This tip is excerpted from the updated/revised 2nd Edition of How to Record and Mix Great Vocals in Studio One. The new edition includes the latest Studio One 5 features, as well as some free files and Open Air impulses, but also has 35% more content than the first edition—it’s grown from 121 to 194 pages. And as a “thank you” to those who bought the original version, you’re eligible for a 50% discount on the 2nd edition. There’s also a bundle with the book and my complete set of 128 custom impulses for Open Air…but so much for how I spent my summer vacation, LOL. Let’s get to the tip.
Suppose you’ve laid down your raw vocal—great! Now it’s time to overdub some instrumental parts and background vocals. Unfortunately, though, that raw vocal is kind of…uninspiring. So you start browsing effects, tweaking them, trying different settings—and before you know it, you’re going down a processing rabbit hole in the middle of your session.
Next time, open up the Vocal QuickStrip. Insert this vocal processing’s “greatest hits” FX Chain in your vocal track, tweak a few settings, admire how wonderful the vocal sounds, and then carry on with your project.
There’s a download link for the Vocal QuickStrip.multipreset file, so you don’t need to assemble the chain yourself. It works with Studio One 4 as well as 5 (note that the Widen button for the Doubler is functional only in Studio One 5).
The Fat Channel (Fig. 1) is the heart of the chain. Of its three available compressors, the Tube Comp model emulates the iconic LA-2A compressor—the go-to vocal compressor for many engineers.
Figure 1: Fat Channel settings for the Vocal QuickStrip FX Chain.
The Fat Channel also includes a built-in high-pass filter. You can place the EQ either before or after the compressor; here, the EQ is before the compressor because boosting certain frequencies “pushes” the compressor harder. This contributes to the Vocal QuickStrip’s character.
The EQ uses all four stages. The most interesting aspect is how the Low Frequency and Low-Mid Frequency stages interact subtly when you edit the Bottom control. The Low-Frequency stage is fixed at 110 Hz with 1 dB of gain, but its Q tracks the Low-Mid Frequency stage’s Gain control. So, when you pull the LMF Gain down, the LF stage’s Q gets broader; increase Gain, and the Q goes up somewhat.
The High-Mid Frequency stage sits at 3 kHz, because boosting in this frequency range can improve intelligibility. The High-Frequency section adds “air” around 10 kHz. However, as you increase the Top control, the frequency goes just a bit lower so that the boost covers a wider section of the high-frequency range. This makes the effect more pronounced.
The Chorus is the next processor in the chain, but it’s used for doubling, not chorusing (Fig. 2).
Figure 2: The Chorus provides a voice-doubling ADT effect.
The parameters are preset to a useful doubling effect, and there are only two control switches—one to enable/bypass the effect, the other to increase the stereo spread.
For echo/delay effects, the Analog Delay comes next (Fig. 3). Although many of the parameters are well-suited to being macro controls, there had to be a few tradeoffs to leave enough space for the crucial controls from other effects.
Figure 3: The Analog Delay is set up for basic echo functionality.
For example, the Delay Time controls beats rather than being able to choose between beats and sweeping through a continuous range. Feel free to change the Macro control assignment. Also, the LFO isn’t used, so if you want to modify the ping-pong effects, you’ll need to open the interface and do so manually. In any event, the Delay Beats, Feedback, and Mix parameters cover what you need for most vocal echo effects.
The final link in the chain is the Open AIR reverb (Fig. 4). Normally I use my own impulse responses (see the Friday Tip Synthesize Open AIR Reverb Impulses in Studio One for info on how to create your own impulse responses), but of the factory impulses, for vocals I’m a big fan of the Gold Plate impulse. (If you have my Surreal Reverb Impulse Responses pack that’s available from the PreSonus shop, I’d recommend using the 1.2 Fast Damped, 1.5 Fast Damped, or 2.25 Fast Damped vocal reverbs. However, note that these three files are also included for free with the second edition of the Vocals book)
Figure 4: The Open AIR reverb plug-in’s Gold Plate impulse response is one of my favorite factory impulses for vocals.
When designing an FX Chain with so many available parameters, you need to choose which parameters (or combinations of parameters) are most important for Macro controls (Fig. 5).
Figure 5: The Vocal QuickStrip Macro controls.
