I like anything that kickstarts creativity and gets you out of a rut—which is what this tip is all about. And, there’s even a bonus tip about how to create a Macro to make this process as simple as invoking a key command.

Here’s the premise. You have a MIDI drum part. It’s fine, but you want to add interest with a fill in various measures. So you move hits around to create a fill, but then you realize you want fills in quite a few places…and maybe you tend to fall into doing the same kind of fills, so you want some fresh ideas.

Here’s the solution: Studio One 4.5’s new Randomize menu, which can introduce random variations in velocity, note length, and other parameters. But what’s of interest for this application is the way Shuffle can move notes around on the timeline, while retaining the same pitch. This is great for drum parts.

The following drum part has a really simple pattern in measure 4—let’s spice it up. The notes follow an 8th note rhythm; applying shuffle will retain the 8th note rhythm, but let’s suppose you want to shuffle the fills into 16th-note rhythms.

Here’s a cool trick for altering the rhythm. If you’re using Impact, mute a drum you’re not using, and enter a string of 16th notes for that drum (outlined in orange in the following image). Then select all the notes you want to shuffle.

Go to the Action menu, and under Process, choose Randomize Notes. Next, click the box for Shuffle notes (outlined in orange).

Click on OK, and the notes will be shuffled to create a new pattern. You won’t hear the “ghost” 16th notes triggering the silent drum, but they’ll affect the shuffle. Here’s the pattern after shuffling.

If you like what you hear from the randomization, great. But if not, adding a couple more hits manually might do what you need. However, you can also make the randomizing process really efficient by creating a Macro to Undo/Shuffle/hit Enter.

Create the Macro by clicking on Edit|Undo in the left column, and then choose Add. Next, add Musical Functions|Randomize. For the Argument, check Shuffle notes; I also like to randomize Velocity between 40% and 100%. The last step in the Macro is Navigation|Enter. Finally, assign the Macro to a keyboard shortcut. I assigned it to Ctrl+Alt+E (as in, End Boring Drum Parts).

With the Macro, if you don’t like the results of the shuffle, then just hit the keyboard shortcut to initiate another shuffle…listen, decide, repeat as needed. (Note that you need to do the first in a series of shuffles manually because the Macro starts with an Undo command.) It usually doesn’t take too many tries to come up with something cool, or that with minimum modifications will do what you want. Once you have a fill you like, you can erase the ghost notes.

If the fill isn’t “dense” enough, no problem. Just add some extra kick, snare, etc. hits, do the first Randomize process, and then keep hitting the Macro keyboard shortcut until you hear a fill you like. Sometimes, drum hits will end up on the same note—this can actually be useful, by adding unanticipated dynamics.

Perhaps this sounds too good to be true, but try it. It’s never been easier to generate a bunch of fills—and then keep the ones you like best.

Studio One has several analysis tools, and you can use them to learn a lot about how effects work. One of my favorite test setups is inserting the Tone Generator at the beginning of the Insert Device Rack to generate white noise (a test signal with equal energy throughout the audio spectrum), the Spectrum Meter at the end of the Rack, and the device under test in between them. Here’s the Tone Generator, set to generate white noise.

As one example of the benefits of testing gear, a lot of engineers like the gentle tone-shaping qualities of Pultec’s MEQ-5 midrange equalizer. So you need an MEQ-5 plug-in, or the hardware, to obtain that effect with Studio One…right? Maybe not.

One reason for the “sound” of Pultec equalizers is that they used passive circuitry, so the EQ curves were broad. But the Pro EQ can do gentle curves as well, simply by choosing a low Q setting. The screen shot shows Pro EQ settings for a “Pultec-like” curve, with a considerable amount of boost and cut.

But how much is it really like a Pultec? Let’s find out. Choose the same settings on the Pro EQ and on an MEQ-5, then run some white noise through both, using the Spectrum Meter’s FFT analysis.

The white, smoothed line shows the average frequency response (white noise is changing constantly because it’s random, so in this case we want a smoothed, average reading). The top graph is the Pro EQ, and the bottom graph is the MEQ-5. Sure, there may be some subtle sonic differences due to the use of different filter topologies. But if you’re looking for those gentle, tone-shaping curves, the Pro EQ does just fine.

