As the quest for expressive electronic instruments continues, many virtual instruments incorporate keyswitching to provide different articulations. A keyswitch doesn’t play an actual note, but alters what you’re playing in some manner—for example, Presence’s Viola preset dedicates the lowest five white keys (Fig. 1) to articulations like pizzicato, tremolo, and martelé.
This is very helpful—as long as you have a keyboard with enough keys. Articulations typically are on the lowest keys, so if you have a 49-key keyboard (or even a 61-note keyboard) and want to play over its full range (or use something like a two-octave keyboard for mobile applications), the only way to add articulations are as overdubs. Since the point of articulations is to allow for spontaneous expressiveness, this isn’t the best solution. An 88-note keyboard is ideal, but it may not fit in your budget, and it also might not fit physically in your studio.
Fortunately, there’s a convenient alternative: a mini-keyboard like the Korg nanoKEY2 or Akai LPK25. These typically have a street price around $60-$70, so they won’t make too big a dent in your wallet. You really don’t care about the feel or action, because all you want is switches.
Regarding setup, just make sure that both your main keyboard and the mini-keyboard are set up under External Devices—this “just works” because the instrument will listen to whatever controllers are sending in data via USB (note that keyboards with 5-pin DIN MIDI connectors require a way to merge the two outputs into a single data stream, or merging capabilities within the MIDI interface you’re using). You’ll need to drop the mini-keyboard down a few octaves to reach the keyswitch range, but aside from that, you’re covered.
To dedicate a separate track to keyswitching, call up the Add Track menu, specify the desired input, and give it a suitable name (Fig. 2). I find it more convenient not to mix articulation notes in with the musical notes because if I cut, copy, or move a passage of notes, I may accidentally edit an articulation that wasn’t supposed to be edited.
So until you have that 88-note, semi-weighted, hammer-action keyboard you’ve always dreamed about, now you have an easy way take full advantage of Presence’s built-in expressiveness—as well as any other instrument with keyswitching.
As you’ve probably figured out, these tips document something I needed, and the solution. If you’ve ever put together an album or collection of songs, you know how difficult it can be to match levels—which I was reminded of all too clearly while preparing the album Joie de Vivre for upload to my YouTube channel. It’s rock-meets-EDM, and is done as a continuous mix that includes not just songs, but transitions. So, all the levels had to be matched very carefully. Fortunately, Studio One’s Project Page made it easy.
The key was using the Project Page’s LUFS meter readings; for a complete explanation of LUFS, please check out the article I wrote for inSync magazine. In a nutshell, it’s a way to measure audio’s perceived level that’s more sophisticated than the usual average, VU, or peak readings. If two songs have the same LUFS reading, they’ll be perceived as having a similar (if not the same) level.
This measurement standard was created in response to issues involved in broadcasting and streaming services, and also in part as a backlash against “the loudness wars.” For example, YouTube doesn’t want you to have to change the level every time a video changes, so they’ve standardized on making all audio -13 LUFS. It doesn’t matter if you squash your master recording until it looks like a sausage, YouTube will adjust the perceived level so that it can slip into a playlist with something like a live acoustic jazz recording.
In Studio One’s Project Page, the Loudness Information section for each song (Fig. 1) shows a song’s LUFS as well as readings for the RMS average level (somewhat like a VU meter) and True Peak, which indicates not just peaks, but whether any peaks are exceeding the maximum headroom on playback, and by how much. The Loudness Information can come from before or after the track’s effects, so to see how editing these alters the LUFS reading, choose the Post FX tab.
Now that we know how to measure levels, here’s one way to tweak them for consistency. We’ll assume you want something fairly compressed/limited, but not enough to become collateral damage in the loudness wars.
For each track (likely all of them) that needs to be set to a certain LUFS measurement, insert the Tricomp compressor followed by the Limiter. The screen shot shows my preferred Tricomp settings, but note that the optimum Compress knob setting depends on the material. You don’t want to compress too much, because the limiter will do most of the leveling anyway. If the gain reduction peaks reach the last “s” in “Compress” on the Limiter’s Reduction meter, you probably won’t hear too many artifacts, but you might not want to go any higher.
Next, decide what your target LUFS reading should be. As a very general rule of thumb, most rock songs are around -8 to -10 LUFS. -11 to -14 LUFS is considered as having a decent amount of dynamics, while classical music hangs out around -23 LUFS. Of course, this is all subjective—you can choose whatever level sounds “right.”
