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Category Archives: Friday Tip of the Week


MIDI-Accelerated Delay Effects

Synchronized echo effects, particularly dotted eighth-note delays (i.e., intervals of three 16th notes), are common in EDM and dance music productions. The following audio example applies this type of Analog Delay effect to Mai Tai.

However, you can also create echoes for virtual instruments by copying, offsetting, and editing an instrument track’s MIDI note data. In the next audio example, the same MIDI track has been copied and delayed by an 8th note (fig. 1).

Figure 1: The Mai Tai Echo track adds delay by copying and offsetting MIDI note data.

In the original instrument track, the MIDI note data velocities are around 127. These values push the filter cutoff close to maximum. Reducing the copied Echo track’s velocity to about half creates notes that don’t open up the Mai Tai’s filter as much. So, the MIDI-generated delay’s timbre is different.

The next track copies the original track again. But this time, the notes are transposed up an octave, and offset by a dotted eighth-note compared to the original (fig. 2). For this track, the velocities are about halfway from the maximum velocity.

Figure 2: The final echo track offsets timing and transposes pitch.

Now we have our “MIDI-accelerated” echo effect for the virtual instrument’s notes, which are still processed by the original Analog Delay effect. The final audio example highlights how the combination of analog delay and MIDI note delay evolves over 8 measures.

These audio examples are only the start of what you can do with MIDI echo by offsetting, transposing, and altering note velocities. You can even feed the different MIDI tracks into different virtual instruments, and create amazing polyrhythms. But why stop there? Hopefully, this blog post will inspire you to come up with your own signature variations on this technique.

Friday Tips in the Real World: the Sequel

This is a follow-up to the Friday Tips in the Real World blog post that appeared in 2020. It was well-received, so I figured it was time for an update.

Although many of the Friday Tips include an audio example of how a tip affects the sound, that’s different from the real-world context of a musical production. So, this blog post highlights how selected tips were used in my recent music/video project, Unconstrained. The project was recorded, mixed, and mastered entirely in Studio One.

As you might expect, the workflow-related tips were used throughout, as were some of the audio tips. For example, all the vocals used the Better Vocals with Phrase-by-Phrase Normalization technique, and all the guitar parts followed the Amp Sims: Garbage In, Garbage Out tip. The Chords Track and Project Page updates were crucial to the entire project as well.

The links take you to specific parts of the songs that showcase the tips, accompanied by links to the relevant Friday Tip blog post. If you’re curious about specific production techniques used in the project, whether they’re included in this post or not, feel free to ask questions in the comments section.

One reader’s comment for the Lead Guitar Editing Hack blog post mentioned how useful this technique is. If you missed it the first time, here’s what it sounds like applied to a solo. Attenuating the attack gives the melody line a synth-like, otherworldly sound. Incidentally, if you back up the video to 6:20, the cello sounds are from Presence. I tried the “industry standard” orchestra programs, but I liked the Presence cellos more. Also, I used the “time trap” technique from the Fun with Tempo Tracks post to slow the cellos slightly before going full tilt into the solo.

The Magic Stereo blog post described a novel way to add motion to rhythm parts, like piano and guitar. This excerpt uses that technique to move the guitar in stereo, but without conventional panning. Later on in the song, the drums use Harmonic Editing to give a sense of pitch. The post Melodify Your Beats describes this technique. But in this song, the white noise wasn’t needed because the drums had enough of a frequency range so that harmonic editing worked well.

I wrote about the EDM-style “pumping” effect in the post “Pump” Your Pads and Power Chords, and it goes most of the way through this song. The reason why I chose this section is because the solo uses Presence, which I think may be underrated by some people.

Another topic that’s dear to my heart is blues harmonica, and it loves distortion—as described in the blog post Blues Harmonic FX Chain. It’s wonderful how amp sims can turn the thin, reedy sound of a harmonica into something with so much power it’s almost like a brass section. However, note that this example uses a revised version of the original FX Chain, based on Ampire. (The revised version is described in The Huge Book of Studio One Tips and Tricks.)

The blog post Studio One’s Session Bass Player generated a lot of comments. But does the technique really work? Well, listen to this example and decide for yourself. I needed a scratch bass part but it ended up being so much like what I wanted that I made only a couple tweaks…done. For a guitar solo in the same song, I tried a bunch of wah pedals but the one that worked best was Ampire’s.

