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Category Archives: Friday Tip of the Week

Creating Room Ambiance with Virtual Mics

By Craig Anderton

Supplementing close-miking techniques with room mics gives acoustic sounds a life-like sense of space. Typically, this technique involves placing two mics a moderate distance (e.g., 10 to 20 feet) from the sound source. The mics add short, discrete echoes to the sound being mixed.

This tip’s goal is to create virtual room mics that impart a room sound to electronic or electric instruments recorded direct, or to acoustic tracks that were recorded without room mics. Unlike a similar FX Chain-based tip from over six years ago, this Track Preset (see the download link at the end) takes advantage of a unique Track Preset feature that makes it easier to emulate the sound of multiple instruments being recorded in the same room.

The following trackpreset file will only work with Studio One Professional and Studio One+.

Using the Track Preset

Load the Track Preset Virtual Room Mics.trackpreset (Studio One+ and Professional only). After opening the Mixer view, in Small view you’ll see an audio track and four FX buses (fig. 1).

Figure 1: The Track Preset in the Mixer’s Small view.

The Track Preset includes a stereo audio track. This hosts the sound you want to process. Its four sends go to four FX Channels, each with an analog delay set for a different, short delay time (11, 13, 17, and 23 ms). These are prime numbers so that the delays don’t resonate easily with each other. The delayed sounds produce a result that’s similar to what room mics would produce.

The FX Channels are grouped together, so altering one Room Mic fader changes all the Room Mic faders. The levels are already offset a bit so that longer delays are at a slightly lower level. However, you can edit individual Room Mic faders by holding Opt/Alt while moving a fader. Note: Because the faders are grouped, you can simplify the Mixer view by hiding Room Mics 2, 3, and 4. Then, the remaining Room Mic 1 FX Channel controls the ambiance level.

Under the Hood

Fig. 2 shows the expanded Track Preset.

Figure 2: Expanded Track Preset view.

The Audio track has four post-fader sends. Each goes to its own virtual mic FX Channel with an Analog Delay. Aside from the delay times, they all use the settings shown in fig. 3.

Figure 3: This shows the delay that’s set to 11 ms. The other delays are set identically, except for the delay time.

Using the Virtual Room Mics with More Than One Track

Loading another Virtual Room Mics.trackpreset does not load four more FX Buses. Instead, a new track appears, with its Sends already configured to feed the existing FX Buses. So, you can treat the Virtual Room Mics.trackpreset as a single room for multiple tracks.

Because new tracks appear with Sends already configured, you can vary the send levels slightly for different tracks to place the instruments in different parts of the room. For example, to move the instrument closer to the listener, turn down the sends going to room mics 3 and 4 (with the longest delays), and turn up the sends going to room mics 1 and 2 (with the shortest delays). To place the instrument further away, do the reverse. This more closely emulates recording multiple instruments in the same room. It’s a cool feature of Track Presets used in this type of application.

Workflow Tips

To hear what this FX Chain can do, load a mono Audioloop like Pop > Guitar > Dry > 01a Basement Jam E min. You’ll hear the guitar playing in a room, with a lifelike stereo image.

The main use for this Track Preset is when mixing a combination of acoustic instruments that are miked in a room, and electronic or electric instruments that are recorded direct. Adding room ambiance to the sounds that are recorded direct will let them blend better with the acoustic sounds. It’s best to insert this Track Preset early in the mixing process, so that your mix starts with a consistent acoustic space.

Don’t Make This Mixing Mistake!

By Craig Anderton

Do you think of mixes in absolute terms, or relative terms? Knowing the difference, and when to apply which approach, can make a huge difference in how easily mixes come together. This can also affect whether you’re satisfied with your mixes in the future.

Mixing is about achieving the perfect balance of all of a song’s tracks. When you start mixing, or if you mix in parallel with developing a song, your mixing moves are absolute moves because you haven’t set up the relationship among all the tracks yet. For example, the guitar might be soft compared to the drums and bass, so you increase the guitar’s level. At that point, you don’t yet realize that when a piano becomes part of the mix, the guitar will mask it to some degree. So, now you’ll need to readjust the guitar’s level not only with respect to the drums and bass, but also in relation to the piano.

The further your mix develops, the more important the relative balance among all the levels becomes. Remember: Any change to any track has an influence on every other track. I can’t emphasize that enough.

A Different Way to Finish a Mix

At some point, your mix will be “almost there.” That’s when you notice little flaws. The drums are a bit overpowering. The bass needs to come up. The background singers don’t have quite the right balance with the lead vocal. Two keyboard parts are supposed to be the same level, but one is slightly louder.

