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Video Using VocALign in Studio One Pro!

 

In this video, producer Paul Drew shows how VocALign seamlessly works inside Presonus Studio One Professional and almost instantly aligns the timing of multiple vocal tracks to a lead using ARA2, potentially saving hours of painstaking editing time.

ARA (Audio Random Access) is a pioneering extension for audio plug-in interfaces. Co-developed by Celemony and PreSonus, ARA technology enhances the communication between plug-in and DAW, and gives the plug-in and host instant access to the audio data. This video shows Studio One but the workflow is very similar in Cubase Pro & Nuendo, Cakewalk by Bandlab and Reaper.

Purchase VocALign today right out of the PreSonus Shop!

Friday Tips: The Best Flanger Plug-In?

Well…maybe it actually is, and we’ll cover both positive and negative flanging (there’s a link to download multipresets for both options). Both do true, through-zero flanging, which sounds like the vintage, tape-based flanging sound from the late 60s.

The basis of this is—surprise!—our old friend the Autofilter (see the Friday Tip for June 17, Studio One’s Secret Equalizer, for information on using its unusual filter responses for sound design). The more I use that sucker, the more uses I find for it. I’m hoping there’s a dishwashing module in there somewhere…meanwhile, for this tip we’ll use the Comb filter.

Positive Flanging

Figure 1: When set properly, the Autofilter’s comb filter can make a superb flanger.

Flanging depended on two signals playing against each other, with the time delay of one varying while the other stayed constant. Positive flanging was the result of the two signals being in phase. This gave a zinging, resonant type of flanging sound.

Fig. 1 shows the control settings for positive flanging. Turn Auto Gain off, Mix to 100%, and set both pairs of Env and LFO sliders to 0. Adding Drive gives a little saturation for more of a vintage tape sound (or follow the May 31 tip, In Praise of Saturation, for an alternate tape sound option). Resonance is to taste, but the setting shown above is a good place to start. The Gain control setting of 3 dB isn’t essential, but compensates for a volume loss when enabling/bypassing the FX Chain.

 

Varying the Cutoff controls the flanging effect. We won’t use the Autofilter’s LFO, because real tape flanging didn’t use an LFO—you controlled it by hand. Controlling the flanging process was always inexact due to tape recorder motor inertia, so a better strategy is to automate the Cutoff parameter, and create an automation curve that approximates the way flanging really varied (Fig. 2)—which was most definitely not a sine or triangle wave. A major advantage of creating an automation curve is that we can make sure that the flanging follows the music in the most fitting way.

Figure 2: The Cutoff frequency automation curve used for the audio examples.

Negative Flanging

Throwing one of the two signals used to create flanging out of phase gave negative flanging, which had a hollower, “sucking” kind of sound. Also, when the variable speed tape caught up with and matched the reference tape, the signal canceled briefly due to being out of phase. It’s a little more difficult to create negative flanging, but here’s how to do it.

  1. Set up flanging as shown in the previous example, and then Duplicate Track (Complete), including the Autofilter.
  2. Turn Resonance all the way down in both Autofilters (original and duplicated track). This is important to obtain the maximum negative flanging effect. Because of the cancellation due to the two audio paths being out of phase, there’s a level drop when flanging. You can compensate by turning up both Autofilters’ Gain controls to exactly 6.00 dB (they need to be identical). This gain increase isn’t strictly necessary, but helps maintain levels between the enabled and bypassed states of the Negative Flanging FX Chain.
  3. In the duplicated track’s Autofilter, turn off the Autofilter’s Automation Read, and turn Cutoff up all the way (Fig. 3).

 

 

Figure 3: Settings for the duplicated track’s Autofilter.

  1. Insert a Mixtool after the Autofilter in the duplicated track, and enable both Invert Left and Invert Right (Fig. 4). This throws the duplicated track out of phase.

 

 

Figure 4: Mixtool settings for the duplicated track.

  1. Temporarily bypass both Autofilters (in the original and duplicated tracks). Start playback, and you should hear nothing because the two tracks cancel. If you want to make sure, vary one of the track faders to see if you hear anything, then restore the fader to its previous position.
  2. Re-enable both Autofilters, and program your Cutoff automation for the original track (the duplicated track shouldn’t have automation). Note that if you mute the duplicate track, or bring down its fader, the sound will be positive flanging (although with less level than negative flanging, because you don’t have two tracks playing at once).

 

So is this the best flanger plug-in ever? Well if not, it’s pretty close…listen to the audio examples, and see what you think.

 

 

Both examples are adapted/excerpted from the song All Over Again (Every Day).

 

The Multipresets

If you like what you hear, download the multipresets. There are individual ones for Positive Flanging and Negative Flanging. To automate the Flange Freq knob, right-click on it and choose Edit Knob 1 Automation. This overlays an automation envelope on the track that you can edit as desired to control the flanging.

 

Download the Positive Flanging Multipresets Here! 

 

Download the Negative Flanging Multipresets Here! 