Compress varies both the Peak Reduction and Gain to maintain a fairly constant output—an old trick (see the EZ Squeez One-Knob Compressor tip), but a useful one. Bottom, Push, and Top control three EQ stages. All of these, and the Compressor, have bypass switches so it’s easy to compare the dry and processed settings.
Delay also has a bypass switch, as well as controls for delay time in beats, delay feedback, and dry/wet delay mix. The only switches for the chorus-based doubling function are bypass and narrower/wider. Reverb includes a bypass button and dry/mix control, because that’s really all you need when you have a gorgeous convolution reverb in the chain.
So go ahead and download the Vocal QuickStrip, use it, and have fun. But remember that an FX Chain like this lends itself to modifications—for example, insert a Binaural Pan after the Open AIR reverb, or optimize some EQ frequencies to work better with your mic or voice. Try the other two compressors in the Fat Channel (or if you’re a PreSonus Sphere member, then try the other eight compressors—they all have different characters). With a little experimentation, you can transform an FX Chain that works for me (and will hopefully work well for you) to an FX Chain that’s perfect for you. Go for it!
Full disclosure: I’m not a big fan of chorusing. In general, I think it’s best relegated to wherever snares with gated reverbs, orchestral hits, DX7 bass presets, Fairlight pan pipes, and other 80s artifacts go to reminisce about the good old days.
But sometimes it’s great to be wrong, and multiband chorusing has changed my mind. This FX Chain (which works in Studio One Version 4 as well as Version 5) takes advantage of the Splitter, three Chorus plug-ins, Binaural panning, and a bit of limiting to produce a chorus effect that covers the range from subtle and shimmering, to rich and creamy.
There’s a downloadable .multipreset file, so feel free to download it, click on this window’s close button, bring the FX Chain into Studio One, and start playing. (Just remember to set the channel mode for guitar tracks to stereo, even with a mono guitar track.) However, it’s best to read the following on what the controls do, so you can take full advantage of the Multiband Chorus’s talents.
The Splitter creates three splits based on frequency, which in this case, are optimized for guitar with humbucking pickups. These frequencies work fine with other instruments, but tweak as needed. The first band covers up to 700 Hz, the second from 700 Hz to 1.36 kHz, and the third band, from 1.36 kHz on up (Fig. 1).
Figure 1. FX Chain block diagram and Macro Controls panel for the Multiband Chorus.
Each split goes to a Chorus. The mixed output from the three splits goes to a Binaural Pan to enhance the stereo imaging, and a Limiter to make the signal “pop” a little more.
Regarding the control panel, the Delay, Depth, LFO Width, and 1/2 Voices controls affect all three Choruses. Each Chorus also has its own on/off switch (C1, C2, and C3), Chorus/Double button (turning on the button enables the Double mode), and LFO Speed control. You’ll also find on/off buttons for the Binaural Pan and Limiter, as well as a Width control for the Binaural Pan. Fig. 2 shows the initial Chorus settings when you call up the FX Chain.
Figure 2. Initial FX Chain Chorus settings.
Because chorusing occurs in different frequency bands, the sound is more even and has a lusher sound than conventional chorusing. Furthermore, setting asynchronous LFO Speeds for the three bands can give a more randomized effect (at least until there’s an option for smoothed, randomized waveform shapes in Studio One).
A major multiband advantage comes into play when you set one of the bands to Doubler mode instead of Chorus. You may need to readjust the Delay and Width controls, but using Doubler mode in the mid- or high-frequency band, and chorusing for the other bands, gives a unique sound you won’t find anywhere else. Give it a try, and you’ll hear why it’s worth resurrecting the chorus effect—but with a multiband twist.
At first, the changes to the effects in Version 5 seem mostly cosmetic. But dig deeper, and you’ll find there’s more to the story—so let’s find out what’s new with Limiter2 (Fig. 1).
Figure 1: Limiter2 has had several design changes for Version 5.
The control parameters are more logical, and easier to adjust. Prior to V5, there was an unusual, by-design interaction with the Ceiling and Threshold controls; setting Ceiling below Threshold gave the limiter a softer knee. However, the tradeoff was difficulty in obtaining predictable results. Besides, if the soft knee aspect is important to you for dynamics control, just use the Compressor with a really high ratio.