The absolute approach to addressing those issues would be to make those changes. The kick comes down a bit. The bass comes up. You balance out the background singers and the keyboards. Then you render another mix to see if the problems have been addressed. It’s better, but now the bass is masking the low end of the keyboards. So, you bring up the keyboards a bit, but now they step on the background vocals…

If you’re not concentrating on how the tracks fit together in relative terms, then you’ll constantly be chasing your tail while mixing. You’ll keep making a series of absolute adjustments, and then wonder why relatively speaking, the mix doesn’t gel.

The Relative Approach to Mixing

VCA Channels are the key to relative mix edits, because they can offset tracks easily compared to the rest of the mix. Take the example above of the drums being a bit overpowering, the bass too soft, etc. Rather than try to fix them all at the same time, here’s what I do:

1. Choose the issue that seems most annoying. Let’s suppose it’s the drums being overpowering. I always start with fixing tracks that are too loud instead of too soft, because lowering the level of the loud track will make all the other tracks louder, relatively speaking.

2. Select the drum tracks (or drum bus) and choose “Add VCA for Selected Channels.”

3. Lower the VCA channel for the drums by (typically) -0.5 dB, but no more than -1.0 dB.

4. Not change any other track levels. Now it’s time to render a new version of the mix, and live with it for a day.

Having softer drums will change the relative perspective of the entire mix. Maybe the bass wasn’t that soft after all; maybe it was just masked a bit by the kick. Maybe the rhythm guitar is actually louder than it seemed, because its percussive strums were blending in with the drum hits—but the strums weren’t noticeable until the drums were softer. And so on.

That -0.5 dB of difference will change how you hear the mix. -0.5 dB may not seem like much, but that’s just one perspective. A different perspective is that it’s making every other track +0.5 dB louder than the drums. So, you need to evaluate the mix with fresh ears, because that one change has altered the entire mix.

An advantage of using VCA channels is that when you add the VCA Channel, its initial setting is 0.0. It’s easy to see how much you’ve offset the track level with the VCA, compared to (for example) changing a drum bus fader from -12.6 to -13.1. It’s also easy to get back to where you started in case after listening to the track, you decide other tracks were the problem, and the drums need to return to where they were. Just reset the VCA to 0.0.

Let’s suppose that after listening to the rendered version a few times at different times of the day, it seems like the drums fit in much better with the overall mix. Make the change permanent by de-assigning the tracks to the VCA Channel, and then removing the VCA Channel. (Or, leave it in and hide it if you think you might need more changes in the future.)

Next, let’s suppose the bass still seems a little soft. I’ll repeat the four steps listed above, but this time with the bass track, and raise it by +0.5 dB (fig. 1). Then it’s time to render the track again, and live with it for a day.

Figure 1: A VCA channel has altered the drum mix by -0.5 dB. That VCA Channel is about to be removed, because -0.5 dB turned out to be the right amount. Meanwhile, a VCA Channel has been added to see if increasing the Bass level by +0.5 dB helps it fit in better with the mix.

It might seem that this one-track-at-a-time approach would take forever, especially because sometimes you may need to revise earlier changes. But it can save time, for two reasons:

  • Mixing sessions don’t go on for hours. Because you listened to the rendered mix with fresh ears and know what you need to change, you make the change. After rendering the new mix, you’re done for the day, aside from listening to it several times under various conditions. Your final mixes now become 5 to 10 minutes at a time spread over multiple days. An additional advantage is that you always hear the mix with fresh ears, instead of having listener fatigue set in during a long mixing session.
  • Often, after taking care of the most problematic tracks, other issues resolve themselves because they weren’t the problem—their relationship to the problematic tracks was the problem.  Fixing those other tracks fixes the relationship.

If after repeated listening over a few days (and being brutally critical!) I don’t hear anything that needs to change, then the song is done.

A Corollary to Relative Mixing

This approach is also one reason why I don’t use dynamics processors in the master bus, except for the occasional preview. All dynamics processors are dependent on input levels. As you change the relationship of the tracks, you’re also changing how a master bus’s dynamics processor influences your mix.

Some people say they need to mix through a dynamics processor, because the mix doesn’t sound right without it. I think that may be due to mixing from an absolute point of view, and the dynamics processor blurs the level differences. I believe that if you achieve the right relative balance without using a master bus dynamics processor, when you do add dynamics processing during the mastering process, the balance will remain virtually identical. Your mix will also gain the maximum benefits from the dynamics processing.