You can also find out exactly what’s going on with some effects. For example, Studio One has a phase shifter effect, and you probably know that phase shifting produces notches in the audio. But how deep are the notches? And how far apart are they? Let’s take a look.

The Phaser is set to 8 stages, so there are 4 notches. For this measurement, we want to know the instantaneous value of the notches, so the average isn’t smoothed. With depth up full, the notches are around -35 dB or so.

While we’re at it, let’s check the frequency response of guitar cabinets, so we can find out why they sound so different from each other.

The upper image shows the response for a Mesa Boogie Mark IV cabinet. Note the prominent peak in the 3 kHz range, and the rolloff below 200 Hz—now you know why those solos can really cut through a mix. Compare that with the lower image of a 1965 Fender Princeton. It has a low end bump to give a full sound, a bit of a notch around 1.5 kHz, and has more high end above 5 kHz than the Mark IV.

As to why these readings matter, suppose you recorded a guitar part, and want it to have more of a Mesa Boogie vibe. Just tweak your EQ accordingly to approximate the curve.

Using white noise for testing can also show why SSL E-series and G-series EQ curves are different, the differences between standard and constant-Q parametric stages, what really happens when you move graphic EQ sliders around, and more. If you’re curious about scratching beneath the GUI of your effects, Studio One’s analysis tools can reveal quite a bit.

Boosting the highs a bit is a time-honored mixing and mastering technique. You don’t want to overdo it, but a little brightness can give a song a lift, increase vocal intelligibility, articulate instruments better, and add some welcome “ear candy.” If you look at the spectral energy of a lot of hit records going back decades, you’ll often find a few extra dB of boost in the 7 to 10 kHz range, to add some “air” and sweetness.

So just boost the EQ a little bit, right? Well, that’s one option…but we can do better. The Air Machine FX Chain (Fig. 1) is equally at home on individual tracks in the Song page (try it judiciously on drums, drum room mics, acoustic guitar, piano, vocals, etc.), or on finished mixes in the Project page. And yes, there’s a download at the end so you don’t even have to create this yourself.

Figure 1: The Air Machine modules.

Here’s how it works. By choosing the Splitter’s Frequency Split mode (outlined in orange), the Splitter acts as a crossover that sends the high frequencies through the right-hand split. The default crossover frequency is 7 kHz, but we’ll have more to say about this later.

The high frequencies then go through the Binaural Pan processor, which spreads out the stereo imaging. Because high frequencies are very directional, this not only increases the sense of “air,” but the feeling of space. Then, the Mixtool adds the appropriate amount of high-frequency boost.

As to assigning controls to the FX Chain, see Fig. 2.

Figure 2: Under the hood of the Air Machine FX Chain.

The Air Width parameter alters the Binaural Pan Width parameter over its full range. The Air Boost control sweeps from no boost (0 dB Gain on the Mixtool) up to a maximum of about 6 dB. You really don’t want to go any higher than that.

The one control I couldn’t put in the FX Chain was the Splitter’s crossover frequency, so you may want to open up the FX Chain to alter this. The higher the frequency, the more boost you can apply without the high frequencies becoming overbearing. Sometimes, though, you might want to bring the crossover frequency down to 5 kHz or so if you want a more pronounced effect…or even up to 10 kHz for just a light dusting of high frequencies.

Yes, it really is that simple. And yes, the effect is best when used subtly. But try it—I think there’s a chance this may end up becoming one of your favorite FX Chains.

Download the Air Machine FX Chain here.

Studio One’s Autofilter has a sidechain, which is a good thing—because you can get some really tight, funky sounds by feeding a drum track’s send into the Autofilter’s sidechain. Then, use the Autofilter’s sidechain to modulate a track’s audio in time with the beat. Funky guitar, anyone?

But (there’s always a “but,” or there wouldn’t be a Friday Tip of the Week!), although this is a cool effect, a real wah pedal doesn’t start instantly in the toe-down position before sliding back to the heel-down position. Your foot moves the pedal forward, then back, and it takes a finite amount of time to do both.