 

And here’s a fine point for the rocket scientists in the crowd. Although most flangers do flanging by delaying one signal compared to another, most delays can’t go all the way up to 0 ms of delay, which is crucial for through-zero flanging where the two signals cancel at the negative flanging’s peak. The usual workaround is to delay the dry signal somewhat, for example by 1 ms, so if the minimum delay time for the processed signal is 1 ms, the two will be identical and cancel. The advantage of using the comb filter approach is that there’s no need to add any delay to the dry signal, yet they can still cancel at the peak of the flanging.

Finally, I’d like to mention my latest eBook—More Than Compressors – The Complete Guide to Dynamics in Studio One. It’s the follow-up to the book How to Record and Mix Great Vocals in Studio One. The new book is 146 pages, covers all aspects of dynamics (not just the signal processors), and is available as a download for $9.99.

 

 

End Boring MIDI Drum Parts!

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

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

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

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

 

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

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

 

 

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

 

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

 

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

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

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

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

 

Friday Tips: Attack that Autofilter!!

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

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

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

 

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

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

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

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

Friday Tips: How to Normalize Comped Takes

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

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

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

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

 

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

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

 

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

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

 

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

 

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

Friday Tips: The Center Stage Reverb

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

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

 

 

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

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

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

 

 

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

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

 

 


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

Friday Tips: The Dynamic Brightener for Guitar

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

How It Works

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

 

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

 

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

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

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

Friday Tips: Crazee Snare Enhancements!

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

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

Skeptical? Check out the audio example.

 

ISOLATING THE SNARE

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

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

 

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

 

  1. Enable the gate, and click on Active to enable the trigger output. Set the Note and Velocity as desired (however when using noise with Mai Tai, the specific Note isn’t that critical). Adjust the Gate’s settings so that it triggers on the snare hits. Like the Pro EQ, the controls here are critical as well.
  • A short attack is usually best.
  • Keep release relatively short (e.g., 10 ms), unless you want to mix in some of the peaked/gated sound from this channel along with the Mai Tai and drums.
  • Hold times around 50 ms can help prevent false triggering. But you can also get creative with this control. If you don’t want to trigger on hits that are close together (e.g., fills), a long Hold time will trigger on the first snare of the series, but ignore subsequent ones that fall within the hold time.
  1. Insert the Mai Tai. Set its input to Gate, and enable the mixer’s Monitor button. Figure 2 shows the finished track layout.

Figure 2: Track layout for snare drum extraction.

 

TWEAKING THE MAI TAI

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

 

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

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

BUT WAIT…THERE’S MORE!

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

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

 

Friday Tips: Frequency-Selective Guitar Compression

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

 

 

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

 

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

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

 

 

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

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

Friday Tips: The Sidechained Spectrum

You’re probably aware that several Studio One audio processors offer sidechaining—Compressor, Autofilter, Gate, Expander, and Channel Strip. However, both the Spectrum Meter and the Pro EQ spectrum meter also have sidechain inputs, which can be very handy. Let’s look at Pro EQ sidechaining first.

When you enable sidechaining, you can feed another track’s output into the Pro EQ’s spectrum analyzer, while still allowing the Pro EQ to modify the track into which it’s inserted. When sidechained, the Spectrum mode switches to FFT curve (the Third Octave and Waterfall options aren’t available). The blue line indicates the level of the signal going through the Pro EQ, while the violet line represents the sidechain signal.

As a practical example of why this is useful, the screen shot shows two drum loops from different drum loop libraries that are used in the same song. The loop feeding the sidechain loop has the desired tonal qualities, so the loop going through the EQ is being matched as closely as possible to the sidechained loop (as shown by a curve that applies more high end, and a slight midrange bump).

Another example would be when overdubbing a vocal at a later session than the original vocal. The vocalist might be off-axis or further away from the mic, which would cause a slight frequency response change. Again, the Pro EQ’s spectrum meter can help point out any differences by comparing the frequency response of the original vocal to the overdub’s response.

The Spectrum Meter

Sidechaining with the Spectrum Meter provides somewhat different capabilities compared to the Pro EQ’s spectrum analyzer.

 

With sidechain enabled, the top view shows the spectrum of the track into which you’ve inserted the Spectrum Meter. The lower view shows the spectrum of the track feeding the sidechain. When sidechained, all the Spectrum Meter analysis modes are available except for Waterfall and Sonogram.

While useful for comparing individual tracks (as with the Pro EQ spectrum meter), another application is to help identify frequency ranges in a mix that sound overly prominent. Insert the Spectrum Meter in the master bus, and you’ll be able to see if a specific frequency range that sounds more prominent actually is more prominent (in the screen shot, the upper spectrum shows a bump around 600 Hz in the master bus). Now you can send individual tracks that may be causing an anomaly into the Spectrum Metre’s sidechain input to determine which one(s) are contributing the most energy in this region. In the lower part of the screen shot, the culprit turned out to be a guitar part with a wah that emphasized a particular frequency. Cutting the guitar EQ just a little bit around 600 Hz helped even out the mix’s overall sound.

Of course, the primary way to do EQ matching is by ear. However, taking advantage of Studio One’s analysis tools can help speed up the process by identifying specific areas that may need work, after which you can then do any needed tweaking based on what you hear. Although “mixing with your eyes” isn’t the best way to mix, supplementing what you hear with what you see can expedite the mixing process, and help you learn to correlate specific frequencies with what you hear—and there’s nothing wrong with that.