Once you start considering when to employ a relative mixing approach compared to a more absolute approach, I think you’ll find it easier to finish mixes—and you’ll end up with mixes you’re satisfied with years later.

Tuff Beats

By Craig Anderton

Calling all beats/hip-hop/EDM/hard rock fans: This novel effects starts with drums modulating the Vocoder’s white noise carrier, and takes off from there. The sound can be kind of like a strange, aggressive reverb—or not, because the best part of this tip is the crazy variety of sounds that editing or automating parameters can create.

The following audio example plays just a few of the possibilities. The first two measures are the original loop. Then, several 2-measure examples alter Vocoder parameters.

Track Layout

Fig. 1 shows the track layout:

  • The Drums track hosts the waveform that modulates the Vocoder via a pre-fader send to the Vocoder track. How you set the Drums track fader depends on whether or not you want to mix in unprocessed sounds.
  • The Carrier track generates white noise as a carrier for the vocoder’s sidechain (fig. 2), as sent through a pre-fader send. You’ll probably want to keep the Carrier track’s fader at minimum.
  • The Vocoder track produces the processed output to mix in with the drums.

Figure 1: Track layout for Tuff Beats processing.

Figure 2: Tone Generator settings.

Editing the Effect

Figure 3: Typical Vocoder settings.

The only crucial setting is that the Carrier Source must be set to Side-Chain (fig. 3). Aside from that, you have plenty of options for subverting the sound:

  • Release. At longer release times, the sound is like a strange reverb. Shorter settings are more like doubling.
  • Release automation. Try drawing waveform automation with the Paint tool, like negative-going sawtooth waves and triangle waves. Freehand drawing can produce even wilder effects.
  • Attack. Turning up Attack reminds me of a transient shaper, because it softens the drum attack.
  • Patch Matrix. This alters the “reverb” character. You can get some pretty whacked out filtering effects.
  • Matrix automation. Now you can really go insane. Choose Write for the Vocoder’s automation mode, and “draw” on the Patch Matrix as you would an Etch-a-Sketch. This changes the filtering effects, and the automation remembers your moves.

It doesn’t take much effort to come up with some pretty novel sounds, so…have fun!

Reinvent Your Stereo Panning

This tip is about working with stereo, NOT about Dolby Atmos® or surround—but we’re going to steal some of what Atmos does to reinvent stereo panning.  Studio One’s Surround panners are compatible with stereo projects, offer capabilities that are difficult to implement with standard panpots, and are  easy to use. Just follow the setup instructions below, and start experimenting to find out how surround panning affects stereo tracks. (Surround panners work with mono tracks too, although of course the stereo spread parameter described later is irrelevant.)


When you’re ready to mix, choose Song > Spatial Audio. Select the parameter values to the left in fig. 1. In the output section (fig. 1 right), select 5.0 for the Bed format, and Stereo for both Speakers and Headphones so you can use either option to monitor in stereo.

Figure 1: Parameter setup for Surround panning with stereo projects.

After choosing Dolby Atmos for spatial audio, channel panpots turn into surround panpots. Double-click on them to see the “head-in-middle-of-soundfield” image shown below. Choose Disable Center, which isn’t used. LFE Level doesn’t matter, unless you’re using a subwoofer.

Using surround panners for stereo offers several adjustable parameters:

Spread. Move the L and R circles to set the left and right pan position spread, or click and drag in the numeric Spread field. The spread (fig. 2) can go from 0 (mono), to 100% (standard panning), to 200% (extra wide, like binaural panning).

Figure 2: (Left to right) 14.8% spread, 100% spread, 200% spread.

Direction. After establishing the spread, click on the arrow and rotate the spread so it covers the desired part of the stereo field (fig. 3). You can also click and drag on the numeric Direction field. Between spread and direction, you can “weight” the stereo spread so that it covers only a sliver of the stereo field, covers center to right or left, mostly left, mostly right, etc.

Figure 3: (Left to right) Panned from left to center, panned to a narrow slice of the stereo field, and panned almost full but tilted toward the right.

Size. This has no equivalent with stereo panpots. Click on the arrow, and move it closer to the head for a “bigger” size, or further from the head for a “smaller” size (fig. 4). You can also click and drag on the numeric Size field. The result isn’t as striking as with true surround, but it’s much more dramatic than standard panning. Note the “cloud” that shows how much the sound waves envelope the head. All the previous images showed a small size.

Figure 4: (Left to right) Biggest size/least distance, moderate size/moderate distance, smallest size/furthest distance.