In Limiter2, the Threshold is relative to the Ceiling—the Ceiling sets Limiter2’s absolute maximum level, while Threshold sets where limiting begins below the Ceiling, based on the Threshold parameter value. For example, if Ceiling is 0.00 and Threshold is -6.00, then the limiter’s threshold is ‑6.00 dB. But if the Ceiling is ‑3.00 dB and the Threshold is -6.00, then the limiter’s Threshold is -9.00 dB. Makeup gain occurs automatically so that as you lower the Threshold parameter, the output level increases as needed to meet the Ceiling’s target output level.
Modes and Attacks
There are now two Modes, A and B, and three Attack time settings. The pre-V5 Limiter had less flexible attack options, which mostly impacted how it responded to low-frequency audio; the waveform could have some visible distortion when first clamped, but the distortion would disappear after the attack time completed.
I’ll spare you the hours I spent listening and nerding around with the (highly underrated) Tone Generator and Scope plug-ins to analyze how the new options affect the sound, so here’s the bottom line.
In applications where you want to apply something like 6 dB of peak reduction to make a track or mix “pop,” the Limiter2 performance in Mode A is essentially perfect. Unless you’re into extreme amounts of limiting or material with lots of low frequencies, Mode A should cover what you need 95% of the time (and it also outperforms the pre-V5 limiter).
If you’re using Limiter2 as a brickwall limiter to keep transients from spilling over into subsequent stages, then use Mode A/Fast attack for the highest fidelity and give up a tiny bit of headroom, or Mode B/Fast Attack for absolute clamping.
Fig. 2 shows how the fast and slow times compare. The following were all set for 50 ms release times, 1 kHz sine wave input, and -20 dB Threshold—so Limiter2 was being hit pretty hard.
Figure 2: Fast and Slow attacks compared for Modes A and B, cropped to 150 ms duration. Top to bottom: Mode A/Fast, Mode A/Slow, Mode B/Fast, Mode B/Slow.
The visuals are helpful, but on signals with fast transients, you may hear more of a difference with the different attack times than these images might indicate. Nonetheless, it’s clear that Mode B/Fast is super-fast. If you look carefully at Mode A/Slow, you’ll see a very tiny downward blip on the first cycle of the attack (it’s less visible on Mode B/Slow). Mode A takes about 20 ms to settle down to its final level.
For more background on the nuts and bolts of how this works, the tradeoff for Mode B’s higher speed mostly involves very low frequencies (under 80 Hz or so, and especially under 50 Hz). With a 50 Hz sawtooth wave, 100 ms Release, and a significant amount of limiting, Mode B/Slow gives some visible overshoot and distortion. Mode B/Fast reduces the overshoot but increases distortion. Mode A does less of both—with Slow, there’s less overshoot, and with Fast, there’s less distortion. Note that any distortion or overshoot occurs only when pushing Limiter2 to extremes: very low-frequency waveforms, with steep rise/fall times, short release times, and lots of limiting. However, this is mostly of academic interest with waveforms that have lots of harmonics, like sawtooth and square. The level of the harmonics is high enough to mask any low-level harmonics generated by distortion.
I also tested with a sine wave, which gives an indication of what to expect with audio like a kick drum (e.g., 40-60 Hz fundamental) or low bass notes. Mode B/Fast has less distortion than Mode B/Slow, while Mode A, fast or slow, flattens peaks almost indiscernibly (Fig. 3).
Figure 3: A 30 Hz sine wave with about 15 dB of limiting. Top: A Mode. Middle: B Mode/Fast. Bottom: B Mode/Slow.
In this situation, Mode A would likely be my choice, but as always, use your ears—the light distortion from Mode B can actually enhance kick drum and bass tracks. Also note that which mode to use depends on the release time. For example, with a short (35 ms) release, B/Slow had the most audible distortion, B/Fast was next, and B/Normal had no audible distortion.
While I was in testing mode, I decided to check out some third-party limiters (Fig. 4) with a different program. These are all marketed as “vintage” emulations, and set to the fastest possible attack time.
Figure 4: The results of testing some other limiters.
