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.
The response to Studio One 5 and the launch of PreSonus Sphere has been OVERWHELMING, to say the very least… We are blown away and extremely grateful for our customers’ enthusiasm! We are working non-stop to respond to specific issues any customers may run into. We wanted to take a minute and answer a few questions about the grace period we’re offering to our customers.
PreSonus is providing a super-sized 90 day grace period for Studio One version 5. This means if you registered a copy of Studio One 4 on or after April 1, 2020, you are within the grace period. Here are all the specifics:
Note: If you registered a copy of Studio One 4 Artist that was bundled with a third-party product during the grace period, we are sorry but you are not eligible for an upgrade.
During this grace period, if you have multiple copies of Studio One Professional or Studio One Artist in your account, all copies are stamped as upgraded automatically. However, only you only get one copy of Studio One 5 or PreSonus Sphere per account, not per product. This is only a rule for those registrations that occurred during the grace period.
After clicking the Redeem button above you will see a screen similar to the one pictured below. Depending on how you acquired your product or your product type you may not have both buttons:
After last week’s thrilling cliff-hanger about how to preserve your WAV files for future generations, let’s look at how to export all your stereo audio tracks and have them incorporate effects processing, automation, level, and panning. There are several ways to do this; although you can drag files into a Browser folder, and choose Wave File with rendered Insert FX, Studio One’s feature to save stems is much easier and also includes any effects added by effects in Bus and FX Channels. (We’ll also look at how to archive Instrument tracks.)
Saving as stems, where you choose individual Tracks or Channels, makes archiving processed files a breeze. For archiving, I choose Tracks because they’re what I’ll want to bring in for a remix. For example, if you’re using an instrument where multiple outputs feed into a stereo mix, Channels will save the mix, but Tracks will render the individual Instrument sounds into their own tracks.
When you export everything as stems, and bring them back into an empty Song, playback will sound exactly like the Song whose stems you exported. However, note that saving as stems does not necessarily preserve the Song’s organization; for example, tracks inside a folder track are rendered as individual tracks, not as part of a folder. I find this preferable anyway. Also, if you just drag the tracks back into an empty song, they’ll be alphabetized by track name. If this is an issue, number each track in the desired order before exporting.
Select Song > Export Stems. Choose whether you want to export what’s represented by Tracks in the Arrange view, or by Channels in the Console. Again, for archiving, I recommend Tracks (Fig. 1).
Figure 1: The Song > Export Stems option is your friend.
If there’s anything you don’t want to save, uncheck the box next to the track name. Muted tracks are unchecked by default, but if you check them, the tracks are exported properly, and open unmuted.
Note that if an audio track is being sent to effects in a Bus or FX Channel, the exported track will include any added effects. Basically, you’ll save whatever you would hear with Solo enabled. In the Arrange view, each track is soloed as it’s rendered, so you can monitor the archiving progress as it occurs.
In Part 1 on saving raw WAV files, we noted that different approaches required different amounts of storage space. Saving stems requires the most amount of storage space because it saves all tracks from start to end (or whatever area in the timeline you select), even if a track-only has a few seconds of audio in it. However, this also means that the tracks are suitable for importing into programs that don’t recognize Broadcast WAV Files. Start all tracks from the beginning of a song, or at least from the same start point, and they’ll all sync up properly.
Note that the tracks will be affected by your Main fader inserts and processing, including any volume automation that creates a fadeout. I don’t use processors in the Main channel inserts, because I reserve any stereo 2-track processing for the Project page (hey, it’s Studio One—we have the technology!). I’d recommend bypassing any Main channel effects, because if you’re going to use archived files for a remix, you probably don’t want to be locked in to any processing applied to the stereo mix. I also prefer to disable automation Read for volume levels, because the fade may need to last longer with a remix. Keep your options open.
However, the Main fader is useful if you try to save the stems and get an indication that clipping has occurred. Reduce the Main fader by slightly more than the amount of clipping (e.g., if the warning says a file was 1 dB over, lower the Main channel fader by -1.1 dB). Another option would be to isolate the track(s) causing the clipping and reduce their levels; but reducing the Main channel fader maintains the proportional level of the mixed tracks.
Saving an Instrument track as a stem automatically renders it into audio. While that’s very convenient, you have other options.
When you drag an Instrument track’s Event to the Browser, you can save it as a Standard MIDI File (.mid) or as a Musicloop feature (press Shift to select between the two). Think of a Musicloop, a unique Studio One feature, as an Instrument track “channel strip”—when you bring it back into a project, it creates a Channel in the mixer, includes any Insert effects, zeroes the Channel fader, and incorporates the soft synth so you can edit it. Of course, if you’re collaborating with someone who doesn’t have the same soft synth or insert effects, they won’t be available (that’s another reason to stay in the Studio One ecosystem when collaborating if at all possible). But, you’ll still have the note events in a track.