Now turn up the Limiter’s input control. The Loudness Information label will change to “Update Loudness.” Click on this; Studio One will analyze the track, and show the LUFS reading. (Note: You can force a reading by right-clicking on the song in the track column, and choosing “Detect Loudness.”)
Adjust the limiter Input level, then update the loudness. If the LUFS is below your target, turn up the Input. If the result is higher than the desired LUFS, turn down the Input. It takes a little trial and error, but eventually you’ll hit the target.
With the Tricomp and Limiter, once you get much above -13 LUFS you can “hear” the limiter because it’s stereo. With a phase-linear multiband maximizer like the Waves L3 Multimaximizer, you can push for higher LUFS readings while still sounding reasonably free of artifacts. Still, I wouldn’t want to go much above -10 LUFS—but as always, that’s a subjective call and there are no rules. If you like the way it sounds, that’s what matters.
However, be aware that even slight tweaks can make a difference, especially with the Tricomp. The Tricomp and the Limiter work together, and you can fine-tune the sound by fine-tuning each processor. For example, having Knee up all the way on the Tricomp gives more perceived loudness, and a narrower dynamic range…which may or may not be what you want. Turning on Autospeed also makes a difference.
When you listen to Joie de Vivre, I think you’ll hear that it benefited considerably by being adjusted in Studio One to a consistent LUFS reading. There’s a decent amount of dynamics, but the average perceived level of all the cuts is very consistent…and that’s what this tip all about.
Humbuckers are known for a big, beefy sound, while single-coil pickups are more about clarity and definition. If you want the best of both worlds, you can warm up a soldering iron, ground the junction of the humbucker’s two coils, and voilà—a single coil pickup. But there’s an easier way: use the Pro EQ, which gives the added benefit of not losing the pickup’s humbucking characteristics.
The main difference between humbucker and single coil pickups is the frequency response. The blue line in Fig. 1 shows a humbucker’s spectral response, while the yellow line shows the same humbucker split for single-coil operation. Unlike the single-coil’s response, which is essentially flat from 150 Hz to 3 kHz, the humbucker has a bump in the 500 Hz to 2 kHz range that contributes to the “beefy” sound. Starting at 3 kHz the humbucker response drops off rapidly, while the single coil produces more high-frequencies than the humbucker from 3 kHz to 9 kHz.
Fig. 2 shows an equalizer curve that modifies a bridge humbucker for more of a single-coil response. Of course different humbucker and different single-coil pickups sound different, so this kind of EQ-based “modeling” is an inexact science. However, I think you’ll find that the faux single-coil sound delivers the distinctive, glassy character you want from a single-coil pickup. Feel free to tweak the EQ further—you can come up with variations on the single-coil sound, or “morph” between the humbucker and single-coil characteristics.
The difference between a neck humbucker and single-coil response isn’t as dramatic, but the curve in Fig. 3 replicates the neck single-coil character, and provides yet another useful variation for your guitar tone.
The bottom line is that you don’t need to break out a soldering (or void your guitar’s warranty) to make your humbucker sound more like a single-coil type—all you need is the right kind of EQ.
This signal processing setup is optimized for single-string guitar solos where you want a lot of sustain—but it has a secret ingredient that puts it ahead of typical guitar stompbox sustainers.
The compression aspect is pretty straightforward. A sustainer is all about a high compression ratio and low threshold, which are set to 20:1 and -35 dB, respectively. The sharp knee keeps the sustain going as long as possible, and a short attack time clamps down the attack. The release time isn’t too critical, although this depends on your playing style; a relatively long one (300-500 ms) usually works best.
This is one of those rare instances where you don’t want to enable the compressor’s Auto or Adaptive feature, because the goal here isn’t the most natural sound—it’s an effect. However, enable Lookahead because it helps to tame the attack.
Because of the extreme amount of compression, you’ll need about 30 dB of makeup gain to compensate for the gain reduction due to compression.
And now, the secret ingredient! With most sustainers, after the release time ends, if there’s a pause between notes you’ll hear a loud “pop” when you play a new note because of the compression kicking back in. A fast attack and lookahead helps, but it’s almost impossible to avoid some kind of nasty transient. If you follow the compressor with an amp sim, the distortion will hide the pop somewhat but it can still lead to an ugly attack.