Flexible automation. A joystick or controller pad can automate two of the parameters simultaneously. Or, modulate all three parameters using three controls from a control surface. This is a huge deal compared to standard panning. For example, suppose an instrument is ending a solo, while another solo starts. The one that’s ending can pan to a narrower spread, move off to the side, and become smaller just by moving three controls.

Other Features

  • Balance Tab. The surround panners can also serve as conventional balance controls. This setting interacts with the panners. For example, if the L and R buttons are close to each other, there won’t be much balance to adjust. I rarely use the balance option. To make sure that surround panning isn’t altered by a Balance parameter setting, check that the balance “dot” is in the center of the virtual head.
  • Size lock. This maintains the same Size setting, regardless of what you do with the Direction and Spread parameters. Hold Shift to bypass Lock temporarily, and fine-tune Size.
  • Object Panner. Right-click on the Surround panner, and you can choose an Object Panner instead. This is less relevant with stereo, because front/back and lower/higher directionality doesn’t exist like it does in a true Atmos system. However, the Object Panner does have Size, Spread, and Pan X (left/right) parameters, so feel free to play around with it—you may like the interface better than the surround panner. It’s also possible to do crazy automation moves. In any case, you can’t break anything.
  • Other. This is another function you reach via a right-click on the Surround Panner. You’ll see a list of other plugins on your system that may have spatial placement abilities, like Waves’ Nx series of control room emulators, Brauer Motion, Ozone Imager, Ambisonics plugins, and the like. If you insert one of these, you can revert to the stock Studio One panners or choose other options by clicking on the downward arrow just under the “other” plugins name.

It may sound crazy to use Surround panners in stereo projects—but try it. You can truly do stereo panning like never before.

Presence Electric 12-String (the Artist Version Remix)

Presence’s sound library includes a fine acoustic 12-string guitar, but not an electric one. So, perhaps it’s not surprising that one of the more popular blog posts in this series was about how to create a realistic electric 12-string preset with Presence.

Unfortunately, that was before Studio One introduced Track Presets. The preset relied on a Multi Instrument, so it worked only with Studio One Professional. However, thanks to Track Presets, we can revisit our electric 12-string, and make a plug ’n’ play version that works for Studio One Artist as well as Professional (download link at the end).

Overcoming the Sampling Problem with 12 String Guitars

Sampling a 12-string is difficult. The sound is constantly changing due to the shimmering effect from slightly detuned strings. Furthermore, some notes are doubled with octave-higher notes, while other notes are doubled with unison notes. My solution is not to try and sample a 12-string guitar, but to construct one from three sets of 6-string guitar samples.  

Each of the three Presence instances (fig. 1) loads the preset Guitar > Telecaster > Telecaster Open from the stock Presence library:

  • One instance provides the main guitar sound.
  • Transposing another instance up 12 semitones provides the octave-above notes.
  • A physical 12-string guitar doesn’t have octaves on the 1st and 2nd strings, so the third Presence instance provides a unison sound for the higher strings.

Figure 1:  With three instances needed to create a single instrument, note that all three Monitor buttons must be enabled to play the instrument from your keyboard or MIDI guitar controller.

Limiting Note Ranges

In the original preset, Multi Instrument Range edits prevented the unison sounds from overlapping with the octave-above sounds. In the Artist preset, two Input Filter Note FX restrict the ranges (fig. 2).

Figure 2: The upper Note FX Input Filter restricts the range of the octave-above notes to A#2 and below. The lower Note FX Input Filter restricts the range of the unison strings to B2 and above.

Emulating the 12-String “Shimmer”

A 12-string is never perfectly in tune, which gives a shimmering effect. The octave instance is transposed up +12 semitones, but the Pitch Fine Tune setting is +5 cents. The unison instance Pitch Fine Tune is -2 cents. This gives the chorus-like that’s inherent in 12-string guitars. Detuning the virtual strings provides a more realistic sound than trying to “fake it” with a time-based modulation effect.

About the Analog Delays

The higher string in a pair of strings plays just a little bit late, because your pick hits the main string before the octave or unison string. To emulate this effect, the Analog Delay (fig. 3) provides a 20 ms delay for the octave and unison instances. (We can’t use Presence’s Delay, because the mix needs to be 100% delay—no dry sound.)

Figure 3: Analog Delay settings used to emulate string pluck delay.

Without this delay, the emulated 12-string lacks realism. The Analog Delay also adds some High Cut to reduce some of the brightness caused by transposing the octave strings. The Width settings provide a big stereo image, but for a more “normal” sound, turn ping-pong mode to Off.