In case you wondered why some people say these vintage limiters have “punch”…now you know why! The time required to settle down to the final level is pretty short (except for the bottom one), but the limiter doesn’t catch the initial peaks. This is why you can insert one of these kinds of limiters, think you’re limiting the signal, but the downstream overload indicators still light on transients. Incidentally, the Fat Channel’s Tube limiter has this kind of “vintage punch” response in the Limit mode, while the Fat Channel’s one-knob, final limiter stage—although basic—is highly effective at trapping transients.
Studio One’s Overlap Correction feature for Note data isn’t new, but it can save you hours of boring work. The basic principle is that if Note data overlaps so that the end of one note extends long enough to overlap the beginning of the next note, selecting them both, and then applying overlap correction, moves the overlapping note’s end earlier so that it no longer overlaps with the next note.
My main use is with keyboard bass. Although I play electric bass, I often prefer keyboard bass because of the sonic consistency, and being able to choose from various sampled basses as well as synth bass sounds. However, it’s important to avoid overlapping notes with keyboard bass for two main reasons:
One option for fixing this is to zoom in on a bass part’s note data, check every note to make sure there aren’t overlaps, and shorten notes as needed. However, Overlap Correction is much easier. Simply:
Normally I’m reluctant to Select All and do an editing function, but any notes that don’t overlap are left alone, and I haven’t yet run into any problems with single-note lines. Fig. 1 shows a before-and-after of the note data.
Figure 1: The notes circled in white have overlaps; the lower copy of the notes fixes the overlaps with the Length menu’s Overlap Correction feature.
Problem solved! The reason for setting overlap to -00.00.01 instead of 00.00.00 is because with older hardware synthesizers or congested data streams, that very slight pause ensures a note-off before the next note-on appears. This prevents the previous note from “hanging” (i.e., never turning off). You can specify a larger number for a longer pause—or live dangerously, and specify no pause by entering 0.
Also, although I referenced using this with keyboard bass, it’s useful for any single-note lines such as brass, woodwind, single-note MIDI guitar solos, etc. It can also help with hardware instruments, including electronic drums, that have a limited number of voices. By removing overlaps, it’s less likely that the instrument will run out polyphony.
There’s some intelligence built into the overlap correction function. If a note extends past another note, there won’t be any correction. It also seems to be able to recognize pedal points (Fig. 2).
Figure 2: Overlap Correction is careful about applying correction with polyphonic lines.
Selecting all notes in the top group of notes and selecting Overlap Correction didn’t make any changes. As shown in the bottom group of notes, preventing the pedal point from overlapping the final chord requires selecting the pedal point, and any of the notes in the last chord with which the pedal point overlaps.
It’s easy to overlook this gem of a feature, but it can really help with instrumental parts—particularly with keyboard bass and solo brass parts.
As you can probably tell, I’m a fan of FX Chains—they satisfy my inner DIY impulse to put things together, and result in some cool, useful, new processor I didn’t have before. For this Friday’s tip, let’s put together a Transient Shaper designed specifically for drums and percussion. It can emphasize the attack, the post-attack section (called “Girth” in the FX Chain), or both, as well as mix any blend of them. Of course, there’s a download link for the multipreset—but first, let’s listen to what transient shaping can do.
The first two measures are the straight Crowish Acoustic Bridge 2 w. Fill drum loop from Studio One’s sound library. The next two measures add Attack shaping, the next two add Girth only, and the final two measures combine Attack and Girth, with 1 dB of limiting. All examples are normalized to the same peak level.
Fig. 1 shows the block diagram. Mixtool 3 adjusts the input level, because when feeding any dynamics processors, you need to find the sweet spot where the processors act as expected. In this case, you want the input level to provide a signal that uses up most of the headroom.
The incoming audio splits into three paths. The left path is an expander, set up to provide upward expansion. This is what emphasizes the attack. The Mixtool adjusts the path’s level.
The middle path is a compressor, set for the shortest attack time possible to reduce any existing attack to a minimum. Some compression brings up the post-attack part of the audio. Mixtool 4 adjusts this path’s level.
The right-most path sets the dry signal’s level. This is an important parameter, because you can take out the dry signal and be left with only what’s contributed by the Attack and Girth paths, or use them to enhance the dry sound.