There are three cautions when exporting Instrument track Parts as Musicloops or MIDI files.
The bottom line: Before exporting an Instrument track as a Musicloop or MIDI file, I recommend deleting any muted Parts, selecting all Instrument Parts by typing G to create a single Part, then extending the Part’s start to the Song’s beginning (Fig. 2).
Figure 2: The bottom track has prepped the top track to make it stem-export-friendly.
You can make sure that Instrument tracks import into the Song in the desired placement, by using Transform to Audio Track. As mentioned above, it’s best to delete unmuted sections, and type G to make multiple Parts into a single Part. However, you don’t need to extend the track’s beginning.
However, unlike a Musicloop, this is only an audio file. When you bring it into a song, the resulting Channel does not include the soft synth, insert effects, etc.
Finally…it’s a good idea to save any presets used in your various virtual instruments into the same folder as your archived tracks. You never know…right?
And now you know how to archive your Songs. Next week, we’ll get back to Fun Stuff.
You’re forgiven if you scoot down to something more interesting in this blog, but here’s the deal. I always archive finished projects, because remixing older projects can sometimes give them a second life—for example, I’ve stripped vocals from some songs, and remixed the instrument tracks for video backgrounds. Some have been remixed for other purposes. Some really ancient songs have been remixed because I know more than I did when I mixed them originally.
You can archive to hard drives, SSDs, the cloud…your choice. I prefer Blu-Ray optical media, because it’s more robust than conventional DVDs, has a rated minimum shelf life that will outlive me (at which point my kid can use the discs as coasters), and can be stored in a bank’s safe deposit box.
Superficially, archiving may seem to be the same process as collaboration, because you’re exporting tracks. However, collaboration often occurs during the recording process, and may involve exporting stems—a single track that contains a submix of drums, background vocals, or whatever. Archiving occurs after a song is complete, finished, and mixed. This matters for dealing with details like Event FX and instruments with multiple outputs. By the time I’m doing a final mix, Event FX (and Melodyne pitch correction, which is treated like an Event FX) have been rendered into a file, because I want those edits to be permanent. When collaborating, you might want to not render these edits, in case your collaborator has different ideas of how a track should sound.
With multiple-output instruments, while recording I’m fine with having all the outputs appear over a single channel—but for the final mix, I want each output to be on its own channel for individual processing. Similarly, I want tracks in a Folder track to be exposed and archived individually, not submixed.
So, it’s important to consider why you want to archive, and what you will need in the future. My biggest problem when trying to open really old songs is that some plug-ins may no longer be functional, due to OS incompatibilities, not being installed, being replaced with an update that doesn’t load automatically in place of an older version, different preset formats, etc. Another problem may be some glitch or issue in the audio itself, at which point I need a raw, unprocessed file for fixing the issue before re-applying the processing.
Because I can’t predict exactly what I’ll need years into the future, I have three different archives.
In this week’s tip, we’ll look at exporting raw WAV files. We’ll cover exporting files with processing (effects and automation), and exporting virtual instruments as audio, in next week’s tip.
Studio One’s audio files use the Broadcast Wave Format. This format time-stamps a file with its location on the timeline. When using any of the options we’ll describe, raw (unprocessed) audio files are saved with the following characteristics:
Important: When you drag Broadcast WAV Files back into an empty Song, they won’t be aligned to their time stamp. You need to select them all, and choose Edit > Move to Origin.
The easiest way to save files is by dragging them into a Browser folder. When the files hover over the Browser folder (Fig. 1), select one of three options—Wave File, Wave File with rendered Insert FX, or Audioloop—by cycling through the three options with the QWERTY keyboard’s Shift key. We’ll be archiving raw WAV files, so choose Wave File for the options we’re covering.
Figure 1: The three file options available when dragging to a folder in the Browser are Wave File, Wave File with rendered Insert FX, or Audioloop.
As an example, Fig. 2 shows the basic Song we’ll be archiving. Note that there are multiple Events, and they’re non-contiguous—they’ve been split, muted, etc.
Figure 2: This shows the Events in the Song being archived, for comparison with how they look when saving, or reloading into an empty Song.
Select all the audio Events in your Song, and then drag them into the Browser’s Raw Tracks folder you created (or whatever you named it). The files take up minimal storage space, because nothing is saved that isn’t data in a Song. However, I don’t recommend this option, because when you drag the stored Events back into a Song, each Event ends up on its own track (Fig. 3). So if a Song has 60 different Events, you’ll have 60 tracks. It takes time to consolidate all the original track Events into their original tracks, and then delete the empty tracks that result from moving so many Events into individual tracks.
Figure 3: These files have all been moved to their origin, so they line up properly on the timeline. However, exporting all audio Events as WAV files makes it time-consuming to reconstruct a Song, especially if the tracks were named ambiguously.
