This complements the tip Better Ducking for Voiceovers and Podcasts and the tip Why I Don’t Use Compression Anymore. It applies the concept of voiceover ducking to your entire mix. Here’s the TL;DR: Insert the Pro EQ3 in a master bus, feed its sidechain from your vocal track, and adjust the Pro EQ3’s dynamic EQ to reduce the vocal frequencies in the stereo mix. When done subtly, it makes the voice stand out more, because the mix behind it stands out less.
Create a Secondary Main Bus
Inserting the Pro EQ3 in the Main bus won’t work, because the vocal goes to the Main bus. So, any dynamic EQ would affect the voice as well as the mix, which we don’t want. The solution is to create a secondary Main bus. We’ll call it the Sub Bus.
1. Select all your tracks in the mix (click on the lowest-numbered track and shift+click on the highest-numbered track).
2. Ctrl/Cmd click on the vocal track to de-select it. The other tracks should still be selected.
3. Right-click on one of the selected tracks and choose “Add Bus for Selected Channels” (fig. 1).
Figure 1: All tracks are selected except for the Vocal track (7). They’re about to be re-assigned from the Main bus to the new Sub Bus.
4. The new Sub Bus feeds the Main bus. Confirm that all track outputs go to the Sub Bus except for the Vocal track, which still goes to the Main bus. Now we can process the Sub Bus independently of the Vocal.
5. Insert a Pro EQ3 into the Sub Bus. Insert a Send from the Vocal track to the Pro EQ3’s sidechain. Fig. 2 is a simplified signal flow diagram.
Figure 2: All tracks except the Vocal go to the Sub Bus. The Sub Bus out and Vocal out go to the Main bus. A send from the Vocal track drives the sidechain of a Pro EQ3 inserted in the Sub Bus.
6. Now comes the most important part—choosing the optimum dynamic EQ settings. The goal is to add dynamic cuts at frequency ranges that correspond to the voice. The settings in fig. 3 are a good start for experimenting with this process:
7. Adjust the Threshold for each stage so the vocal peaks trigger a subtle cut. Fig. 3 shows a range of 12 dB, so the cuts shown are around -2 dB to -3 dB. That may not seem like much, but it’s sufficient to open up space for the vocal.
8. Optimize the setting for each Threshold parameter. To compare the processed and unprocessed sound, turn off the Vocal track’s Send. Don’t disable the Sidechain at the Pro EQ3 instead, or the EQ will respond to the mix’s dynamics instead of the vocal’s.
Figure 3: The violet waveform is the Vocal feeding the sidechain. The blue waveform is the stereo mix. The colored curves are the five EQ stages. Set LF and HF to Peaking mode. The wavy white line toward the center shows how much of a cut was occurring when this screen shot was taken.
9. Use the Vocal track’s Send slider to fine-tune the level going to the Pro EQ3’s sidechain.
This technique assumes the vocal has a consistent dynamic range, either from compression or from the techniques in the blog post about why I don’t use compression anymore. Otherwise, loud parts will push the background down further—but a lower level isn’t needed when the vocal is at its loudest. If the vocal’s dynamics are an issue, automate the Vocal track’s Send to reduce the amount of cut for loud vocal sections.
Because there are so many variables, there are many ways to optimize the sound:
Figure 4: Broader Q settings give less focused cuts.
Remember that this technique’s intention is to add a subtle enhancement—it’s not a cure for a mix, arrangement, or vocal that needs work. However, it can provide that extra little something that makes a vocal stand out a bit more and fit more comfortably into a mix.
Engineers sometimes advocate using high-pass filters to “clean up the low end and tighten the sound.” Others believe that because of issues inherent in highpass filters, they should be used sparingly, if at all. So, who’s right? Well, as with many aspects of audio technology, they both are in some ways. Let’s dive deeper, and then explore Studio One’s clever highpass filtering solution.
But before getting into the details, let’s look at the big picture. Joe Gilder did a video called THIS is Eating Up All Your Headroom in Your Productions where he shows how low-end energy can accumulate and reduce headroom with no sonic benefits. This is why engineers often apply a highpass filter to a final mix. So, let’s look at the sources that can lead to this accumulation of low-frequency energy, and the right way to fix them within individual tracks before they start accumulating in your final mix.