Enter the noise gate. This doesn’t just remove hum, noise, and other low-level signals from being sustained, but the 55 ms attack time (coupled with the enabled lookahead button) means that when you hit a note after a pause, the note attack ramps up more slowly, so the compressor can “grab” the note without creating a pop (or if it does, the pop will be greatly reduced). If there’s an amp sim involved, you’ll hear a cleaner attack, and better overall sound. Note that depending on how fast you play and the compressor’s release time, you may need to shorten the Noise Gate’s Release and Hold times. In any event, when you want serious ssssssuuuuussssstttttaaaaaiiiiinnnnn for your single-note guitar solos, this is the ticket.
As with so many aspects of audio, the subject of compression presets polarizes people. The purists say there’s no point in having presets, because every signal is different, and the same compressor settings will sound very different on different sources. On the other hand, software comes with presets, and there are plenty of recording blogs on the web that dispense advice about typical preset settings. So who’s right?
And as with so many aspects of audio, they all are. If a preset works “out of the box,” that’s just plain luck. However, there are certain ranges of settings that work well in many cases for particular types of signals. In any case, the effects of compression are totally dependent on the input signal level anyway—if the threshold is set to -10, then signals that peak at 0 will sound very different compared to signals that peak at -10.
The most effective way to approach compression is to decide what effect you want the compression to accomplish, then adjust the compression settings accordingly. It’s also important to remember that compression isn’t just some monolithic effect that “squashes things.” For example, with kick and snare, compression can act just like a transient/decay shaper due to a drum’s rapid decay.
The usual goal for compressing kick is an even sound, yet one that doesn’t reduce punch. However, you have a great deal of latitude in deciding how to implement that goal.
The preset in Fig. 1 uses a fairly high ratio, and hard knee, to even out the highest levels. You want the compression to take hold relatively rapidly, but not take away from the punch. The best option is to start with the attack time at 0, and increase it until you hear the initial hit clearly (but don’t go past that point). Because a kick decays fast, release can be fast as well.
For transient shaping, slowing the attack time softens the attack. Raising the ratio increases the sustain somewhat, while making space for the attack (assuming an appropriate attack time). Between the attack and ratio controls, you can pretty much tailor the kick drum’s attack and sustain characteristics, as well as even out the overall sound. A higher threshold is another way to emphasize the attack, by letting the decay occur naturally. Lowering the threshold reduces the level difference between the attack and decay.
Snare responds similarly to kick, however with an acoustic drum kit, the kick is more isolated physically than the snare. As a result, compressing the snare has the potential to emphasize leakage. Fortunately, the snare is often the focus of a drum part. As a result, you can simply compress the snare, and accept that leakage is part of the deal. With individual, multitracked drums (including electronic drums) where leakage is not a problem, it’s still usually the snare and kick that get compression.
With snare, you may want to use a lower ratio (2:1 – 3:1) for a fuller snare sound. Or, increase the ratio to emphasize the attack more. Again, use the attack time to dial in the desired attack characteristics.
With both kick and snare, you’ll usually want a hard knee. However, the knee control is a fantastic way to fine-tune the attack—and once you have that dialed in, you’ll be good to go.
Summer may be over in the northern hemisphere, but we can still splash around. This is one of those “hiding in plain sight” kind of tips, but it’s pretty cool.
The premise: Sometimes you don’t want reverb all the time, so you kick up the send control to push something like a snare hit into the reverb for a quick reverb “splash” (anyone who’s listened to my music knows this is one of my favorite techniques). The reverb adds a dramatic emphasis to the rhythm, but is short enough that it doesn’t wear out its welcome—listen to the audio example, which demos this technique with Studio One’s Crowish Acoustic Chorus 1 drum loop.
However, although this technique is great with drums, it also works well with rhythm guitar, hand percussion, synths, you name it… even kick works well in some songs. I’m not convinced about bass, but aside from that, this has a lot of uses.
Studio One offers an easy way to produce regular splashes automatically (like on the second and fourth beats of a measure, where an emphasizing element hits). Insert X-Trem before the reverb, select 16 Steps as the “waveform,” click Sync, and choose your rhythm. The screenshot shows Beats set to 1/2 so that the reverb splash happens on 2 and 4, which in the case of the audio example, adds reverb to the snare on 2, and to the closed high-hat on 4.
And that’s pretty much it. Because the reverb is in a bus, set Mix to 100%. The 480 Hall from Halls > Medium Halls is one of my faves for this application, but hey… use whatever ’verb puts a smile on your face.