Figure 4: Before archiving, the Events in individual tracks have now been joined into a single track Event by selecting the track’s Events, and typing Ctrl+B.
After dragging the files back into an empty Song, select all the files, and then after choosing Edit > Move to Origin, all the files will line up according to their time stamps, and look like they did in Fig. 4. Compare this to Fig. 3, where the individual, non-bounced Events were exported.
When collaborating with someone whose program can’t read Broadcast WAV Files, all imported audio files need to start at the beginning of the Song so that after importing, they’re synched on the timeline. For collaborations it’s more likely you’ll export Stems, as we’ll cover in Part 2, but sometimes the following file type is handy to have around.
Figure 5: All tracks now consist of a single Event, which starts at the Song’s beginning.
When you bring them back into an empty Song, they look like Fig. 5. Extending all audio tracks to the beginning and end is why they take up more memory than the previous options. Note that you will probably need to include the tempo when exchanging files with someone using a different program.
To give a rough idea of the memory differences among the three options, here are the results based on a typical song.
Option 1: 302 MB
Option 2: 407 MB
Option 3: 656 MB
You’re not asleep yet? Cool!! In Part 2, we’ll take this further, and conclude the archiving process.
A VCA Channel has a fader, but it doesn’t pass audio. Instead, the fader acts like a gain control for other channels, or groups of channels. In some ways, you can think of a VCA Channel as “remote control” for other channels. If you assign a VCA to control a channel, you can adjust the channel gain, without having to move the associated channel’s fader. The VCA Channel fader does it for you.
Inserting a VCA channel works the same way as inserting any other kind of channel or bus. (However, there’s a convenient shortcut for grouping, as described later.) To place a channel’s gain under VCA control, choose the VCA Channel from the drop-down menu just below a channel’s fader…and let’s get started with the applications.
APPLICATION #1: EASY AUTOMATION TRIM
Sometimes when mixing, you’ll create detailed level automation where all the moves and changes are perfect. But as the mix develops, you may find you want to increase or decrease the overall level. There are several ways to do this, like inserting a Mixtool and adjusting the level independently of the automation, or using the automation’s Trim control. However, a VCA control is sometimes easier, and besides, it can control several channels at once if desired, without having to feed them to a bus. The VCA fader can even offset automation for multiple tracks that are located within a Folder Track (Fig. 1)
If the automation changes are exactly as desired, but the overall level needs to increase or decrease, offset the gain by adjusting the VCA Channel’s fader. This can be simpler and faster than trying to raise or lower an entire automation curve using the automation Trim control. Furthermore, after making the appropriate adjustments, you can hide the VCA Channel to reduce mixer clutter, and show it only if future adjustments are necessary.
APPLICATION #2: NESTED GROUPING
One of the most common grouping applications involves drums—when you group a drum kit’s individual drums, once you get the right balance, you can bring their collective levels up or down without upsetting the balance. Studio One offers multiple ways to group channels. The traditional option is to feed all the outputs you want to group to a bus, and vary the level with the bus fader. For quick changes, a more modern option is to select the channels you want to group, so that moving one fader moves all the faders.
But VCAs take this further, because VCA groups can be nested. This means groups can be subsets of other groups.
A classic example of why this is useful involves orchestral scoring. The first violins could be assigned to VCA group 1st Violins, the second violins to VCA group 2nd Violins, violas to VCA group Violas, and cellos and double basses to VCA group Cellos+Basses.
You could assign the 1st Violins and 2nd Violins VCA groups to a Violins Group, and then assign the Violins group, Violas group, and Cellos+Basses group to a Strings group. Now you can vary the level of the first violins, the second violins, both violin sections (with the Violins Group), the violas, the cellos and double basses, and/or the entire string section (Fig. 2). This kind of nested grouping is also useful with choirs, percussion ensembles, drum machines with multiple outputs, background singers, multitracked drum libraries, and more.
Figure 2: The 1st Violins and 2nd Violins have their own group, which are in turn controlled by the Violins group. Furthermore, the Violins, Violas, and Cellos+Basses groups are all controlled by the Strings group.
Although it may seem traditional grouping with buses would offer the same functionality, note that all the channel outputs would need to go through the same audio bus. Because VCA faders don’t pass audio, any audio output assignments for the channels controlled by the VCA remain independent. You’re “busing” gain, not audio outputs—that’s significant.
If you create a group, then all the faders within that group remain independent. Although with Studio One you can temporarily exclude a fader from a group to adjust it, that’s not necessary with VCA grouping—you can move a fader independently that’s controlled by a VCA, and it will still be linked to the other members of a VCA group when you move the VCA fader.
Bottom line: The easiest way to work with large numbers of groups is with VCA faders.