The Case for Highpass Filtering
Sometimes, low-frequency artifacts that you don’t want travel along with your audio. One potential issue is a directional microphone’s proximity effect, where getting closer to the mic boosts bass unnaturally. Hitting guitar pickups accidentally can generate low-frequency sounds. Transposing samples down by a couple octaves may put energy in the sub-sonic range and even if you can’t hear sub-sonic audio, it uses up headroom.
Highpass filtering has other applications. Too many instruments overlapping in low-frequency ranges can create a muddy sound. For example, you might want to reduce some of a guitar’s low frequencies to avoid interfering with bass. This is usually what people mean by “tightening up” the sound.
The Case Against Highpass Filtering
Analog filters produce phase shifts, but the phase shifts in highpass filters with steep slopes are particularly drastic around the cutoff frequency. Most digital filters model analog filters, so they also have phase shifts. Fig. 1 shows a typical example of phase shift in a digital highpass filter with a 100 Hz cutoff and a 24 db/octave slope. The bright line is the filter curve, the darker line shows the amount of phase shift.
Figure 1: Phase shift around the cutoff frequency of a 24 dB/octave highpass filter. The horizontal axis goes from 20 Hz to 20 kHz, so the phase shift impacts the audio well into the midrange. The vertical axis covers -180 degrees to +180 degrees of phase shift.
Wow! That looks scary, doesn’t it? It would look even scarier if the slope was steeper than 24 dB/octave. Fortunately, our hearing can’t detect phase changes—unless the audio interacts with other audio. This is why if you set a phase shifter effect’s dry/wet control to all wet or all dry with no modulation, the sound is the same. You hear the phase-shifted effect only when there’s a mix of wet and dry sounds.
When mixing, this phase shift can matter if two signals with similar waveforms are in parallel. For example, consider bass through a DI in parallel with bass through an amp, or two mics at different distances from an acoustic guitar or group of backup singers. The worst-case scenario is multiple acoustic instruments being recorded simultaneously in an acoustic space—the inevitable leakage means the various tracks will have quite a bit of audio in common.
Differing mic distances from a source will cause phase shifts anyway, but using a typical highpass filter (e.g., to reduce p-pops with vocals) will accentuate the phase shifts. In either case, if the parallel tracks have enough audio in common, when they’re mixed together phase shifts will alter the sound.
Solution #1: Don’t Worry, Be Happy
In many cases, tracks won’t be in parallel with other tracks that have similar or identical audio, so there won’t be any audible degradation from phase changes caused by a highpass filter. If using a highpass filter does introduce phase problems, cutting with a shelf or peak filter instead can reduce the low frequencies (albeit not in the same way). These filter types produce less drastic phase changes.
Solution #2: The Pro EQ3’s LLC Filter
Highpass filters with steep slopes are popular because of their ability to remove bass artifacts and tighten up the sound. But if you need to tighten up or reduce bass artifacts with multiple tracks, inserting all those highpass filters will contribute significant phase changes. Phase variations in the bass range can be a problem (as anyone who’s tried to mix in a small, untreated room knows), so it’s unfortunate that highpass filters are the best choice for solving these problems.
However, linear-phase highpass filters don’t produce phase shifts—that’s their outstanding characteristic. You can use as many linear-phase highpass filters as you want, and not introduce any phase shifts.
But that has its own complications, because linear-phase filters require a lot of processing power. Using several multi-stage, linear-phase EQs in a project can bring your CPU to its knees pretty quickly, as well as introduce an annoying amount of latency.
This is why Studio One’s solution is particularly elegant. The Pro EQ3 has a single linear-phase highpass filter stage (fig. 2). This covers the most common need for a linear-phase filter in most projects, without the CPU burden of a multi-stage linear-phase EQ where you may not need linear-phase operation for the other stages anyway.
Figure 2: The LLC linear-phase highpass filter (outlined in white) is producing the low-frequency cut outlined in light blue.