I still think Studio One’s ability to do polyphonic MIDI guitar courtesy of Melodyne (even the Essential version) is underrated. This “keyboard” part uses Mai Tai driven by MIDI guitar. The MIDI part was derived from the guitar track that’s doubling the Mai Tai. For more information, see the blog post Melodyne Essential = Polyphonic MIDI Guitar. Incidentally, except for the sampled bass and choir, all the keyboard sounds were from Mai Tai. If you’ve mostly been using third-party synths, spend some time re-acquainting yourself with Mai Tai and Presence. They can really deliver.

As the post Synthesize OpenAIR Reverb Impulses in Studio One showed, it’s easy to create your own reverb impulses for OpenAIR. In this excerpt, the female background vocals, male harmony, and harmonica solo all used impulses I created using this technique. (The only ambience on the lead vocal was the Analog Delay). Custom impulses are also used throughout Vortex and the subsequent song, What Really Matters (which also uses the Lead Guitar Hack for the solo).

I’m just getting started with my project for 2023, and it’s already generating some new tips that you’ll be seeing in the weeks ahead. I hope you find them helpful! Meanwhile, here’s the link to the complete Unconstrained project.

The Surprising Channel Strip EQ

Announcement: Version 1.4.1 of The Huge Book of Studio One Tips and Tricks is a free update to owners of previous versions. Simply download the book again from your PreSonus account, and it will be the most recent version. This is a “hotfix” update for improved compatibility with Adobe Acrobat’s navigation functions. The content is the same as version 1.4.

Studio One has a solid repertoire of EQs: the Fat Channel EQs, Pro EQ3, Ampire’s Graphic Equalizer, and the Autofilter. Even the Multiband Dynamics can serve as a hip graphic EQ. With this wealth of EQs, it’s potentially easy to overlook the Channel Strip’s EQ section (fig. 1). Yet it’s significantly different from the other EQs.

Figure 1:  The Channel Strip plug-in looks basic, but the EQ hides a secret talent.

Back to the 60s

In the late 60s, Saul Walker (API’s founder) introduced the concept of “proportional Q” in API equalizers. To this day, engineers praise API equalizers for their “musical” sound, and much of this relates to proportional Q.

The theory is simple. At lower gains, the bandwidth is wider. At higher gains, it becomes narrower. This is consistent with how we often use EQ. Lower gain settings are common for tone-shaping. Increasing the gain likely means you want a more focused effect.

The concept works similarly for cutting. If you’re applying a deep cut, you probably want to solve a problem. A broad cut is more about shaping tone. Also, because cutting mirrors the response of boosting, proportional Q equalizers make it easy to “undo” equalization settings. For example, if you boosted drums around 2 kHz by 6 dB, cutting by 3 dB produces the same curve as if the drums had originally been boosted by 3 dB.

Proportional Q also works well with vocals and automation. For less intense parts, add a little gain in the 2-4 kHz range. When the vocal needs to cut through, raise the gain to increase the resonance and articulation.

How to Enable Proportional Q

The Channel Strip’s Adapt Q button converts the response to proportional Q. Let’s look at how proportional Q affects the various responses.

High and Low Stages

The High band is a 12 dB/octave high shelf. In fig. 2, the Boost is +15 dB, at a frequency of 2.5 kHz. The top image shows the EQ without Adapt Q. The lower image engages Adapt Q, which increases the shelf’s resonance.

Figure 2: Adapt Q’s effect on high-shelf resonance with 15 dB of gain.

Fig. 3 shows what happens with 6 dB of gain. With Adapt Q enabled, the Q is actually less than the corresponding amount of Q without Adapt Q.

Figure 3: Adapt Q’s effect on high-shelf resonance with 6 dB of gain.

Cutting flips the curve vertically, but the shape is the same. With the Low shelf filter, the response is the mirror image of the High shelf.

Midrange Stage

The Midrange EQ stage has variable Gain and Frequency. There’s no Q control, but the filter works with Adapt Q to increase Q with more gain or cut (fig. 4).

Figure 4: The top image is the standard EQ curve. The lower image engages Adapt Q.

With 6 dB Gain, the Q is essentially the same, regardless of the Adapt Q setting (fig. 5).

Figure 5: The top image is the standard EQ curve. The lower image engages Adapt Q.