It takes a little effort to get familiar with the controls. The Attack shaper is the main point of this FX Chain, so to acquaint yourself with what the Attack parameters do, load up a drum loop of your choice, and then do the following.
One final comment: It’s easy to go overboard with transient shaping, but after the novelty wears off, you’ll find that even a little bit of enhanced attack can make a track sound more lively. And while we’ve covered this only with drums, it also works for bass attacks, plucked strings, and strange percussion instruments…basically if something has an attack, this FX Chain can shape it.
I’ve always loved having one track impart its characteristics to a different track (“cross-modulation”), particularly for EDM. A good example is using a vocoder for “drumcoding,” where drums—not a microphone—provide the vocoder’s modulation source. Previous Friday Tips along these lines include The Ultra-Tight Rhythm Section, Smoother/Gentler Sidechain Gating, Pumping Drums – With No Sidechain, and most recently, Rockin’ Rhythms with Multiband Gating.
Sending audio from one track over a sidechain to control dynamic EQ in another track is great for cross-modulation effects—and now this is easy to do in Studio One 5 because sidechaining has been added to the Multiband Dynamics processor. One of my favorite effects is using the kick drum to boost the upper midrange on a rhythm guitar part or keyboard pad so that the guitar or pad emphasizes the rhythm…and that’s just one possibility.
This isn’t about “textbook” dynamic EQ in the sense of being able to use any type of filter (e.g., highly resonant bandpass) as the EQ, but as pointed out in the Friday Tip Studio One’s Secret Equalizer, the Multiband Dynamics combines both EQ and dynamics. We’ll use that to our advantage—and in a way, a relatively broad filter response is better for this kind of application. (The typical dynamic EQ application involves fixing a problem, and for that, you often need precise filtering.)
Insert the Multiband Dynamics in the Target track, like guitar, pad, organ, etc. Then, insert a Send (pre-fader is probably best) in the Source track (e.g., kick or snare drum). Assign the Send to the Multiband Dynamics sidechain (Fig. 1).
Figure 1: This technique requires a source track to trigger the Multiband Dynamics’ sidechain and a target track that’s processed by the Multiband Dynamics.
This is where the fun begins. The sidechain feeds all Multiband Dynamics bands simultaneously, so the most basic implementation would be bypassing all the bands except for one, which you then set to either cut or boost a particular frequency range. The amount of boost or cut depends on the level that the source track sends to the sidechain.
For example, with compression, you can create pumping effects (Fig. 2).
Figure 2: The Multiband Dynamics attenuates the selected frequency range whenever it receives a signal from the source track.
In this example, a kick drum is modulating a pad. Every kick drum hit attenuates the HM (High-Mid) range; the amount of attenuation fades over the 1000 ms Release time. A shorter Release parameter creates a more percussive effect. Choose the frequency range you want to modulate by adjusting the crossover frequencies. Even better, note that you can automate the Multiband Dynamics’ crossover frequencies, so the frequency range can sweep over time—this is a novel effect that adds considerable animation.
Another option is to raise the target band’s Gain so that any modulation lowers the band’s level. In other words, the default state for that band is boosted, and modulation reduces the boost.
You can also boost a band’s level in response to dynamics, by setting the Multiband Dynamics parameters for upward expansion (Fig. 3). Note how the graphic in the upper left shows an expansion curve instead of one for compression.
Figure 3: Upward expansion boosts the target audio in the selected frequency range.
The control settings here are fairly crucial. Ratio must be set for upward expansion, so the second number in the ratio control needs to be greater than one—the bigger the number, the steeper the expansion. For the maximum expansion effect, set High Threshold to 12.00. The Low Threshold parameter determines where expansion begins, and Gain increases the overall level to compensate for the lower level below the point where expansion kicks in. Adjust Attack and Release to shape the boost’s dynamics. Because upward expansion boosts the output signal level, you may need to reduce the Global gain somewhat.
The best way to understand all the possibilities is to create a basic setup like the one in Fig. 1 with a kick drum as the source and a very simple, sustained pad (e.g., a chord with sawtooth waves) as the target. This will make it easy to hear the results of playing around with the Multiband Dynamics’ controls. And of course, it is a multiband processor, and the sidechain feeds all the bands, so you could have one band attenuating while another is boosting. If you get into automating parameters, the sky’s the limit.