Enabling the LLC stage contributes about 31 ms of latency compared to the 50-100 ms from a typical multi-stage linear EQ, and takes care of the frequency range where you’re most likely to need a linear-phase EQ. If adding the LLC filter is still too CPU-intensive for a project, or if the latency is bothersome, you can always invoke the “Transform to Rendered Audio” option. Check “Preserve Real Time State” if you think you may need to edit the tracks effects’ settings in the future. This allows you to return to the unrendered audio, make your edits, and then re-render.
The LLC’s natural slope is 24 dB/octave, which will cover most low-frequency problems that need a solution. For general tightening, you may prefer the 12 dB/octave slope obtained by clicking the “Soft” button.
So—Highpass Filtering, Yes or No?
The answer is…both! If you need to highpass an individual track, go ahead and do it if the results sound better—you won’t need to use a linear-phase filter, so there won’t be CPU issues. But if you need to use highpass filtering with parallel tracks that have audio in common, the LLC stage is a clever solution. Furthermore, if you need a slope that’s steeper than 24 dB/octave, supplement the LLC with a shelf filter set for a mild slope. This cuts the low-frequency audio further, but won’t add as substantial a phase shift as a non-linear-phase highpass filter set to a steeper slope.
The GRAMMY-nominated artist, producer, and songwriter shows us how he uses Studio One to cook a musical idea from scratch.
Josh Cumbee is a triple threat: The LA-based artist, producer, and songwriter started releasing his own music while he was enrolled in the music business program at USC, then branched out into television and film composition before landing an engineering gig with super producer and John Legend co-writer, Toby Gad.
In 2015, he was nominated for Album of the Year at the Latin Grammy Awards and has since gone on to work with Olivia Rodrigo, Madonna, Sia, Sean Paul, Janet Jackson, Armin Van Buuren, and more. The secret to his sweeping success? “There’s so much room in the sandbox, and there’s so much flexibility with audio and with all the tools we have. And for me, I feel like I have to capture everything. I have to suck all that oxygen out of the room and put it into the idea.”
Watch this episode to see how Josh weaves lush pads and dynamic vocal stacks into bespoke percussion and found sounds with Studio One.
In this Brief Exploration, Josh Cumbee builds a richly-layered track from scratch using a dynamic range of instruments and effects from Studio One’s complete suite of included plug-ins.
Josh starts with a base layer of haunting synth pads before improvising a short, syncopated guitar riff over the top to generate the initial rhythmic pattern. “I usually just have ‘takes to layers’ on, do a couple passes, and use a little bit of quantization because I don’t want to lose that human feel…I like to get a couple takes, until I feel like I’ve felt it emotionally. Then if there’s any problem spots, I comp through from previous ones.” Then the whole composition gets a healthy dose of Studio One’s Analog Delay to generate ambience and depth.
With the melodic foundation in place, Josh drums out a beat on a tightly mic’d road case and drenches it in “juicy, fat, nasty” Red Light Distortion “to create something as bespoke as possible, and then work the samples around that, rather than having to try and make the samples fit into this world.” Then he improvises lyrics and a vocal melody using a combination of templates and onboard effects: “I’m a total sucker for harmonies, and I find it’s very inspiring – especially once you find a hook line – to play around with that. For me, the way that I’m able to do those quickly is by having a template already set up” and “for background vocals, I can instantly fix it with Melodyne.”
In the home stretch, Josh transitions to the low end with a bone-rattling bassline courtesy of Studio One’s stock monophonic subtractive synth, Mojito, and the Eris Sub 8. Then he creates avant-garde percussion samples from the sounds of clanking camera equipment he captured during setup – “I could dig through a sample bank for a thousand years, and could never find what happened in this room today” – and drenches the entire composition in additional textures and tones for a hauntingly atmospheric finish.
Any final words of wisdom? “Studio One can be intimidating because there are so many options. And when you see somebody using it in an advanced fashion, it gets a little scary; but the truth of the matter is Studio One isn’t the deep end of the pool – it’s more like an ocean. You kind of can walk in from the beach. You bring your key commands from another DAW, you can bring a lot of your workflows. You just find that there’s more depth and flexibility every step of your process that allow you to continue to level up, so you never really plateau.”