The octave and unison instance levels are -6 dB below the main guitar sound. With physical 12-string guitars, the octave strings are thinner than the strings that generate the standard pitch. So, they generate less output. Lowering the level of the octave strings gives a better overall balance. Technically, the unison strings could be at the same level, but their levels are also a little lower to avoid an unbalanced sound compared to the octave strings.

EQ Settings

The Octave and Unison string instances use Presence’s internal EQ to attenuate the highest and lowest frequency bands. The main instance attenuates the low band, but peaks +3 dB at 3 kHz. Regarding the Pro EQ3 settings, open up the preset if you want to deconstruct the programming. The main aspects are a bass cut to give a more trebly, “Ricky”-like 12-string sound, a high-shelf boost for a little extra brightness, and a narrow notch around 3.2 kHz to reduce some “string ping” inherent in the original samples. As always, though, adjust for your tastes in guitar tone.

So, Studio One Artist aficionados, what are you waiting for? Download the preset, and get ready to make some cool electric 12-string sounds.

Download CA 12-String Electric Artist.trackpreset here!

Faster, Simpler, and Better Comping

At first, this might not seem too exciting. But follow the directions below, and try comping using this method—I don’t think you’ll be disappointed. This tip shows how to:

  • Audition, select sections of, and promote Takes while listening to the rest of the mix, at any level you want.
  • Listen to the edited Parent track made up of the Takes you’ve promoted, at any time during the comping  process. Again, this is in context with the mix.
  • Do all of the above while looping, so there’s never a break in the editing process.
  • Do comping with only the Arrow tool—you don’t need the Listen tool.

Preparation: Set Up Dim Solo

First, implement the Dim Solo function described in the blog post Super-Simple Dim Solo Functionality. Dim Solo allows soloing a track or tracks, while all the other tracks are at an adjustable lower level. The process works by assigning all tracks except for the one you want to solo (e.g., a vocal track with its Take layers) to a VCA channel. You can then “dim” all the non-soloed tracks with the VCA level fader to whatever level you want while you comp, and hear the Takes in context with the song. After auditioning and selecting the desired sections of your Takes, set the VCA fader back to 0.0 to return to the original mix levels. The minute or two it takes to set up Dim Solo is more than offset by the benefits it offers to comping. For more details, refer to this blog post for how to create the Dim Solo function.

Faster Take Auditioning, Selecting, and Promoting

After setting up Dim Solo and using the VCA Channel fader to adjust the level of the mix (which excludes the track being comped, because it isn’t part of the VCA group), here’s how to audition and select Takes:

1. Safe Solo (Shift+Click) the parent track with the Takes. This is important! It allows soloing the Parent track without muting the tracks that are playing back at the dimmed level.

2. Loop the section with the Takes you want to audition.

3. Click a Take’s Solo button to audition it while the song loops (fig. 1).

Figure 1: Take 3 is being soloed for auditioning, and for selecting sections to be promoted to the Parent track. Turning off the Take’s Solo would solo the Parent track, so you could audition the edited parent track and hear any Take sections that had been promoted.

4. If you hear a section in a Take you want to promote to the Parent track, use the Arrow tool (which turns into an I-beam cursor when hovering over a selected Take) to click+drag over the section.

5. Continue soloing Takes while the music loops, and select the sections you want to promote to the Parent track. If needed, alter the loop start and end points.

6. If at any time you want to hear the edited Parent Track with the Takes you’ve promoted up to that point, make sure no Take layers are in solo mode.

Better Music Through Better Comping

One reason I wrote up this tip is because of an interesting side effect. The Takes I selected as “best” when auditioned in the usual way were often not the same Takes chosen as “best” when listening to them in context with the music. A technically perfect Take is not necessarily the same thing as a Take with the best feel. Listening to, selecting, and promoting the Takes in context with the mix makes a big difference in helping to select Takes that fit the music like a glove.

How to Quickly Slash Your Latency

You know the feeling: You’re tracking or doing an overdub with a virtual instrument or amp sim, but you’re frustrated by the excessive latency inherent in complex projects with lots of plugins. And with older computers, latency may be an annoying fact of life.

Of course, Studio One has clever low-latency native monitoring. However, there are some limitations: plugins can’t introduce more than 3 ms of latency, FX Chains can’t use Splitter devices, and external effects using the Pipeline plugin are a non-starter.

This tip’s universal technique has only one significant limitation: it’s oriented exclusively toward having the lowest latency when tracking or doing overdubs. Fortunately, most of the time that’s when low latency is most important. Latency doesn’t matter that much when mixing down.