Dynamic EQ can also be useful to process signal processing. For example, suppose there’s a main reverb inserted in a bus, to which you send drums, guitar, voice, etc. To avoid muddiness, insert a Multiband Dynamics after the reverb, use kick as the sidechain source, and attenuate the low frequencies whenever the kick hits.
Cross-modulation with dynamic EQ can be serious fun…give it a try.
I wanted a Bucket Brigade Delay (BBD) effect in Studio One. Seriously.
Although some analog delays (e.g., Binson Echorec) were based on tape, others used analog “bucket brigade” technology. Bucket brigade integrated circuits (like the Panasonic/Matsushita MN3005 or Reticon SAD-1024) incorporated thousands of capacitive elements controlled by a clock. Each clock cycle passed the analog signal at the input from one stage to the next, so slower clocks meant longer delays. But because sampling (albeit analog) was involved, so was the Nyquist theorem—the more you slowed down the clock, the more likely you’d hear aliasing and distortion. At really long delays, sometimes you’d even hear leakage from a clock that had gotten down to the audible range.
So I emailed Arnd Kaiser, the General Manager for PreSonus Software, and told him I wanted to modify the Analog Delay into a BBD. He seemed puzzled and said that if you turn up the Drive and lower the High Cut frequency at longer delays, you’ll get the BBD sound. True, but that sound is of a clean, well-designed BBD where the designer didn’t push the chips, and knew how to layout a circuit board. That’s fine, but I wanted filth…time for an FX Chain (Fig. 1).
Figure 1: The Analog Delay plug-in is the heart of this FX Chain.
The solution was tracking the Delay Time with the BitCrusher’s Downsampling parameter, so at longer delay times, those lovely violations of Mr. Nyquist’s theorem could grace the sound with aliasing and sonic nastiness.
I was running late submitting the tip because of going down this crazy BBD rabbit hole, so I emailed Ryan Roullard at PreSonus (who among a zillion other things makes sure my Friday Tips roll along smoothly every week) to apologize for the delay and give him a heads-up of what to expect for this week’s tip. He asked if I’d included the clock leakage whine as part of the sound. I was embarrassed to say that I had overlooked it, but Ryan said that if I figured out how to add it in, he’d never tell anyone of my shameful omission.
You can download the Bucket Bridge Delay.multipreset, so I won’t go into much detail—reverse-engineer it to find out how it works, or modify it to do even stranger things. Please note: It’s probably best to insert this into an FX Bus with the Mix control set to wet only because unless none of the three switches is enabled, there’s no way to have a completely clean sound.
The important part is the three switches—Arnd, Craig, and Ryan. You can select none of them or any/all of them. With none of the three switches selected, you have a standard Analog Delay sound, with the other knobs and buttons doing their standard Analog Delay functions. But…
And there you have it—delicious, modern digital meets filthy, vintage analog. Have fun!
Any time you want to do detailed edits in context with a mix, Dim Solo is your friend. When you solo a track with a Dim Solo function enabled, the non-soloed tracks aren’t muted but instead play back at a lower (dimmed) level. I find this essential for many workflows, particularly comping. When you use the Listen Tool to audition various comps, normally you don’t hear them along with the rest of the mix. So one of the comps might sound wonderful, but when you play it back in context, find the timing was off. Dim Solo provides an immediate reality check.
I wanted this function so much in Studio One that my second “Friday Tip” blog post was about how to kludge a Dim Solo function by adding a Sub bus. But kludges are no longer needed, because V5’s new Listen Bus provides an efficient, flexible Dim Solo solution.
This technique works best with interfaces that have a mixer applet (like Universal Control) with virtual outs. I’ve tested this with the 1824c and Studio 192; just make sure you don’t bypass the Universal Control mixer. Start by right-clicking in a channel, enabling the Listen Bus, and checking Solo through Listen Bus (Fig. 1).
Now go to the Audio I/O setup and choose the Listen Bus output (Fig. 2). Although the line outs can feed physical outs, with the Universal Control mixer they can also feed virtual outs. The Main bus can feed the usual 1+2 outs, while the Listen Bus feeds the 3+4 outs.
Figure 2: Audio I/O Setup for the 1824c or Studio 192.