This wasn’t a conscious decision, or something I planned. But when I looked through my last few songs while seeking candidates for a book’s screenshots, I was shocked to realize there were almost no channel inserts with compressors. Curious, I went back even further, and found that I’ve been weaning myself off compressors for the past several years without even knowing it. WTF? What happened?
First of All, It’s 2023
Compressors are at their very best when solving problems that no longer exist. I certainly don’t need to control the levels of the PA installations at the 1936 Berlin Olympics (one of the first compressor applications). In the pre-digital era, compression kept peaks from overloading tape, and lifted the quiet sections above the background hiss and hum. But it’s 2023. Compression isn’t needed to cover up for vinyl or tape’s flaws. Besides, we have 24-bit A/D converters and increasingly quiet gear. Today’s technology can accommodate dynamic range.
Even though compression does bring up average levels for more punch and sometimes more excitement, to my ears (flame suit on) the tradeoff is that compression sounds ugly. It has artifacts, and adds an unnatural coloration. Furthermore, the way it reduces dynamic range takes away an emotional element we may have forgotten existed—because it’s been clouded by compression for decades. So, how can we retain the benefits of compression, yet avoid the ugly aspects?
Replacing Bass Compression
In my studio musician days, compressing bass was a given (not compressing bass may even have been against Local 802’s by-laws, haha). But I prefer light saturation. It clips off the peaks, gives more sustain because the average level can be higher, and doesn’t produce a “pop” on note attacks. (Yes, lookahead helps with compression pops, but then this neuters the attack.)
Saturation not only allows a higher average level, but adds harmonics that help bass stand out in a track. And bass needs all the help it can get, because low frequencies push speakers and playback systems to the limit. It’s amazing how much saturation you can put on bass, yet when part of a mix, the bass sounds completely clean—and you can hear all the notes distinctly.
Fig. 1 shows settings I’ve used in recent projects. RedLightDist sounds wonderful, while Autofilter (used solely for its State Space saturation) has the advantage of including a lowpass filter. So, you can trim the harmonics’ higher frequencies if needed.
Figure 1: (top) RedLightDist settings for saturation. (bottom) Autofilter taking advantage of State Space distortion.
Replacing Vocal Compression
Compression keeps vocals front and center by restricting dynamics, so the soft parts don’t get lost. But there’s a better option. Gain Envelopes and normalization allow tailoring vocal dynamics any way you want—without attack or release times, pumping, breathing, overshoot, or other artifacts. The sound is just as present and capable of being upfront in a mix as if it’s compressed. However, the vocal retains clarity and a natural vibe, because gain envelopes and normalization have no more effect on the sound than changing a channel fader’s level (fig. 2).
Figure 2: A typical vocal, before and after using a Gain Envelope to edit the level for more consistency.
Even better, while you’re editing you can also tweak mouth clicks, pops, and breaths in a way that compressors cannot. I’ve covered using Gain and Event Envelopes before, so for more info, check out the video Get Better Vocals with Gain Envelopes. Also, see the blog post Better Vocals with Phrase-by-Phrase Normalization.
I’m not the world’s greatest vocalist by any means, yet people invariably comment on how much they like my vocals. Perhaps much of that is due to not using compression, so my voice sounds natural and connects more directly with listeners.
Replacing Drum Compression
Adding compression on drums for more “punch,” as well as to bring up room ambience, is common. However, drums were the first instrument where I ditched compression in favor of limiting. Limiting keeps the peaks under control, but doesn’t change their shape and also allows for a higher average level. This brings up the body and room sound. (Note: Limiter2 is particularly good for this application. You may not have equally good results with other limiters.)
Look at the drum waveforms in fig. 3. Both have been normalized to the same peak levels, but the lower one had 5 dB of limiting. The peaks are still strong, and the average level is higher. Fortunately, this amount of limiting isn’t enough to reduce the drum’s punch. In fact, the punch and power is stronger than what I’ve been able to obtain with compression.