The “decay-only” nature of autofilters in general is certainly useful with drums. After all, drums are a percussive instrument, and a percussive filter sweep is usually what you want. But the other day I was working on a song, and really wanted an attack/decay filter effect that was more like a real wah pedal—where the filter moved up to the peak, before moving back down again. Here’s the result.

On the Autofilter, ctrl+click on the LFO sliders to zero them out, so that the LFO isn’t adding its own signal (although of course, you can do that if you want—the 16 Step option is particularly useful if you do). The screen shot gives a good idea of a typical initial setting.

The dark blue track is the guitar, and the green track, the drum part. I often cut up tracks are that modulating other tracks, and Track 3—a copy of the main drum track—is no exception. This track’s pre-fader send goes to the Autofilter’s sidechain input. The track’s channel fader is down, so that the audio doesn’t go through the mixer. We’re using this track only to provide a signal to the Autofilter’s sidechain.

Track 2 is a reversed version of the drum part. It also has a pre-fader send that goes to the Autofilter sidechain (conveniently, you don’t need to bus signals together to send signals from multiple tracks into a Studio One effect’s sidechain). Like Track 3, the track’s channel fader is down, so that the audio doesn’t go through the mixer

The end result is that the reversed drums provide an attack time that sweeps the filter up, while the forward drums provide a decay that sweeps the filter down. So is the sound more animated than using only the forward drum part? Listen to the audio example, and decide for yourself. The first section uses the forward trigger only, while the second section adds in the attack trigger—the effect is particularly noticeable toward the end.

Comping’s goal is to piece together the best parts of multiple Takes (vocals, guitar, etc.) into a single, cohesive part. This involves Studio One’s loop recording, which repeats a section of music over and over during a looped section. You record another Take during each pass, while previous Takes are muted. Doing multiple takes without having to stop lets you get comfortable, and try different approaches. Once you have multiple versions, you audition and select the best sections.

However, when auditioning the Takes to decide which sections are best, it’s helpful to compare levels that are as similar as possible. Normalization is the right tool for this—but while it’s not yet possible to normalize individual Takes, there’s a simple solution.

1. Right-click on the main, parent Track for the Takes and choose Unpack Layers to Tracks (Fig. 1).

Figure 1: The four Takes right immediately below the parent vocal have been unpacked into four Tracks (color-coded blue).

2. Next, select all the audio in the new Tracks.
3. Type Ctrl+B and then Alt+N. This normalizes all the Tracks.
4. Right-click on each Take’s audio and choose Delete (do not delete the Take itself; see Fig. 2).

Figure 2: The Takes have been deleted. The four normalized tracks are below.

5. Select the audio in the new Tracks.
6. Drag the audio from the new Tracks up, so that they replace where the Takes were (Fig. 3).

Figure 3: The normalized Track audio now occupies the Take Lanes.

7. The empty Tracks are no longer needed, so remove them.

• And that’s all there is to it—now you can take advantage of Studio One’s Take-oriented comping tools, as well as the Listen tool (keyboard shortcut 8), with normalized audio.

If you’ve ever played a large venue like a sports arena, you know that reverb is a completely different animal than what the audience hears. You hear your instrument primarily, and in the spaces between your playing, you hear the reverb coming back at you from the reflections. It might seem that reverb pre-delay would produce the same kind of effect, but it doesn’t “bloom” the way reverb does when you’re center stage in a big acoustical space.

This week’s tip is inspired by the center stage sound, but taken further. The heart of the effect is the Expander, but unlike last week’s Expander-based Dynamic Brightener tip, the Expander is in Duck mode, and fed by a sidechain. Here’s the Console setup.

In the audio example, the source is a funk guitar loop from the PreSonus loop collection; but any audio with spaces in between the notes or chords works well, especially drums (if the cymbals aren’t happening a lot), vocals that aren’t overly sustained, percussion, and the like. I deliberately exaggerated the effect to get the point across, so you might want to be a little more tasteful when you apply this to your own music. Or maybe not…

The guitar’s channel has two sends. One goes to the FX Channel, which has a Room Reverb followed by an Expander. The second send goes to the Expander’s sidechain input. Both are set pre-fader so that you can turn down the main guitar sound by bringing down its fader, and that way, you can hear only the processed sound. This makes it easier to edit the following Room Reverb and Expander settings, which are a suggested point of departure. Remember to enable the Expander’s Sidechain button in the header, and click the Duck button.