Finally, another Adapt Q characteristic is that the midrange section’s slope down from either side of the peak (called the “skirt”) hits the minimum amount of gain at the same upper and lower frequencies, regardless of the gain. This is different from a traditional EQ like the Pro EQ3, where the skirt narrows with more Q (fig. 6).

Figure 6: The top image shows a Q of 1, while the bottom image has a Q of 8. Note that its skirt is much narrower.

Perhaps best of all, the Channel Strip draws very little CPU power. So, if you need more stages of EQ, go ahead and insert several Channel Strips in series, or in parallel using a Splitter or buses. And don’t forget—the Channel Strip also has dynamics 😊!

Set Up the Quantum Interface Preamps with One Track

A single Automation track can set up a session’s preamp levels and phantom power in the Quantum interface, as well as the older Studio 192. So, you can stop taking the time to reset preamp levels if you do lots of different sessions—let Studio One set up the preamps whenever you call up a specific song.

For example, I mostly use three vocal mics. However, their optimum gain settings vary for narration, music vocals, or recording my main background vocalist—who needs different gain settings depending on whether she’s doing upfront vocals, or ooohs/ahhhs.

To call up specific preamp levels for different songs, simply create an automation track (or tracks) at the song’s beginning. Then, when you first hit record, the track sets up levels and (with Quantum and Studio 192) phantom power on/off for up to 8 channels. The next time you call up that song, the mics will be at the right levels, with phantom power set as desired. Here’s how to do it.

1. In Universal Control, under MIDI Control, select Internal. Or, choose Enabled if you also want to be able to control Quantum from an external controller.

2. Choose Studio One > Options (Windows) or Preferences (macOS), and select External Devices.

3. Select Add. Choose New Instrument.

4. For Receive From, choose Quantum MIDI In. For Send To, choose Quantum MIDI Out. Also tick MIDI Channels 1 – 8 (fig. 1). Then, click OK.

Figure 1: Setting up Quantum as an External Device.

5. Create an Automation track. To show automation, type keyboard shortcut A.

6. From the track’s automation drop-down menu, click on Add/Remove. In the right pane, unfold the External Devices folder, then unfold the folder for the Channel that corresponds to the preamp you want to control (Channel 1 = Preamp 1, Channel 2 = Preamp 2…Channel 8 = Preamp 8). See fig. 2.

Figure 2: How to locate the automation parameters you need to add.

7. For each preamp you want to set up, add CC7 (this controls preamp volume) and CC14 (controls phantom power). For example, I typically set up channels 6, 7, and 8. After adding these continuous controllers, the Automation menu’s right side looks like fig. 3 (except with your specific channel numbers). Click on Close after making your selections.

Figure 3: Assignments are complete for preamps 6, 7, and 8.

8. To set up the preamp levels, choose the parameter you want to program in the Automation track’s drop-down menu. Note that in the documentation, the phantom power control settings are reversed. The correct values for CC14 are 0 to 63 = Off, and 64 to 127 = On. To set the preamp level, with Universal Control open, adjust the envelope for the desired preamp gain reading (you can also see the level on the Quantum’s display).

9. Set the initial level and phantom power parameter values for the chosen preamps. Now your automation track will reproduce those settings, exactly as programmed, the next time you open the song. Given that I do voiceover or narration for at least one video a week, I can’t tell you how much time that saves—I load my narration template, and don’t even have to think about adjusting levels before hitting record.

Getting Fancy

A cool trick is to reserve a song’s first measure for doing the setup. Turn on phantom power at the song start, but fade up the volume to the desired level after the phantom power is on. That avoids power-on spikes from the mics. However, when adjusting the level envelope, the preamp knobs change only if you adjust the left-most node. So, set the preamp level you want with this node, then move it to the right on the timeline. Create another node at 0 that fades up to the node you moved, which sets the final volume.

Another trick is to have more than one automation track. For example, on most songs I have a setup track for me, and a setup track for the background singer. When she does overdubs, I turn my automation track to Off, and set her automation track to Read so it sets her levels.

