The basic idea is that setting a white or pink noise track to follow the chord track gives the noise a sense of pitch. Although having a long track of noise isn’t very interesting, if we gate it with a percussion part, then now we’ve layered the percussion part’s rhythm with the tonality of the noise. Add a little dotted 8th note echo, and it can sound pretty cool.
Noise needs to be recorded in a track to be affected by harmonic editing, so open up the mixer’s Input section, and insert a Tone Generator effect in tracks 1 and 2. Set the Tone Generator to Pink Noise, and trim the level so it’s not slamming up against 0 (Fig. 1).
Figure 1: We need noise in each channel to implement this technique.
Record-enable both tracks (set them to Mono channel mode), enable Input Monitor, and start recording noise into the tracks. The reason for recording into two tracks is we want to end up with stereo noise, so the tracks can’t be identical.
Now that the noise is recorded, you can remove the Tone Generator effects from the track inputs. At the mixer, pan one channel of noise left, and one right. In each track’s Inspector, choose Universal Mode for Follow Chords, and Strings for Tune mode (Fig. 2).
Figure 2: How to set up the tracks for stereo noise. The crucial Inspector settings are outlined in yellow.
Set each track’s output to a Bus, and now we have stereo noise at the Bus output. Insert a Gate in the Bus, and any other effects you want to use (I insert a Pro EQ to trim the highs and lows somewhat, and a Beat Delay for a more EDM-like vibe—but use your imagination).
Choose the percussion sound with which you want to control the Gate sidechain, insert a pre-fader send in the percussion track, assign the send to the Gate, and then adjust the Gate parameters so that the percussion track modulates the noise percussively. Fig. 3 shows the track setup.
Figure 3: Track layout used in the audio example.
Tracks 1 and 2 are the mono noise tracks that follow the Chord Track, and feed the Bus. Tracks 4 and 5 both have pre-fader sends to control the Gate, so that for the first 7 measures only the cowbell controls the gate, but at measure 8, a tambourine part also modulates the Gate.
Track 6 has the cowbell and tambourine audio, which is mixed in with the pitched noise, while the folder track has the kick, snare, and hi-hat loops. (The reason for not using post-fader sends on the percussion tracks is so that the tracks controlling the Gate are independent of the audio, which you might want to process differently.)
With a longer gate, the sound is almost like the rave organ sound that was so big at the turn of the century. And there are options other than gating, like using X-Trem…or following the Gate with X-Trem. Or draw a periodic automation level waveform for the bus, and use the Transform function to make everything even weirder. In any case, now you have a new, and unusual tool, for percussive accents.
Do you feel kind of left out because of the cool guitar amps that Studio One added in version 4.6? Well, this week’s tip is all about having fun, and bringing power chord mentality to keyboard, courtesy of those State Space amps. Listen to the audio example, and you’ll hear what I’m talking about.
And so you can get started having fun, you don’t even have to learn what’s going on to get that sound you just heard. Download Power Chordz.instrument, drag it into the track column, feed it from your favorite MIDI keyboard, and start playing.
Figure 1: The Multi-Instrument is pretty basic—it just bundles a Chorder Note FX and Mai Tai together.
The preset starts with a Multi-Instrument (Fig. 1) that consists of the Chorder Note FX, and Mai Tai synthesizer. The Chorder plays tonic, fifth above, an octave above, octave+fifth above, and two octaves above when you hit a keyboard key—your basic “it’s not major, and it’s not minor” type of power chord.
The Mai Tai uses a super-simple variation on the Init preset. In Fig. 2, anything that’s not relevant is grayed out. Turn off Osc 2, Noise, LFO 1, and LFO 2. There’s no modulation other than pitch bend, and no FX. Envelope 2 and Envelope 3 aren’t used. I set Pitch Bend to 7 semitones to do whammy bar effects, but adjust to taste. Also, you might want to play around with the Quality parameter. I’m allergic to anything called “normal,” so if you are as well, try the 80s, High, and Supreme settings to see if you like one of those better.
Figure 2: The Mai Tai preset uses simple waveforms, which is what you want when feeding amp sims and other distortion-oriented plug-ins.
Look in the instrument’s mixer channel, and you’ll see four Insert effects: Pro EQ, Ampire, Open Air, and Binaural Pan. You can check out their settings by opening them up, but the Ampire settings (Fig. 3) deserve a bit of explanation.
Figure 3: Ampire is using the Dual Amplifier and 4×12 MFB speaker cabinet, but just about any amp and cab has their merits.
The reason for choosing the Dual Amplifier is because it’s really three amps in one, as selected by the Channel knob on the right—I figured you’d appreciate having three separate sounds without having to do anything other than adjust one knob. Try different cabs and amps, but be forewarned—you can really go down an Endless Rabbit Hole of Tone, because there are a lot of great amp and cab sounds in there. I’ll admit that I ended up playing with various permutations and combinations of amps, effects, and cabs for hours.
You can also get creative with the Mai Tai, specifically, the Character controls. I didn’t assign any controls to a Control Panel, or set up modulation because having a pseudo-”whammy” bar pitch wheel was enough to keep me occupied. But, please feel free to come up with your own variations. And of course…post your best stuff on the PreSonus Exchange!
Many people don’t realize there are two types of expansion. Downward expansion is a popular choice for minimizing low-level noise like hiss and hum. It’s the opposite of a compressor: compression progressively reduces the output level above a certain threshold, while a downward expander progressively reduces the output level below a certain threshold. For example, with 2:1 compression, a 2 dB input level increase above the threshold yields a 1 dB increase at the output. With 1:2 expansion, a 1 dB input level decrease below the threshold yields 2 dB of attenuation at the output.