Figure 3: The top drum waveform is prior to limiting. The bottom one has had 5 dB of limiting. Both are normalized to the same peak value.
Although I don’t use compression much anymore for audio, I do use it as a tool. Gobs of compression can increase guitar sustain. Inserting a compressor prior to an envelope filter can restrict the dynamic range for more predictable triggering. Using two compressors in series, set to very low ratios and high thresholds, “glues” tracks and buses together well. Then again, that’s because the effect is so subtle the result doesn’t sound compressed.
But let’s get one thing straight: I certainly don’t mean this as a diss to those who like the sound of compression. It has its own sound, and it’s become a fixture in pop and rock music for a reason. Top engineers who line their walls with gold records have gotten a lot of mileage out of compression. I would never recommend that people not use compression. What I do recommend is trying other options. To hear what these techniques sound like, check out any recent music on my YouTube channel.
I’m sure my avoidance of compression is a personal bias. I’ve worked on many classical music sessions, and my gold standard for sound is live acoustic music. Neither one has anything to do with compression. So, it’s probably not surprising that even in rock and EDM productions, I strive for clarity and a natural, lifelike sound. Compression just doesn’t do that for me. But hey, it’s 2023! Now we have tools that can give us the goodness of compression, without the drawbacks.
First, a follow-up: In the October 13 tip about creating Track Presets for parallel processing, I mentioned that Track Presets can’t include buses, which is wrong. Sorry! However, the premise behind the tip is still valid. For example, by using tracks (which can record) instead of buses (which can’t), you can create Track Presets for recording tracks with effects that produce random changes in real time. Or, create Track Presets optimized to record hands-on effects control while recording the results. Another part of the tip about creating a dummy bus so that a channel fader can provide a master “send” to tracks, without sending audio to the main bus, can also be useful. Now, on to this week’s tip.
We have tools to correct pitch, and tools to correct timing. But one of the most important aspects of expressive music is phrasing, especially with vocals. Pushing a vocal slightly ahead of the beat, or lingering on a phrase a bit longer, can enhance the vocal performance.
Fortunately, the Bend Tool can alter phrasing. Of course, vocals don’t have the fast, strong transients to which bend markers usually lock. But we can place bend markers where we want them, and ignore the rhythmic grid. Think of bend markers not only as tools to correct timing, but as tools to add expressiveness to vocals by altering their phrasing.
Figure 1: Note the show/hide “eye” button toward the top. Moving a bend marker affects the audio between it and both the previous and next bend markers.
Remove Unneeded Transients, then Edit the Phrasing
Because vocal phrases don’t have defined transients, asking Studio One to detect transients won’t give the same results as percussive instruments. Often, transient pairs will be close together. You want only one of these (usually the one closest to the note beginning). Hover over the unneeded bend markers with the Bend Tool, and double-click to delete them (fig. 2).
Figure 2: In the top image, the bend markers to be eliminated are colored red. The lower image shows the result of altering the phrasing.
In this case, I wanted to tighten the timing and move some notes closer to the beat. But also look at the second bend marker from the right in fig. 2. This had nothing to do with timing, but altered a word’s phrasing to slow down the audio leading up to it, and speed up the audio after it. This led the note more dramatically into the final note.
Here’s another before and after example. The words toward the end have had their phrasings adjusted considerably.
Figure 3: The phrasing slows down toward the end to give the lyrics more emphasis. The bend marker at the far right was added to keep the final word’s length closer to its original length.
Remember that moving a bend marker affects audio before and after the marker. In the bottom of fig. 3, the added bend marker at the end of the last word prevented the audio from being too compressed. Otherwise, the word would have been shortened. In cases where you need to move an entire word earlier or later on the timeline, sometimes it’s easier to split at the beginning and end of the word, and move the Event itself rather than use bend markers.