Coda: Windows Meets Thunderbolt

The first time I tried Quantum on a Mac, it worked perfectly. With Windows, well…it’s Windows. My computer is a PC Audio Labs Rok Box (great machine, by the way) with dual Thunderbolt 3 ports. I used Apple’s TB3-to-TB2 adapter—no go. I found a new Thunderbolt driver for the motherboard, and asked PC Audio Labs tech support about whether I should install it. They advised doing so, and said if I had problems, they’d bail me out. But after installation, the Quantum’s power button’s color was still Unhappy Red instead of Happy Blue. I was about to contact support again, but stumbled on a program in the computer called Thunderbolt Control Center. I opened it, which showed Quantum was connected—but I hadn’t given the computer “permission” to connect. So, I gave permission. With its new-found freedom, the Quantum burst into its low-latency glory.

The moral of the story: Thunderbolt has many variables on Windows than macOS. But as with life itself, perseverance furthers.

The “Drenched” Chorus

Studio One’s chorus gives the “wet” sound associated with chorus effects. But I wanted a chorus that went beyond wet to drenched—something that could swirl in the background of a thick arrangement, and shower the stereo field like a sprinkler system. Check out the audio example: the first part is the Drenched Chorus, and the second part is the standard Chorus.

This FX Chain for Studio One Professional (see the download link at the end) supercharges the wet sound by inserting two choruses in parallel, and reversing the phase for one of them (fig. 1). This cancels any dry sound, leaving only the animation from the stereo chorus. A Mixtool provides the phase reversal. Another Mixtool at the input adds gain, to compensate for the level that’s lost through phase cancellation.

Fig. 1: FX Chain for the Drenched Chorus.

The Choruses are essentially set to the default preset, but with Low Freq at minimum and High Freq at maximum for the widest frequency response. Fig. 2 uses a phase meter to compare the Drenched Chorus (left) with the standard Chorus (right), playing the same section of a guitar part.  Note how the Drenched Chorus puts a lot of energy into the sides, which accounts for the big stereo image. Meanwhile, the center has less level than the standard chorus, due to any vestiges of dry signal being removed.

Figure 2: Drenched Chorus (left), and standard Chorus (right).

This effect is designed for stereo playback, but note that in fig. 2, the Drenched Chorus’s correlation is negative. Normally you want to avoid this, because audio with negative correlation will cancel when played back in mono. However, the correlation swings wildly between positive and negative, so it’s not much of an issue. With mono playback, all that happens is a slight level loss due to occasional negative correlations. The effect still sounds like a chorus, although of course you lose the cool stereo effects.

How to Use It

Figure 3: Macro controls for the Drenched Chorus.

Download the FX Chain, and drag it into a channel’s insert. The Macro Controls (fig. 3) affect only Chorus 2. Here’s what they do:

  • Delay: Set this to 9.00 for maximum cancellation. The sound is somewhat like a combination of chorusing and flanging. Offsetting from this time increases the chorus effect. The maximum Drenched Chorus effect occurs between approximately 7 and 11 ms.
  • LFO Speed: I prefer settings below 0.30 Hz, but higher settings have a bit of a rotating speaker vibe.
  • LFO Width: More Width increases the chorusing effect. If you turn this up, I recommend keeping LFO Speed below 0.30 Hz.
  • LFO Shape: Setting this to Triangle uses the same shape as the other Chorus. Sine gives a subtly different sound.
  • Stereo Width: Extends the stereo image outward when turned clockwise.
  • Exempt Lows: This turns up the Low Freq filter, which reduces cancellation at those frequencies. Use this when you want more of the direct sound instead of maximum drenching.
  • Exempt Highs: This turns down the High Freq filter, which reduces cancellation at those frequencies. Personally, I leave both controls all the way down for maximum moisture, but turn them up if you want the track to swim a little less in the background.

Download the FX Chain below!

Solve Vocal Problems with the De-Esser

Studio One offers several ways to “de-ess” excessive sibilants (“s” sounds). De-essing combines compression and EQ. The EQ focuses on the frequency range where sibilants are most prominent. Then, compression reduces this range’s level when sibilants are present. Prior to version 6, using Multiband Dynamics was the best way to do de-essing.

With version 6, the Pro EQ3’s new dynamic EQ functionality is excellent for reducing ess sounds. However, the equally new De-Esser (fig. 1) is designed specifically for the job of fixing excessive sibilance, quickly and effectively.

To get the most out of the De-Esser, note that some of the parameters work together as a team. So, alternating edits between some controls is often the best way to optimize the effect.

Figure 1: The De-Esser can reduce the intensity of both ess and shh sounds.