The reverb time is long—almost six seconds. This is because we want it going constantly in the background, so that after the Expander finishes ducking the reverb sound, there’s plenty of reverb available to fill in the spaces.

To tweak the settings, turn down the main guitar channel so you can monitor only the processed sound. The Expander’s Threshold knob determines how much you want the reverb to go away when the instrument audio is happening. But really, there are no “wrong” settings—start with the parameters above, play around, and listen to what happens.

This is a pretty fertile field for experimentation…as the following audio example illustrates. The first part is the guitar and the reverb effect; the reverb tail shows just how long the reverb time setting is. The second part is the reverb effect in isolation, processed sound only, and without the reverb tail.

This is a whole different type of reverb effect—have fun discovering what it can do for you!

When you play an acoustic guitar harder, it not only gets louder, but brighter. Dry, electric guitar doesn’t have that quality…by comparison, the electrified sound by itself is somewhat lifeless. But I’m not here to be negative! Let’s look at a solution that can give your dry electric guitar some more acoustic-like qualities.

How It Works

Create an FX Channel, and add a pre-fader Send to it from your electric guitar track. The FX Channel has an Expander followed by the Pro EQ. The process works by editing the Expander settings so that it passes only the peaks of your playing. Those peaks then pass through a Pro EQ, set for a bass rolloff and a high frequency boost. Therefore, only the peaks become brighter. Here’s the Console setup.

The reason for creating a pre-fader send from the guitar track is so that you can bring the guitar fader down, and monitor only the FX Channel as you adjust the settings for the Expander and Pro EQ. The Expander parameter values are rather critical, because you want to grab only the peaks, and expand the rest of the guitar signal downward. The following settings are a good point of departure, assuming the guitar track’s peaks hit close to 0.

The most important edit you’ll need to make is to the Expander’s Threshold. After it grabs only the peaks, then experiment with the Range and Ratio controls to obtain the sound you want. Finally, choose a balance of the guitar track and the brightener effect from the FX Channel.

The audio example gets the point across. It consists of guitar and drums, because having the drums in the mix underscores how the dynamically brightened guitar can “speak” better in a track. The first five measures are the guitar with the brightener, the next five measures are the guitar without the brightener, and the final five measures are the brightener channel sound only. You may be surprised at how little of the brightener is needed to make a big difference to the overall guitar sound.

Also, try this on acoustic guitar when you want the guitar to really shine through a mix. Hey, there’s nothing wrong with shedding a little brightness on the situation!

You never know where you’ll find inspiration. As I was trying not to listen to the background music in my local supermarket, “She Drives Me Crazy” by Fine Young Cannibals—a song from over 30 years ago!—earwormed its way into my brain. Check it out at https://youtu.be/UtvmTu4zAMg.

My first thought was “they sure don’t make snare drum sounds like those any more.” But hey, we have Studio One! Surely there’s a way to do that—and there is. The basic idea is to extract a trigger from a snare, use it to drive the Mai Tai synth, then layer it to enhance the snare.

Skeptical? Check out the audio example.

ISOLATING THE SNARE

If you’re dealing with a drum loop or submix, you first need to extract the snare sound.

1. Create an FX Channel, and insert a Pro EQ followed by a Gate.
2. Add a pre-fader send from your mixed drums to the FX Channel. Aside from providing a more consistent level for triggering, a pre-fader send lets you turn down the main drum track. This way you hear only the FX Channel, which makes it easier to tweak the EQ and isolate the snare.
3. With the Gate bypassed, tune the Pro EQ to the snare frequency. Use the LC and HC with a 48 dB/octave slope to provide the preliminary isolation, then use a super-sharp bandpass setting to zero in on the snare frequency (Figure 1). The EQ’s background spectrum analyzer can help by making sure the bar in the range you’re boosting goes as high as possible. In stubborn cases, you may need to double up the bandpass filter with a second sharp bandpass filter.

Figure 1: Use the Pro EQ and Gate to extract a snare drum trigger.