If the transient detection isn’t useful with vocal phrasing, you’re better off placing bend markers manually. The Bend Tool options are:
Vocoder’s aren’t “normal” effects. For example, Arturia’s Vocoder V is an effect/instrument hybrid, because it’s an audio effect that includes a synthesizer driven by a MIDI input. Prior to Studio One version 6, which added instrument sidechaining, Vocoder V didn’t work in Studio One. However, as mentioned in the Studio One forum by users Feugeu1 and Tremo, Blue Cat Audio’s PatchWork could provide a workaround.
With version 6, the workaround was no longer necessary, with one exception we’ll get to shortly (which also applies to the PreSonus Vocoder). Referring to fig. 1, here’s how to use Vocoder V with Studio One:
Figure 1: Track setup for using Arturia’s Vocoder V in Studio One.
1. Create an audio track for the Modulator audio (e.g., a microphone). Input Monitor must be on for real-time mic use, but not if you’re playing back audio.
2. Drag the Vocoder V into Studio One’ Arrange View to create an instrument track.
3. Instead of assigning the modulator’s track Output to the Main bus, assign the track Output to the Vocoder V’s sidechain.
4. In the Vocoder V’s Advanced settings section, choose Voice Input and a suitable preset (e.g., Vocoder). Close the Advanced settings section.
5. Verify that the Vocoder V instrument track’s input is assigned to your MIDI keyboard, its output goes to Vocoder V, and the instrument track’s Input Monitor is enabled.
Talk into the mic or play back audio from the track, and play your keyboard to trigger the Vocoder V. You can record the Modulator audio and the MIDI data feeding the Vocoder V at the same time.
Finally, under the “it’s always something” category, sometimes calling up the Vocoder V editor from PatchWork doesn’t work. Simply copy the Vocoder instance in PatchWork, cut it, and then paste it back into where you cut it.
About Low-Latency Monitoring and Vocoders
However, there’s a potential complication. PreSonus forum user Bailatosco found that when enabling low-latency monitoring for instruments, Vocoder V can’t accept audio input with Studio One. This is also true if you want to use a virtual instrument’s audio output as a carrier signal for Studio One’s Vocoder.
If you don’t use low-latency monitoring for your instruments, you can ignore the rest of this tip. But if low-latency monitoring is essential to what you do, the workaround used prior to v6.0 with Blue Cat Audio’s PatchWork plugin provides a solution. Fig. 2 shows the track setup.
Figure 2: Track setup for using Vocoder V with Low-Latency Monitoring enabled.
1. Create an audio track. Choose your Mic (or other modulator source) as the input, and enable Input Monitor.
2. Insert Blue Cat Audio’s PatchWork in the audio track.
3. Load (not drag and drop) Vocoder V into one of the PatchWork slots.
4. Create an Instrument track. Assign its input to your MIDI keyboard, its output to PatchWork, and enable the Instrument track’s Input Monitor.
As with the previous example of using Vocoder V without low-latency monitoring, you can talk into the mic or play back audio from the track, and play your keyboard to trigger Vocoder V. Recording the Modulator audio and the MIDI data feeding the Vocoder V at the same time is also possible.
We can do a similar trick to feed an instrument carrier input into the PreSonus Vocoder, and be able to use low-latency monitoring. As with Vocoder V, the synth driving the Vocoder won’t benefit from low-latency monitoring, but any other instruments will. Fig. 3 shows the track setup.
Figure 3: Track setup for using the PreSonus Vocoder with low-latency instrument monitoring.
1. Create an audio track. Choose your Mic (or other modulator source) as the input, and enable Input Monitor. Insert the Vocoder into this track.
2. Create a second audio track, and insert Blue Cat Audio’s PatchWork.
3. Load (not drag and drop) the instrument you want to use as a Carrier into one of the PatchWork slots.
4. Assign the second track’s output to the Vocoder sidechain (as shown in fig. 3), or add a pre-fader Send and assign it to the Vocoder sidechain if you want to be able to mix in the carrier sound with the second track’s channel fader.
5. At the Vocoder, choose the appropriate sidechain input, and choose Side-chain for the Carrier Source.
6. Create an Instrument track. Assign its input to your MIDI keyboard, its output to PatchWork, and enable the Instrument track’s Input Monitor.