Getting Started

Ess sounds tend to be short. By the time you’ve started to tweak a parameter, the ess sound has likely already ended. So, for easy editing, create a short loop on a word with the prominent ess sound.

Frequency and Listen

1. Enable Listen.

2. Vary Frequency until you find the frequency where the ess sound is most prominent.

Solo and S-Reduction

3. After identifying the frequency, enable Solo. You’ll hear only the ess sound whose volume is being reduced.

4. Adjust S-Reduction to get a feel for the optimum ess reduction amount. At 0.00 dB, there’s no reduction. At ‑60.00, sounds in addition to the ess sound will likely be reduced.

5. Next, turn off Solo, and adjust S-Reduction in context with the looped word. Less negative S-Reduction values concentrate on reducing the ess sound’s initial transient. More negative values reduce more of the ess sound past the initial transient.

6. Do a final Frequency parameter check to optimize the high-frequency response with de-essing. For example, you may be able to raise the frequency to retain more highs, yet still have effective ess reduction.

The metering is helpful in optimizing the De-Esser’s settings. The orange meter shows the amount of reduction. The blue meter shows the input level.

How to Use the Shape Parameter

Ess sounds cover a fairly small range of high frequencies. The Narrow Shape is best for this application because it compresses a narrow band. Frequencies above and below that band remain untouched.

In addition to ess sounds, the De-Esser can also reduce “shhh” sounds (e.g., like the shh sound in “action” or “compression”). Shh sounds cover a wider range of frequencies, and often require a lower Frequency setting. For these sounds, the Wide Shape splits the audio into high and low bands, and processes the entire high band.

You may need to do two passes, one with a Wide Shape for shh sounds, and one with a Narrow Shape for ess sounds. Be conservative with the settings for the two passes, because the changes will reinforce each other.

How to Use the Range Parameter

This parameter is the De-Esser’s unsung hero. Setting Range to Full allows the full amount of reduction dialed in by S-Reduction. Gentle Range restricts reduction to ‑6 dB.

The Gentle setting is useful for more than just guaranteeing a subtler effect. With the Gentle Range enabled, you can dial in huge amounts of S-Reduction. This allows processing as much of the ess or shh sound as possible, not just the initial transient. However, limiting the amount of reduction to -6 dB prevents the amount of reduction from being objectionable.

Final Note

The De-Esser is not just for singing, but also podcasts, voiceovers, and narration. It can even reduce harshness with amp sims, as described in De-Esser Meets Amp Sims. And it can probably do other things that are yet to be discovered!

EZ Vocal Plosive Control

No matter how carefully you set up a mic’s pop filter, some pops are bound to get through the filter with vocalists who sing close to the mic. But you don’t need to redo or punch the vocal—let’s explore several ways to fix these problems in the mix.

Pro EQ3 dynamic EQ. Dynamic EQ is a fast, simple way to reduce pops (fig. 1):

1. Enable the low-frequency (LF) stage. Click the D button to reveal the dynamic EQ parameters.

2. Choose a 12 or 24 dB/octave low-frequency shelf.

3. Start with a substantial negative Range (e.g., ‑12 to ‑20).

4. Set Thrs. (threshold) to 0.00.

5. Loop the vocal section with the plosives. Adjust the low-shelf frequency so that the low-frequency attenuation affects only the plosives, not the voice’s usual warmth.

Figure 1: Dynamic EQ can control plosives smoothly and automatically.

This technique also allows for a cool trick. Increase the Q slightly. When the plosives are being reduced, the increased Q adds a slight boost just above the shelf’s corner frequency. This gives some extra warmth to compensate for the lows being reduced.

Although this technique is simple to set up and works on an entire track, it may not be effective enough with super-nasty pops. The following methods are labor-intensive but can annihilate pretty much any plosive.

Split the Event before and after the pop, then insert Low Cut EQ as an Event FX. Roll off the low frequencies with the Event FX EQ (fig. 2). The only drawback is that if there are clicks when transitioning in or out of the isolated event, you’ll need to add crossfades or fades.

Figure 2: The purple Event is the isolated plosive. A Pro EQ3 LC filter is being applied as an Event FX (outlined in white) in the Inspector. Note the LC frequency setting in the upper left.