I’m not one of those people who wants to do heavy compression all the time, but I do feel bass is an exception. Mics, speakers, and rooms tend to have response anomalies in the bass range; even if you’re using bass recorded direct, compression can help even out the response for a smoother, rounder sound.
Although stereo compressors are the usual go-to for bass, I often prefer a multiband dynamics processor because it can serve simultaneously as a compressor and EQ. Typically, I’ll apply a lot of compression to the lowest band (crossover below 200 Hz or so), light compression to the low-mid bands (as well as reduce their levels in the overall mix), and medium compression to the high-mid band (from about 1.2 kHz to 6 kHz). I often turn down the level for the band above 5-6 kHz or so (there’s not a lot happening up there with bass anyway), but sometimes I’ll set a ratio below 1.0 so that the highest band turns into an expander. If there’s any hiss in the very highest band, this will help reduce it. Another advantage of using Multiband Dynamics is that you can tweak the high and low band gain parameters so that the bass fits well with the rest of the tracks.
The preset in the following screenshot gives a sound like “Tuned Thunder,” thanks to heavy compression in the lowest band. To choose a loop that’s good for demoing this sound, choose Rock > Bass > Clean, and then select 08 02 P Ransack D riff.audioloop. Insert the Multiband Dynamics processor, and start with the default preset.
As with most dynamic processing presets, the effect is highly dependent on the input level. For this preset, normalize the bass loop. Then change the L band to 125 Hz, with a ratio of 15:1, and a Low Threshold of -30 dB. Mute the LM band.
With the Multiband Dynamics processor bypassed, observe the peak value for the bass track. Now enable Multiband Dynamics, and adjust the Low band’s Gain until the peak value matches the peak value with the Multiband Dynamics bypassed-—you’ll hear a big, fat, round sound that sort of tunnels through a mix.
Now let’s go to the other extreme. A significant treble boost can help a bass hold its own against other tracks, because the ear/brain combination will fill in the lower frequencies. The next screen shot shows settings for extreme articulation so the bass really “pops,” and cuts through a track. Again, start with the default preset but set the Low band frequency to 110 Hz or so.
The only band that’s compressed is the Mid band (320 – 1.2 kHz, with parameter settings shown in the screen shot). A bit of gain for the High Mid band emphasizes pick noise and harmonics—5 dB or so seems about right—and to compensate for the extra highs, add some gain to the low band below 110 Hz. Again, about 4-5 dB seems to work well.
When adjusting the Multiband Dynamics processor, note that you can zero in on the exact effect you want for each band by using the Solo and Mute buttons on individual stages. So next time you want to both compress and equalize bass, consider using Multiband Dynamics instead—and get the best of both worlds.
The “pumping” effect is a cool EDM staple that also works with other intense forms of music. One of the best-known examples is Eric Prydz’s seminal EDM track from 2004, “Call on Me.” Usually, this technique requires sidechaining, but with the PreSonus Compressor sidechain filter, we’re covered. The effect works best if there are some sustaining sounds with which it can work—like cymbals for drum parts, or pads if you want to pump a non-drum track. Listen to the audio example to hear how the pumping effect alters a drum track.
To start, let’s try pumping some drums. Insert the Compressor in the track, and click on the Compressor’s Filter and Listen Filter buttons. To have the kick create the signal that provides the pumping, set the Lowcut frequency to off, and lower the Highcut filter until you hear pretty much nothing but kick. Once you’ve isolated the kick (or snare, or whatever you want to isolate), turn off Listen Filter but leave Filter on.
The control settings are quite crucial; the screenshot shows some potential initial settings, but you’ll need to edit the controls based on the source audio and the desired effect.
The effect’s depth, like any compression effect, depends on the Threshold and Ratio settings. For a pretty heavy-duty effect, set Threshold between -20 and -30 dB and Ratio around 10. You’ll want to tweak this depending on the program material, but it’s a good place to start.
Now for the pumping. Start with Attack at minimum, and set Release for the desired amount of pumping—you’ll probably want a time between 100 and 300 ms, depending on the song and the material. To restore some of the attack at the start of the pumping, increase the Attack time. Even a little bit, like 5 ms, restores most of the attack’s effect.
Finally, note that because this effect does in fact compress, you’ll probably want to add some makeup gain. And once you do, there you have it—the pumping sound.