4. Enable the gate, and click on Active to enable the trigger output. Set the Note and Velocity as desired (however when using noise with Mai Tai, the specific Note isn’t that critical). Adjust the Gate’s settings so that it triggers on the snare hits. Like the Pro EQ, the controls here are critical as well.

• A short attack is usually best.
• Keep release relatively short (e.g., 10 ms), unless you want to mix in some of the peaked/gated sound from this channel along with the Mai Tai and drums.
• Hold times around 50 ms can help prevent false triggering. But you can also get creative with this control. If you don’t want to trigger on hits that are close together (e.g., fills), a long Hold time will trigger on the first snare of the series, but ignore subsequent ones that fall within the hold time.

5. Insert the Mai Tai. Set its input to Gate, and enable the mixer’s Monitor button. Figure 2 shows the finished track layout.

Figure 2: Track layout for snare drum extraction.

TWEAKING THE MAI TAI

Now the fun begins! Figure 3 shows a typical starting point for a snare-enhancing sound.

Figure 3: Starting point for a cool snare drum sound with Mai Tai.

The reason for choosing Mai Tai as the sound source is because of its “Character” options that, along with the filter controls, noise Color control, and FX (particularly Reverb, EQ, and Distortion), produce a huge variety of electronic snare sounds. The Character module’s Sound and Amount controls are particularly helpful. The more you play with the controls, the more you’ll start to understand just how many sounds are possible.

BUT WAIT…THERE’S MORE!

If the snare is on a separate track, then you don’t need the Pro EQ or FX Channel. Just insert a Gate in the snare track, enable the Gate’s trigger output, and adjust the Gate Threshold controls to trigger on each snare drum hit. The comments above regarding the Attack, Release, and Hold controls apply here as well.

Nor are you limited to snare. You can isolate the kick drum, and trigger a massive, low-frequency sine wave from the Mai Tai to give those car door-vibrating kick drums. Toms can sometimes be easy to isolate, depending on how they’re tuned. And don’t be afraid to venture outside of the “drum enhancement” comfort zone—sometimes the wrong Gate threshold settings, driving the wrong sound, can produce an effect that’s deliciously “right.”

Conventional wisdom says that compared to compression, limiting is a less sophisticated type of dynamics control whose main use is to restrict dynamic range to prevent issues like overloading of subsequent stages. However, I sometimes prefer limiting with particular signal sources. For example:

• For mixed drum loops, limiting can bring up the room sound without having an overly negative effect on the drum attacks.
• With vocals, I often use a limiter prior to compression. By doing the “heavy lifting” of limiting peaks, the subsequent compressor doesn’t have to work so hard, and can do what it does best.
• When used with slightly detuned synth patches, limiting preserves the characteristic flanging/chorusing-like sound, while keeping the occasional peaks under control.
• Limiting is useful when following synth sounds with resonant filters, or with instruments going through wah or autofilter effects

THE E-Z LIMITER

Some limiters (especially some vintage types) are easy to use, almost by definition: One control sets the amount of limiting, and another sets the output level. But Studio One’s limiter has four main controls—Input, Ceiling, Threshold, and Release—and the first three interact.

If the Studio One Limiter looked like Fig. 1, it would still take care of most of your needs. In fact, many vintage limiters don’t go much beyond this in terms of functionality.

Figure 1: If Studio One’s Limiter had an “Easy Mode” button, the result would look something like this.

To do basic limiting:

1. Load the Limiter’s default preset.
2. Turn up the input for the desired limiting effect. The Reduction meter shows the amount of gain reduction needed to keep the output at the level set by the Threshold control (in this case, -1.00 dB). For example, if the input signal peaks at 0 dB and you turn up the Input control to 6 dB, the Limiter will apply 7 dB of gain reduction to keep the Limiter output at -1.00.

Note that in this particular limiting application, the Threshold also determines the maximum output level.

THE SOFT CLIP BUTTON

When you set Threshold to a specific value, like 0.00 dB, then no matter how much you turn up the Limiter’s Input control, the output level won’t exceed 0.00 dB. However, you have two options of how to do this.

• With Soft Clip off, gain reduction alone prevents the waveform from exceeding the ceiling.
• With Soft Clip on, clipping the peaks supplements the gain reduction process to keep the waveform from exceeding the ceiling (Fig. 2).