Split just before the pop, then fade in over the beginning (fig. 3). This reduces the level of the pop’s most prominent part—the beginning. For the best results, you need to find just the right split point prior to the pop, and carefully edit the fade shape.

Figure 3: The Event is the same as in fig. 2, but a fade is reducing the plosive’s attack.

For a solution that fixes plosives and sibilant (“ess”) sounds, check out The Vocal Repair Kit blog post. It’s also a tip in The Huge Book of Studio One Tips & Tricks v1.4 (see page 174).

15 Free “Analog” Cab IRs for Ampire

This week, I wanted to give y’all a little gift: 15 “analog cab” IRs that provide alternate User Cabinet sounds for Ampire. Just hit the download link at the bottom, and unzip.

If you’re not familiar with the concept of an analog cab, it’s about using EQ (not the usual digital convolution techniques) to model a miked cab’s response curve. This gives a different sonic character compared to digitally-generated cabs. (For more information, see Create Ampire Cabs with Pro EQ2.) An analogy would be that convolution creates a photograph of a cab, while analog techniques create a painting.

The 15 impulse responses (IRs) in the Ampire Analog Cab IRs folder were made by sending a one-sample impulse through EQ, and rendering the result. This process creates the WAV file you can then load into Ampire’s User Cabinet. The IRs include the following cab configurations: 1×8, 1×10, (4) 1×12, (3) 2×12, 4×10, and (5) 4×12.

How to Use Analog Cabs

  • The simplest application is dragging an analog cab IR into Ampire’s User Cabinet image.
  • To create cab stacks, insert different cabs in the User Cabinet’s three Mic slots. Vary their mix with the Mic Edit Controls.
  • Layer two Ampires, one with a convolution-based cab impulse, the other with an analog cab impulse. This gives a “best of both worlds” sound.
  • Create stereo analog cabs that work with a single Ampire User Cabinet. Insert different analog cab IRs in two tracks, pan them oppositely, then export the mix (fig. 1 shows a typical setup). Drag the Event created by the export into Ampire’s User Cab. Note that the impulse response WAV files are very short—only 2,048 samples.
Figure 1: How to create stereo cabs from the analog cab impulses.

In any event, whether you go for individual impulses, layering, or creating stereo impulses, I think you’ll find that “analog” cab IRs extend Ampire’s sonic palette even further. And if you have any questions, or feedback on using analog cabs, feel free to take advantage of the Comments section!

Download the Ampire Analog Cab IRs.zip file below:

Higher-Def Amp Sim Sounds: The Sequel

Many of these tips have their genesis in asking “What if?” That question led to the Higher-Def Amp Sim Sounds blog post, which people seemed to like. But then I thought “What about taking this idea even further?” Much to my surprise, it could go further. This week’s tip, based on the Ampire High Density pack, is ideal for increasing the definition and articulation of high-gain and metal amp sims.

Fig. 1 shows the FX Chain (the download link is available at the end of this post). The Splitter is in Channel Split mode. If your guitar track is mono so it doesn’t have two channels, change the track mode to stereo and then bounce the Event to itself. This creates a dual mono track, which is optimum for this application.

Figure 1: Effects chain for super-focused multiband processing.

With traditional multiband processing, each band represents a range of frequencies. Distorting a limited range of frequencies reduces intermodulation distortion. The result is a more defined, articulated sound quality.

Fig. 1 implements a variation on multiband processing. It has four amps, but inserts Ampire’s Ten Band Graphic Equalizer before each amp. The graphic EQ sends two narrow frequency bands into each amp. Choosing frequency bands that are as far apart as possible reduces intermodulation distortion even further than standard multiband processing.

Referring to fig. 2, two bands in each graphic EQ are at +6 dB. The others are all at 0. Note how the various EQs offset the bands to different frequencies.

Figure 2: These graphic EQ settings send different frequency bands to different amps.

The Dual Pan plug-ins create a stereo image. With a traditional multiband setup, I tend to pan the low- and high-frequency bands to center, and spread the lower mids and upper mids in stereo. That doesn’t apply here, because there aren’t wide frequency ranges. Use whatever panning gives a stereo image you like.

A waveform is worth a thousand words, so check out the audio example. The first half is guitar going through Ampire’s German Metal amp sim. The second half uses this technique, with the same guitar track and amp sim settings. I think you’ll hear quite a difference.

Can This Be Taken Even Further?