Here’s the process:

1. Make a premix of all tracks except the one with the virtual instrument or amp sim you want to track with or overdub. Do this by exporting the mix (Song > Export Mixdown) and checking Import to Track (fig. 1). The imported track becomes a premix of your tracks. Note that if any of the tracks use Pipeline, the premix must be done in real time.

Figure 1: The first step is to create a Premix of all your tracks. Make sure you select Import to Track (outlined in white).

2. Select all tracks except for the Premix and the one with the virtual instrument or amp sim you want to use for your overdub.

3. In the Arrange view, right-click on the selected tracks. Choose Disable Selected Tracks (fig. 2).

Figure 2: In this example, all the tracks are selected for disabling, except for the Mai Tai instance in Track 23 and the Premix.

4. Now you can overdub or track while listening to the premix. Because there’s now so little load on the CPU (fig. 3), you can reduce the Device Block Size and Dropout Protection (under Studio One > Options > Audio Setup) to lower the latency.

Figure 3: The Performance Meter on the left shows CPU consumption with all tracks except the Mai Tai and Premix disabled. The Performance Meter on the right shows CPU consumption with all the tracks enabled.

5. While listening to the premix as your reference track, you’ll be able to play your virtual instrument or through your amp sim with much lower latency.

6. When you’re done with your overdub or tracking, you can delete or mute the Premix, and return the latency to a higher setting that allows for mixing without dropouts or other problems.

How to Fix Phase Issues

Recording audio using more than one feed from the same source may create phase issues. For example, when miking a bass amp and taking a DI (dry) input, the DI’s audio arrives at your interface instantly. But because sound travels 12″ (30 cm) in 1 millisecond, the mic’s audio will arrive later due to the distance between the mic and speaker. This means it won’t be time-aligned with the direct sound, so there will be a phase difference.

Miking an acoustic guitar with two mics, or drum overheads that are a distance from the drum kit, may also lead to potential phase problems. Even partial phase cancellations can thin or weaken the sound.

It’s best to check for phase issues and fix them prior to mixing. One way to resolve phase issues is to look at the two waveforms on the timeline, and line them up visually. However, the waveforms may not look that similar (fig. 1), especially when comparing audio like a room mic with a close-miked sound.

Figure 1: Despite being zoomed in, you wouldn’t necessarily know from the waveforms that they are at their point of maximum cancellation.

Another option is to sum the sources in mono, delay or advance one relative to the other, and listen for what sounds the strongest. That can work, but it’s not always easy to hear exactly when the waveforms are in phase. Let’s make the process more foolproof.

The Phase Correction Process

The goal is to find the alignment of the two waveforms where the audio sounds the strongest. However, it’s easier to hear phase cancellation than addition. So, by inverting the phase of one channel, we can align the tracks for maximum cancellation. Returning the phase to normal then gives the strongest sound. Here’s the process:

1. Solo the channels that may have a potential phase issue. Determine whether they do by listening for a thinner sound when they’re summed to mono, or both panned to center.

2. Match the channel levels as closely as possible.

3. Pan the two channels hard left and hard right.

4. Insert the Phase Meter plugin in the master bus.

5. Insert a Mixtool in one of the instrument channels. Invert the Mixtool’s phase. If a track uses stereo audio, invert the left and right channels.

6. Turn off Snap. This allows nudging an Event in 1 millisecond increments. Select the Event and type Alt+Right Arrow to move the track earlier by 1 ms, or Alt+Left Arrow to move the track later by 1 ms.

7. Nudge one of the tracks with respect to the other track until the Phase Meter shows the most negative correlation (i.e., the lower bar graph swings to the left as much as possible, as in fig. 2). This indicates maximum cancellation. Note: The time range for maximum cancellation is extremely narrow. One millisecond can make the difference between substantial cancellation and no cancellation. Be patient. Adjust 1 ms at a time.

Figure 2: A negative correlation reading indicates that the two sources have phase differences.

8. You’ve now aligned the two tracks. Remove the Mixtool plugin so that the two signals are in phase.

Tip: For audible confirmation, in Step 7 listen as you make these tiny changes. If you enable the Mono button in the Main bus, aim for the thinnest possible sound. If you monitor in stereo, listen for the narrowest stereo image.

Super-Fine Tuning

1 ms is a significant time shift when trying to match two signals’ phase. Step 7 will isolate a 1 ms window where the two tracks are in phase (or at least close to it), but you may be able to do better.