I’ve always appreciated Studio One’s analytics—the spectrum analyzer, the dynamic range meter in older versions and the more modern LUFS metering in Studio One 4, the K-Scale meters based on Bob Katz’s research, the strobe tuner, and the ability to stretch the faders in the Mix view when you want to couple high resolution with long fader travel. But I wonder if the Phase Meter and its companion Correlation Meter get the props they deserve, so let’s look at what this combo can do for you.
Phase Meters—Not Just for Mixdowns!
Most people consider a tool like the Phase Meter as being only for checking final mixes. However, one very useful technique is putting it in the master output bus, and soloing one track at a time (remember, you can Alt+click on a track’s Solo button for an “exclusive solo” function). This gives some insights into the phase, level, and stereo spread of individual tracks in a way that’s more revealing than just looking over panpots.
Correlation Meter Basics
In brief, the Correlation meter (the bar graph at the Phase meter’s bottom) indicates a stereo signal’s mono compatibility. This was of crucial importance when mastering for vinyl, because it could indicate if there were out-of-phase audio components in the audio that could possibly cause the stylus to jump out of its groove. These days, it’s largely a stereo world but it’s still important to check for mono compatibility—after all, when listening to speakers, you don’t have perfect stereo separation. You’ll usually monitor correlation in the master bus, but for individual tracks, it can indicate whether (for example) a signal processor is throwing a track’s left and right channels out of phase.
The Correlation meter reading spans the range between -1 (the right and left channels are completely out of phase, with no correlation) and +1 (the right and left channels are identical, and correlate completely). With most mixes, the bar graph will fluctuate between 0 and +1.
If the Phase meter displays a single vertical line, then the left and right channels are identical, and the track is mono. The Correlation bar graph meter at the bottom confirms this with its reading of 1.00, which means the left and right channels correlate completely—in other words, they aren’t just similar, but identical.
Left and Right Readings
If there’s a single, diagonal line on the L axis, that means that all the signal’s energy is concentrated in the left channel. Similarly if there’s a single, diagonal line on the R axis, then all the signal’s energy is concentrated in the right channel. If you pan a track where the left and right channels are identical (as shown by the Correlation meter displaying 1.00), then the line will move from one channel to the other.
With stereo, you’ll see an excellent visual representation of how much the signal extends into the stereo field. The vertical size indicates the level. As you pan the signal left or right, the stereo field will become narrower around the line that moves from left to right until at one extreme or the other, you’ll see only a diagonal line on the L or R axis.
Note the correlation meter is showing +0.47. This means that there’s about an equal amount of similarity between the left and right channels as there are differences, but nothing is out of phase.
Mid-Side Encoded Audio
With Mid-Side encoded audio, you’ll see amplitude around the L and R axes, as well as along the M axis. Because the L signal is the center and the R signal the sides, you’ll see a lot more level along the L axis. Also, note the Correlation meter setting of 0.00—this means that there’s no similarity between the right and left channels, which is what you’d expect with a Mid-Side encoded signal.
Binaural Pan Signal
Studio One’s Binaural Pan processor widens the stereo image so that there’s much more energy in the right and left sides than in the center; this image shows what happens when you set the widening to maximum. Compare this to the reading for stereo signals—you can see that in this case, the energy extends further out to the right and left. Furthermore, the Correlation meter shows that there are no significant similarities between the right and left channels, which is a result of the Binaural Pan processor being based on Mid-Side processing.
Here, the Correlation meter shows a negative number, which means there are out-of-phase elements within the stereo mix. Occasional negative blips aren’t a problem, but if the Correlation meter spends a substantial amount of time to the left of 0, then there’s a phase issue that will interfere with mono compatibility.
The Shepard Tone (aka Barberpole) is an audio illusion where a tone always seems to keep rising (or falling). You may have heard it before—to build tension in music by Swedish House Mafia, Beatsystem, Data Life, and Franz Ferdinand, as the sound effect for the endless staircase in Super Mario 64, for the sound of constant acceleration for the Batpod in The Dark Knight and The Dark Knight Rises, at the end of Pink Floyd’s “Echoes” from the Meddle album, or in the soundtrack for the film Dunkirk in sections where the goal was to produce a vibe of increasing intensity. Check out the audio example, and you’ll hear how the tone just goes on forever.
Thanks to Studio One’s Tone Generator, it’s easy to produce a Shepard Tone loop—just follow the step-by-step instructions, in a song with the tempo set to 120 BPM.