Yes, it can—I also tried using eight splits. Because the Splitter module handles a maximum of five splits, I duplicated (complete) the track with the FX Chain, and fed both tracks with the same guitar part. The 31.2 Hz and 16 kHz bands aren’t particularly relevant, so I ignored those and fed one band from each EQ into an amp. As expected, this asks quite a bit of your CPU. Consider transforming the track to rendered audio (and preserving the realtime state, in case you need edits in the future).

However, I’m not convinced I liked the sound better. That level of definition seemed a little too clean for a metal amp sim. Sure, give it a try—but I feel the setup in this tip is the sweet spot of sound quality and convenience.

Download the FX Chain below!

Making Sense of Custom Colors

Over four years ago, the blog post Colorization: It’s Not Just about Eye Candy covered the basics of using color. However, v6.1’s Custom Colors feature goes way beyond Studio One’s original way of handling colors.

The Help Viewer describes Custom Color operations, so we’ll concentrate on the process of customizing colors efficiently for your needs. For example, my main use for colors is to differentiate different track types (e.g., drums, synth, loops, voice, guitar, etc.). Then, changing the color’s brightness or saturation can indicate specific attributes within a track group, like whether a track is a lead part or background part, or whether a part is finished or needs further editing.

Opening the Custom Colors window and seeing all those colors may seem daunting. But as you’ll see, specifying the colors you want is not difficult.

What Are Hex, HSL, and RGB?

Electronic displays have three primary colors—red, green, and blue. Combining these produces other colors. For example, combining red and blue creates purple, while combining green and blue creates cyan. The three fields at the bottom of the expanded Custom Colors window (fig. 1) show the three main ways to define colors (left to right): Hex, HSL (Hue, Saturation, Lightness), and RGB (Red, Green, Blue). These are simply three different ways to express the same color.

RGB uses three sets of numbers, from 0 to 255, to express the values of Red, Green, and Blue. 255, 0, 0 would mean all red, no green, and no blue.

Hex strings together three sets of two hex digits. The first two digits indicate the amount of red, the second two the amount of green, and the final two the amount of blue.

HSL is arguably the most intuitive way to specific custom colors, so that’s the option selected in fig. 1.

Fig. 1: The color wheel used to create colors based on HSL.

You can think of the spectrum of colors as a circle that starts at red, goes through the various colors, and ends up back at red. So, each color represents a certain number of degrees of rotation on that wheel. The number of degrees corresponds to the Hue (color), represented by the H in HSL. Each main color is 30 degrees apart along the wheel:

Figure 2: Custom colors for 12 different track groups.

S represents the amount of saturation, from 0 to 100%. This defines the color’s vibrancy—with 100% saturation, the color is at its most vibrant. Pulling back on saturation mutes the color more. L is the luminance, which is basically brightness. Like saturation, the value goes from 0 to 100%. As you turn up luminance, the color becomes brighter until it washes out and becomes white. Turn luminance down, and the color becomes darker.

The Payoff of Custom Colors

Here’s why it’s useful to know the theory behind choosing colors. As mentioned at the beginning, I use two color variations for each group of tracks. For example, vocal tracks are green. I wanted bright green for lead vocals, and a more muted green for background vocals. For the bright green color, I created a custom color with HSL values of 120, 100%, and 50%. For the alternate color, I used the same values except for changing Saturation to 50%.

Fig. 2 shows the custom color parameter values used for the 12 main track groups. The right-most column in fig. 1 shows the main track group colors. The next column to the left shows the variation colors, which have 50% saturation. In the future, I’ll be adding more colors to the 12 original colors (for example, brown is the 13th color down from the top in fig. 1’s custom colors). Fortunately, the custom color feature lets you save and load presets.

The brain can parse images and colors more quickly than words, and this activity occurs mostly in the brain’s right hemisphere. This is the more intuitive, emotional hemisphere, as opposed to the left hemisphere that’s responsible for analytical functions like reading words. When you’re in the right hemisphere’s creative zone, you want to stay there—and v6.1’s track icons and custom colors help you do that.

But Wait…There’s More!

Don’t forget that Studio One also has overall appearance preferences at Options > Appearance. This is kind of like a “master volume control” for colors. If you increase contrast, the letters for track names, plugins, etc. really “pop.” For my custom colors, increasing the overall Luminance emphasizes the difference between the main track color and the variation track color.