Select Samples for the Timebase and turn off Snap. Zoom in far enough, and you’ll be able to click on the Event and drag it earlier or later in 1 sample increments (fig. 3). That can tighten the phase further, if needed.

Figure 3: With Timebase set to Samples and Snap turned off, the resolution for moving an Event is 1 sample.

This may be overkill, but I know some of you are perfectionists 😊. Happy phase phixing!

Bigger, Wider Sounds: Try Stereo Miking

If you haven’t experimented yet with mid-side stereo miking, you’ll be in for a treat when you do. Here’s why:

  • Record background singers with gorgeous stereo imaging
  • Capture a guitar amp with stereo room sound
  • Run synths and drum machines through a PA or amp, and obtain the vibe of recording electronic instruments in an acoustic space
  • Capture full stereo with choirs, plays, podcasts, seminars…you name it

Even better, mid-side stereo miking collapses perfectly to mono, and Studio One Professional bends over backward to make the recording process easy.

Wait—Is This Déjà Vu?

The post Mid-Side Processing with Artist covered mid-side processing for Studio One Artist, and Mid-Side Processing Made Easy did the same for Studio One Professional. However, these were only about processing mid and sides audio extracted from an already recorded stereo track.

This post is about recording with mid-side miking, and monitoring the result as it’s being recorded. This lets you achieve optimum mic placement by hearing the results of changes in mic positioning, in real time. You don’t have to record the audio first, and then play it back to see if you got it right. And on playback, there’s no need to extract the mid and sides, because you already recorded them as separate tracks.

Mid-Side Recording? I’m Already Intimidated!

You needn’t be. You might have heard about mid-side recording requiring math, phase changes, encoding, decoding, a degree in physics, etc. Nope! As I said, Studio One Professional makes it easy.

There are many ways to record in stereo. The most common is X/Y miking, where one directional mic points toward the left, and one toward the right. But this technique is not ideal. Directional mics color sound coming in from outside the pickup pattern, and the two signals don’t play nice when collapsed to mono.

Mid-side miking uses a mic with a cardioid pickup pattern to capture the center of a stereo sound, and a mic with a bi-directional (aka figure-8) pickup pattern to capture the left and right sides. Ribbon mics inherently have a figure-8 pickup pattern, but there are also multi-pattern condenser mics that can be switched to a figure-8 pickup pattern. With this pickup pattern, the mic’s front and back pick up sound, while the sides don’t pick up any sound.

Fig. 1 shows how you’d set up the two mics for mid-side stereo recording. Pretend you’re a guitar amp looking at the mics. The top, figure-8 mic is mounted upside down because the diaphragms of the two mics need to be as close together as possible. The figure-8 mic’s side faces you. Its front faces to the left, and its rear faces to the right. Meanwhile, the cardioid mic is pointing at you.

Figure 1: Mid-side miking setup. (Mic images courtesy Sweetwater Publishing)

The Studio One Connection

You need two tracks and mixer channels, one for the cardioid (mid) mic and one for the figure-8 (sides) mic. Record the two mics like you would any other mics. However, this is where many people get lost—it’s counterintuitive that the single mic cable coming from the figure-8 mic can turn into right and left channels in Studio One. Fortunately, Studio One knows how to do this.

Figure 2: Studio One FX Chain for mid-side recording.

Fig.2 shows audio being recorded. Pan the Cardioid mic to center for your mid (center) audio. For the Sides mic, set the track input to mono, but set the track mode to stereo (important!). This combination allows recording in mono, but playing back in stereo.

A Splitter in Channel Split mode splits the mono audio into two paths. All the Mixtool plugin does is invert the right channel path’s polarity. Now the left and right channels are out-of-phase, so there’s a wide stereo spread. At the mixer, set the Sides channel’s Pan mode to Dual Pan. Pan the controls hard left and right. Now we have the Splitter’s left channel audio panned to the left, and the right channel’s audio panned to the right.

You might think “wait a second…if the left and right channels are out-of-phase, then if they play back in mono, they’ll cancel.” Exactly! And that’s why mid-side miking offers superb mono compatibility. When the sides cancel, you’re left with the mono audio from the cardioid mic, which also picks up sounds that are in front of the mic to the left and right.  So, you’re not losing the audio that ends up being stereo—just the stereo imaging. And of course, mono audio has no problem being played back in mono.

Now, Here’s the Really Cool Part

You don’t have to understand how all this works, just create the setup described above. The Sides channel varies the level of the stereo image, and the Mid channel varies the mono center. So, you can make the stereo image wider by raising the level of the Sides channel, and lowering the level of the Mid channel. Or, lower the Sides channel for a narrower image. Or, process the two separately, and gain the advantages of mid-side stereo processing.