As another example, Universal Audio’s Apollo Twin USB also creates virtual outs. Fig. 3 shows the Audio I/O setup.
Figure 3: In Universal Audio’s Apollo Twin USB, the Listen Bus goes to a set of virtual outputs.
With the Listen Bus assigned to a virtual output, you can hear both the Main and Listen buses within your usual monitoring system. If virtual outputs aren’t available, then the Listen Bus needs to go to a hardware output, which requires a way to monitor the Listen Bus audio. For example, the Listen Bus could go to a Monitor Station input.
Now it’s time for the Dim part. Insert a Mixtool in your Main Bus, and lower the Gain to whatever creates an ideal balance for listening to the soloed track compared to the rest of the mix (Fig. 4).
Figure 4: The Mixtool controls the Main bus level.
I usually choose about -12 dB of attenuation. To Dim the mix, enable the Mixtool. Bypass it to return the mix to its normal level. (Sometimes I even insert two Mixtools, one set to -6 dB and the other to -12 dB.)
So now we have the option of a continuously variable amount of dimming, down to -24 dB. But, Studio One V5 has a couple other tricks up its sleeve.
The Listen bus has a pre-/post-fader option. The soloed track will still appear in the dimmed mix if its fader is up, but this probably won’t matter because the Listen Bus level will be louder. However if you do need to excise the soloed sound from the dimmed mix, pull down the fader on the channel you’re soloing, and set the Listen Bus pre/post fader switch to pre-fader.
Another small but useful feature is that if there’s a fadeout on the Main bus, the Listen bus isn’t affected by the fade, so it’s easy to hear your edits even as a song fades out. Also, if you want to hear the track feeding the Listen Bus in isolation, no problem—just mute the Main bus.
Dim Solo improves workflow considerably when comping and editing, and thanks to the Listen Bus, it’s now easy to do.
Yes, Studio One 5’s Pro EQ2 has a more “pro” look…but there are also some major improvements under the hood, so let’s investigate.
This is arguably the most significant change, and appears as an eighth filter stage just below the left of the frequency response display (Fig. 1).
Figure 1: The phase-linear Low-Cut filter section offers three cutoff frequencies and two different slopes.
There’s much mythology around linear-phase EQ, so here are the basics. Traditional EQ introduces phase shifts when you boost or cut. With multiple EQ stages, these phase differences can produce subtle cancellations or reinforcements at particular frequencies. This may or may not create a sometimes subtle, sometimes obvious effect called “smearing.” However, it’s important to note that these phase shifts also give particular EQs their “character” and therefore, can be desirable.
Linear-phase EQ technology delays the signal where appropriate so that all bands are in phase with each other. This tends to give a more “transparent” sound. You might wonder why there’s only one linear-phase stage, with a low-cut response, but there’s a good reason for this. Many engineers like to remove unneeded low frequencies for utilitarian purposes (e.g., remove p-pops or handling noise from vocals), or for artistic reasons, like reducing lows on an amp sim cab to emulate more of an open-back cab sound. Standard EQ introduces phase changes above the cutoff frequency; with linear-phase EQ, there are no phase issues. This can be particularly important with doubled audio sources, where you don’t want phase differences between them due to slightly different EQ settings.
The Pro EQ2 is very efficient, but note that enabling linear-phase EQ requires far more CPU power, and adds considerable latency—it’s not something you’ll want to add to every track. Fortunately, in many cases, it’s a setting that you apply and don’t think about anymore. This makes it a good candidate for “Transform to Rendered Audio” so you can reclaim that CPU power, and then use conventional EQ going forward.
By the way, an argument against linear-phase EQ is that it can create pre-ringing, which adds a low-level, “swooshing” artifact before audio transients. Fortunately, it’s a non-issue here, because pre-ringing is audible only at low frequencies, with high gain and Q settings. (Note that traditional EQ can add post-ringing, although you usually won’t hear it because the audio masks it.)
I’ve wanted this feature for a long time. Some EQ changes are extremely subtle, particularly when mastering. With range set to 24 dB, it’s difficult to drag nodes around precisely. What’s more, when making fine gain changes, with the 24 dB view it’s easy to move slightly to the right or left, and end up editing frequency instead. Holding Shift provides fine-tuning, but for fast EQ adjustments, the 6 dB view is welcome (Fig. 2).