Figure 2: The left screen shot shows the waveform with the input 6 dB above the Threshold, and Soft Clip off. The right screen shows the same waveform and levels, but with Soft Clip turned on. Note how the waveform peak is flattened somewhat due to the mild saturation.

While it may sound crazy to want to introduce distortion, in many cases you’ll find you won’t hear the effects of saturation, and you’ll have a hotter output signal.

ENTER THE CEILING

There are two main ways to set the maximum output level:

• With the Threshold set to 0.00, set the maximum output level with the Ceiling control (from 0 to -12 dB).
• With the Ceiling set to 0.00, set the maximum output level with the Threshold control (from 0 to -12 Db).

It’s also possible to set maximum output levels below -12.00 dB. Turn either the Ceiling or Threshold control all the way counter-clockwise to -12.00 dB, then turn down the other control to lower the maximum output level. With both controls fully counter-clockwise, the maximum output level can be as low as -24 dB.

SMOOTHING THE TRANSITION INTO LIMITING

Setting the Ceiling lower than the Threshold is a special case, which allows smoothing the transition into limiting somewhat. Under this condition, the Limiter applies soft-knee compression as the input transitions from below the threshold level to above it.

For example, suppose the Ceiling is 0.00 dB and the Threshold is -6.00 dB. As you turn up the input, you would expect that the output would be the same as the input until the input reaches around -6 dB, at which point the output would be clamped to that level. However in this case, soft-knee compression starts occurring a few dB below -6.00 dB, and the actual limiting to -6.00 dB doesn’t occur until the input is a few dB above -6.00 dB.

The tradeoff for smoothing this transition somewhat is that the Threshold needs to be set below 0.00. In this example, the maximum output is -6.00 dB. If you want to bring it up to 0.00 dB, then you’ll need to add makeup gain using Mixtool module.

Studio One’s Limiter is a highly versatile signal processor, so don’t automatically ignore it in favor of the Compressor or Multiband Dynamics—with some audio material, it could be exactly what you need.

Some instruments, when compressed, lack “sparkle” if the stronger, lower frequencies compress high frequencies as well as lower ones. This is a common problem with guitar, but there’s a solution: the Compressor’s internal sidechain can apply compression to only the guitar’s lower frequencies, while leaving the higher frequencies uncompressed so they “ring out” above the compressed sound. (Multiband compression works for this too, but sidechaining can be a faster and easier way to accomplish the same results.) Frequency-selective compression can also be effective with drums, dance mixes, and other applications—like the “pumping drums” effect covered in the Friday Tip for October 5, 2018. Here’s how to do frequency-selective compression with guitar.

1. Insert the Compressor in the guitar track.
2. Enable the internal sidechain’s Filter button. Do not enable the Sidechain button in the effect’s header.
3. Enable the Listen Filter button.
4. Turn Lowcut fully counterclockwise (minimum), and set the Highcut control to around 250 – 300 Hz. You want to hear only the guitar’s low frequencies.
5. You can’t hear the effects of adjusting the main compression controls (like Ratio and Threshold) while the Listen Filter is enabled, so disable Listen Filter, and start adjusting the controls for the desired amount of low-frequency compression.
6. For a reality check, use the Mix control to compare the compressed and uncompressed sounds. The high frequencies should be equally prominent regardless of the Mix control setting (unless you’re hitting the high strings really hard), while the lower strings should sound compressed.

The compression controls are fairly critical in this application, so you’ll probably need to tweak them a bit to obtain the desired results.

If you need more flexibility than the internal filter can provide, there’s a simple workaround.

Copy the guitar track. You won’t be listening to this track, but using it solely as a control track to drive the Compressor sidechain. Insert a Pro EQ in the copied track, adjust the EQ’s range to cover the frequencies you want to compress, and assign the copied track’s output to the Compressor sidechain. Because we’re not using the internal sidechain, click the Sidechain button in the Compressor’s header to enable the external sidechain.

The bottom line is that “compressed” and “lively-sounding” don’t have to be mutually exclusive—try frequency-selective compression, and find out for yourself.