Go ahead—bathe your ears with stereo recording instead of mono. As long as you have one mic with a directional pickup pattern, one mic with a figure-8 pickup pattern, and two mic stands, you’re good to go. And there’s a bonus: if you start getting involved with Atmos mixing, having stereo audio gives you more fun options than mono.

How to Prioritize Vocals with Mix Ducking

This complements the tip Better Ducking for Voiceovers and Podcasts and the tip Why I Don’t Use Compression Anymore. It applies the concept of voiceover ducking to your entire mix. Here’s the TL;DR: Insert the Pro EQ3 in a master bus, feed its sidechain from your vocal track, and adjust the Pro EQ3’s dynamic EQ to reduce the vocal frequencies in the stereo mix. When done subtly, it makes the voice stand out more, because the mix behind it stands out less.

Create a Secondary Main Bus

Inserting the Pro EQ3 in the Main bus won’t work, because the vocal goes to the Main bus. So, any dynamic EQ would affect the voice as well as the mix, which we don’t want. The solution is to create a secondary Main bus. We’ll call it the Sub Bus.

1. Select all your tracks in the mix (click on the lowest-numbered track and shift+click on the highest-numbered track).

2. Ctrl/Cmd click on the vocal track to de-select it. The other tracks should still be selected.

3. Right-click on one of the selected tracks and choose “Add Bus for Selected Channels” (fig. 1).

Figure 1: All tracks are selected except for the Vocal track (7). They’re about to be re-assigned from the Main bus to the new Sub Bus.

4. The new Sub Bus feeds the Main bus. Confirm that all track outputs go to the Sub Bus except for the Vocal track, which still goes to the Main bus. Now we can process the Sub Bus independently of the Vocal.

5. Insert a Pro EQ3 into the Sub Bus. Insert a Send from the Vocal track to the Pro EQ3’s sidechain. Fig. 2 is a simplified signal flow diagram.

Figure 2: All tracks except the Vocal go to the Sub Bus. The Sub Bus out and Vocal out go to the Main bus. A send from the Vocal track drives the sidechain of a Pro EQ3 inserted in the Sub Bus.

6. Now comes the most important part—choosing the optimum dynamic EQ settings. The goal is to add dynamic cuts at frequency ranges that correspond to the voice. The settings in fig. 3 are a good start for experimenting with this process:

  • The Frequencies are set an octave apart
  • The Ranges are at their maximum negative values
  • The Q settings are 2.0
  • The Sidechain is enabled for the signal coming from the vocal

7. Adjust the Threshold for each stage so the vocal peaks trigger a subtle cut. Fig. 3 shows a range of 12 dB, so the cuts shown are around -2 dB to -3 dB. That may not seem like much, but it’s sufficient to open up space for the vocal.

8. Optimize the setting for each Threshold parameter. To compare the processed and unprocessed sound, turn off the Vocal track’s Send. Don’t disable the Sidechain at the Pro EQ3 instead, or the EQ will respond to the mix’s dynamics instead of the vocal’s.

Figure 3: The violet waveform is the Vocal feeding the sidechain. The blue waveform is the stereo mix. The colored curves are the five EQ stages. Set LF and HF to Peaking mode. The wavy white line toward the center shows how much of a cut was occurring when this screen shot was taken.

9. Use the Vocal track’s Send slider to fine-tune the level going to the Pro EQ3’s sidechain.

This technique assumes the vocal has a consistent dynamic range, either from compression or from the techniques in the blog post about why I don’t use compression anymore. Otherwise, loud parts will push the background down further—but a lower level isn’t needed when the vocal is at its loudest. If the vocal’s dynamics are an issue, automate the Vocal track’s Send to reduce the amount of cut for loud vocal sections.

Because there are so many variables, there are many ways to optimize the sound:

  • Narrowing the Q may provide much of the benefits of this technique without affecting the submix as much.
  • You might not want to set the Range parameter as low as it is in fig. 3, so that there are “guardrails” against the Threshold settings being too low and causing too deep of a cut.
  • Broadening the Q makes the effect cover a wider frequency range, so you may not need to cut as much (fig. 4). This is more like the traditional ducking used in voiceovers.

Figure 4: Broader Q settings give less focused cuts.

Remember that this technique’s intention is to add a subtle enhancement—it’s not a cure for a mix, arrangement, or vocal that needs work. However, it can provide that extra little something that makes a vocal stand out a bit more and fit more comfortably into a mix.