed by a limiter on their stereo output mix bus for this reason.
Finally, some compressors include a built-in limiter, often referred to as a brick wall limiter. This is a second stage of compression with a tighter ratio. Graphically, the slope after the knee would appear flat. The limiter threshold is designed to fully compress all peaks that exceed the set level of the compressor. Typically the limiter values would be set somewhat higher than the compressor adjustments in order to allow for some dynamic range between the two.
Audio mixers and audio editors who spend their time at the business end of a DAW certainly have a solid understanding of audio plug-ins. But often it’s something many video editors don’t know much about. Every NLE includes a useful complement of audio filter effects (plug-ins) that can also be augmented by a wide range of third party options. So it’s worth understanding what you have at your fingertips. After all, audio is at least 50% or more of most video projects. For this and the following three posts, I’ll focus on some thoughts pertaining to what video editors should know about commonly used audio filters.
Numerous audio effects have been highlighted in previous posts. I personally use various Accusonus and iZotope effects on my work, most often for audio clean-up. That’s been very important in this past year with restricted production activity. Quite a lot of my recent edit jobs worked with source material from Zoom calls and self-recorded smartphone video – all with marginal audio quality. So clean-up tools like iZotope RX have been quite important.
Since a lot of what I do is corporate in nature, the mixes are relatively simple – usually voice and music with a minimum of sound effects. Other than some clean-up processing (noise or reverb removal and so on), my most frequently used effects are equalization and compression. These tools let me shape the mix and control levels.
All audio plug-ins are the same. Or are they?
Audio effects typically come in two flavors. One group could be described as “digital” and is intended to process audio in a transparent fashion without adding tonal color on its own. The other group is considered “analog,” because these filters are intended to emulate the sound of certain analog processing equipment. Naturally, since these are software plug-ins, the processing is actually digital. However, analog-style emulations are designed to mimic the tonal qualities of classic outboard gear or of channel strip circuits built into analog consoles like Neve and SSL.
Tonal color is often created by how the audio is processed, such as the slope of the attack and release characteristics when the filter begins to affect the sound. In theory, you should be able to take a digital-style EQ and boost a frequency by a given amount and Q value (the width of the effect around that frequency). Then, if you apply a second instance of the EQ and cut (lower) that same frequency by the same dB and Q values, the two should cancel each other out and the signal should sound unaffected. An analog-style filter that has been designed to emulate certain models of peripheral gear will not be transparent if you try this same experience.
If you buy two competing digital audio plug-ins that have the same controls and features, then the way each alters the sound will likely be more or less the same. The only difference is the “skin,” i.e. the user interface. However, when you buy an analog audio plug-in, you are looking for certain sound characteristics found in current or vintage analog hardware. A developer could go the route of licensing the exact signal path from the original company. They can then legally display a branded UI that is skeuomorphic and looks just like the physical version that it represents. Waves has an entire repertoire of such effects. So if you want an SSL 4000-series E-type channel strip, they’ve got a software version for you.
The other development approach is to reverse-engineer the sound of that physical gear and release a plug-in that emulates the sound. It might be dead-on or it might only be reminiscent. The skeuomorphic interface is designed to look and feel like that gear. If you know the real device, then you’ll know what that plug-in can be expected to sound like. Apple Logic Pro has a wealth of effects that are emulations. If you want to use a Vox or a Marshall guitar amp filter, simply pick the one that features a similar faceplate. Nowhere does Logic actually call it a Marshall or a Vox, because Apple hasn’t licensed the exact circuits from the original manufacturer. Instead, they classify these as “inspired by” certain musical eras or genres.
Native versus third party effects
Audio plug-ins are installed using one of several protocols, including AAX, AU, and VST/VST3. This means that you can use the same effect in multiple host applications. However, DAWs and NLEs also install their own native effects that are only available within that single application. This can mean better performance versus third-party effects, which is especially true with current versions of Final Cut Pro and macOS.
One of my favorite native filters is the Logic compressor found in both Logic Pro and Final Cut Pro. It features seven compressor styles built into a single plug-in. The choices start with Platinum Digital, which is the digital (clean or transparent) version of this filter. The next six panes are different analog models, which are emulations of such popular outboard gear as Focusrite and DBX. There are two choices each for VCA, FET, and opto-electrical circuit designs.
Set the exact same adjustments in any of the compressor’s panes and the tonal color will vary slightly as you toggle through them. If you are unfamiliar with these, then check out some of the YouTube tutorials that explain the Logic compressor’s operation and which of the actual gear each of these panes is intended to emulate. I personally like the Studio VCA pane, which is based on a Focusrite Red compressor.