APPLICATION #3: GROUPS AND SEND EFFECTS
A classic reason for using a VCA fader involves send effects. Suppose several channels (e.g., individual drums) go to a submix bus fader, and the channels also have post-fader Send controls going to an effect, such as reverb. With a conventional submix bus, as you pull down the bus fader, the faders for the individual tracks haven’t changed—so the post-fader send from those tracks is still sending a signal to the reverb bus. Even with the bus fader down all the way, you’ll still hear the reverb.
A VCA Channel solves this because it controls the gain of the individual channels. Less gain means less signal going into the channel’s Send control, regardless of the channel fader’s position. So with the VCA fader all the way down, there’s no signal going to the reverb (Fig. 3)
APPLICATION #4: BUS VS. VCA
There’s a fine point of using VCAs to control channel faders. Suppose individual drums feed a bus with a compressor or saturation effect. As you change the channel gain, the input to the compression or saturation changes, which alters the effect. If this is a problem, then you’re better off sending the channels to a standard bus. But this can also be an advantage, because pushing the levels could create a more intense sound by increasing the amount of compression or saturation. The VCA fader would determine the group’s “character,” while the bus control acts like a master volume control for the overall group level.
And because a VCA fader can control bus levels, some drums could go to an audio bus with a compressor, and some drums to a bus without compression. Then you could use the VCA fader to control the levels of both buses. This allows for tricks like raising the level of the drums and compressing the high-hats, cymbals, and toms more, while leaving the kick and snare uncompressed…or increasing saturation on the kick and snare, while leaving other drum sounds untouched.
Granted, VCA Channels may not be essential to all workflows. But if you know what they can do, a VCA Channel may be able to solve a problem that would otherwise require a complex workaround with any other option.
You like to mix with mastering processors in the Main bus to approximate the eventual mastered sound, but ultimately, you want to add (or update) an unprocessed file for serious mastering in the Project page. However, reality checks are tough. When you disable the master bus processors so you can hear the unprocessed sound you’ll be exporting, the level will usually change. So then you have to re-balance the levels, but you might not get them quite to where they were. And unfortunately, one of the biggest enemies of consistent mixing and mastering is varying monitoring levels. (Shameless plug alert: my book How to Create Compelling Mixes in Studio One, which is also available in Spanish, tells how to obtain consistent levels when mixing.)
Or, suppose you want to use the Tricomp or a similar “maximizing” program in the master bus. Although these can make a mix “pop,” there may be an unfair advantage if they make the music louder—after all, our brains tend to think that “louder is better.” The only way to get a realistic idea of how much difference the processor really makes is if you balance the processed and unprocessed levels so they’re the same.
Or, maybe you use the cool Sonarworks program to flatten your headphone or speaker’s response, so you can do more translatable mixes. But Sonarworks should be enabled only when monitoring; you don’t want to export a file with a correction curve applied. Bypassing the Sonarworks plug-in when updating the Project page, or exporting a master file, is essential. But in the heat of the creative moment, you might forget to do that, and then need to re-export.
The Pre-Main bus essentially doubles up the Main bus, to create an alternate destination for all your channels. The Pre-Main bus, whose output feeds the Main bus, serves as a “sandbox” for the Main bus. You can insert whatever processors you want into the Pre-Main bus for monitoring, without affecting what’s ultimately exported from the Main bus.
Here’s how it works.
Figure 1: The Pre-Main bus, outlined in white, has the Tricomp and Sonarworks plug-ins inserted. Note that all the channels have their outputs assigned to the Pre-Main bus.
With all channels selected, changing the output field for one channel changes the output field for all channels. Assign the outputs to the Main bus, play some music, and look at the Level Meter to check the LUFS reading.
Now assign the channel outputs to the Pre-Main bus. Again, observe the Level Meter in the Master bus. Adjust the Pre-Main bus’s level for the best level match when switching the output fields between the Main and Pre-Main bus. By matching the levels, you can be sure you’re listening to a fair comparison of the processed audio (the Pre-Main bus) and the unprocessed audio that will be exported from the Main bus.
The only caution is that when all your channels are selected, if you change a channel’s fader, the faders for all the channels will change. Sometimes, this is a good thing—if you experience “fader level creep” while mixing, instead of lowering the master fader, you can lower the channel levels. But you also need to be careful not to reflexively adjust a channel’s level, and end up adjusting all of them by mistake. Remember to click on the channel whose fader you want to adjust, before doing any editing.
Doubling up the Main bus can be really convenient when mixing—check it out when you want to audition processors in the master bus, but also, be able to do a quick reality check with the unprocessed sound to find out the difference any processors really make to the overall output.
Acknowledgement: Thanks to Steve Cook, who devised a similar technique to accommodate using Sonarworks in Cakewalk, for providing the inspiration for this post.
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