Figure 2: It’s much easier to see subtle EQ changes by setting the level range to 6 dB.
Granted, you adjust EQ with your ears, not your eyes—but learning how to correlate sound to frequency is important. I knew one guitar player who when he said something like “that track really needs to come down about 2.5 dB at 1.25 kHz,” he was 100% spot-on. When mixing, he could zero in on EQ settings really fast.
And there’s another implication. Those learning how to use EQ often overcompensate, so at seminars, I advise applying what I call “the rule of half”: if you think a sound needs 6 dB of boost, try 3 dB of boost instead and get acclimated to it before adding more boost. If you choose the 6 dB view, you’ll be forced to use smaller boost and cuts in order to adjust or see them graphically—and you might find those smaller changes are all you need.
12th Octave Frequency Response Display
The Third-Octave Display is good eye candy, and gives a rough idea of how EQ affects the sound. The new 12th-Octave resolution option gives far better definition. In Fig. 3, note how many of the peaks and dips visible in the 12th-Octave display are averaged out, and lost, in the Third-Octave version.
In addition to the more “marquee” improvements, several other additions make working with Pro EQ2 a better experience than the original Pro EQ.
Keyboard Display. Now you can correlate frequency to note pitches; note that these line up with the bars in the 12th-octave display.
Band Controls. In Studio One 4, there was a little, almost invisible arrow between the controls and the frequency response display. Clicking on this hid the controls. The Band Controls button does the same thing, and you won’t overlook it.
Curves Button. Similarly, Studio One 4’s All/Current buttons that control how curves are displayed have been consolidated into a single Curves button.
Sidechaining. We already covered Pro EQ sidechaining in the blog post The Sidechained Spectrum. However, when choosing the FFT curve, now there’s a sidechain spectrum peak hold button for the sidechain input. Clicking on the “snowflake” button freezes peaks (hence the name) until you click the button again.
Better Metering. Studio One 4’s Pro EQ had only output metering, whereas Pro EQ2 has metering for both input and output. This is a highly useful addition. If the output is too hot, you can always turn down the output level, but you won’t know if the reason why it’s hot is because you’ve boosted some frequencies too much, or the input level is hitting the EQ too hard. Now you’ll know. As with Studio One 4, the metering shows both peak and average levels.
And that’s a wrap for Pro EQ2. I guess you could say the newer version is ahead of the curve…the EQ curve, that is 😊
Tape Resampler, a new Studio One 5 feature, replicates an “old school” time-stretch technique that varied pitch and tempo simultaneously and proportionately. Today’s DSP can change pitch and tempo independently, which is cool. But the price you pay is artifacts, because when changing tempo or pitch, you need to either delete or add data.
With resampling, the data stays the same—so there are no artifacts, and the sound is natural. Although extreme speedups give the “Chipmunks” sound and extreme slowdowns evoke Darth Vader on tranquilizers, subtler speed changes were used all the time with tape. It was common to speed up a master tape by a few per cent to give the tempo a slightly faster, “peppier” sound, as well as some added brightness. (If you’ve ever tried to play along with a song that was several cents sharp, it was probably sped up a bit.)
The manual mentions using Tape Resampler to fit loops to tempo (assuming accurate pitch isn’t crucial), but there’s another application that at least to me, is worth the update price by itself. With tape, it was common to slow the tape down or speed it up, play along with the part, and then return the speed to normal. This produced a timbral and formant shift, and was popular for background vocals. For example, if a song was in the key of A, you’d slow down to the key of G, sing along with it in G, then return the tape to normal. The vocal would have a brighter formant change that often worked well. This could also help you hit notes that were just out of your range. (We covered similar techniques in the blog post Varispeed-Type Formant Changes, but because they used DSP, at least some artifacts were unavoidable.)
The Handy Transposition Chart
|Semitones||Pitch Up||Pitch Down|
Figure 1: The overdub is being raised two semitones.
Note that the transpose numbers relate to the 12th root of 2. This irrational number (its numerical value has been taken out to over twenty billion decimal digits, but it still doesn’t repeat!) sets the ratio between semitones of the even-numbered scale. Fortunately, three significant digits covers our needs.