Upward expansion doesn’t alter the signal’s linearity below the threshold—if the input changes by 2 dB, the output changes by 2 dB. But above the threshold, levels increase. For example, with 1:2 expansion, a 1 dB increase above the threshold becomes 2 dB of increase at the output. Fig. 1 shows the difference between downward and upward expansion.
That’s Nice…So What?
Upward expansion is a useful tool for drums, hand percussion, and other percussive instruments. One function is transient shaping, to emphasize attacks. Suppose you have a drum loop with too much room sound. Traditional expansion can make the room sound decay faster, but using upward expansion brings the peaks above the room sound, while leaving the characteristic room sound alone.
Another use is with percussion parts, like hand percussion, that are playing along with drums. A lot of times you don’t want the percussion hits to be too uniform in level, but instead, the most important hits should be a little louder compared to the rest of the part. Again, that’s where upward expansion shines. Dip the threshold just a tiny bit below the peaks—the peaks will stand out, and sound more dynamic.
Let’s listen to an audio example. The first two measures use no upward expansion with a drum track. The next two measures add a subtle amount of upward expansion. You’ll hear that the peaks from the kick and snare are still prominent, but the room sound and cymbals are a bit lower by comparison. The final two measures use the settings shown in Fig. 2. The kick and snare peaks are still there, but the rest of the part is more subdued, and the overall sound is “tighter,” with more dynamics.
The only difference among the two-measure sections is the Range control setting. For the first two measures, it’s 0.00 dB; nothing can be above the threshold, because there is no threshold. In the second two measures, the Range is -2.00, so anything above that threshold goes through 1:4 expansion. In the final two measures, the range is -4.00 (I rarely take it lower, as long as the Event hits close to 0 on peaks).
Here’s the coolest part: Automating the Range parameter lets you alter a drum part’s dynamics and feel, without having to change the part itself. This is particular wonderful for compressed drum loops, because you can lower the range to keep the peaks, while making the rest of the loop less prominent. When you want a big sound, slam the Range back up to 0.00.
But Wait! There’s More!
The Multiband Dynamics processor can do frequency-selective upward expansion. You can isolate just the high frequencies where a drum stick hits, and emphasize only that frequency. Another use is making acoustic guitars sound more percussive, as in this audio example.
The first two measures are the original acoustic guitar track, and the next two use Multiband Dynamics to accent the strums (Fig. 3).
The Multiband Dynamics are in a separate, parallel track (you could build this into an FX Chain, but I think showing this in two channels illustrates the process better). Because the Multiband Dynamics is listening to only the high frequencies, which are quite weak and not sufficient to go over the expander threshold, the Input control is adding +10 dB of gain. Alternately, you could insert a Mixtool before the Multiband Dynamics.
This effect is best when used subtly, but next time you want to reach for a transient shaper, try this instead. It’s a flexible way to emphasize percussive hits and strums.
I think the Autofilter is a great effect—which you probably already figured out if you saw my blog posts The Best Flanger Plug-In?, Attack that Autofilter, and Studio One’s Secret Equalizer. But the one effect that has always eluded me was the Autofilter effect itself, when used with guitar or bass. It never seemed to cover quite the right range—like it wouldn’t go high enough if I hit the strings hard, but if I compensated for that by turning up the filter cutoff then it wouldn’t go low enough. Furthermore, the responsiveness varied dramatically depending whether I was playing high up on the neck, or hitting low notes on the E and A strings. So basically, I’ve never really used the Autofilter for its intended purpose—until now, because I’ve finally figured out the recipe. Hey, better late than never!
This technique involves dedicating two tracks to the same guitar audio—the Autofilter processes one of the tracks, while the other track provides a pre-fader send to the Autofilter’s sidechain (Fig. 1). By processing the send, we can make the Autofilter respond pretty much any way we want.
Figure 1: Both tracks are being fed from the guitar audio. The track on the right processes the audio to control the sidechain of an Autofilter, which is inserted in the track on the left.
The Autofilter (Fig. 2) has a lower filter cutoff than what I would normally use, were it not for this technique; the envelope amount slider is up all the way (the LFO is at zero, so it doesn’t influence the envelope effect).
Figure 2: Initial Autofilter settings, when controlled by a processed sidechain signal.
As to what’s conditioning the send to make the Autofilter happy, it’s the underappreciated Channel Strip plug-in (Fig. 3). The strip is both compressing and expanding because, well, that’s what ended up sounding right. But the key here is also the EQ. The higher-output low strings are attenuated, so that the filter response for the lower strings is consistent with the upper strings—thanks to the massive high-frequency boost. Meanwhile, the Gain is slammed all the up, so that it drives the Autofilter to a suitably high frequency with strong input signals.
Figure 3: The Channel Strip is ideal for conditioning the signal controlling the Autofilter sidechain.
Here’s another tip: The technique of duplicating a track, and processing it to provide a custom sidechain signal, has a lot more uses than just this. Try using the X Trem as a step sequencer and control the sidechain in a compressor…or the Autofilter’s sidechain, for that matter.
Remember, if you want to come up with something novel, ask “what if?”—not so much “how to?” I guarantee you won’t find a single, click-bait YouTube video called “SECRET AUTOFILTER PRO TRICK YOU MUST KNOW!” I’d never claim this is a tip the pros use; the only reason I came up with it is because I was frustrated that I couldn’t get the Autofilter to do what I wanted, and thought “What if I process the signal going to sidechain?” I can’t help but wonder how many other “what ifs” are waiting to be discovered…well, see you next week!
There are many uses for alternate mixes, and Studio One makes it not only easy to create alternate mixes, but also to store them as part of a song. We’ll cover traditional uses for alternate mixes, and then get into some more unusual applications.
How to Create and Store Alternate Mixes
When sending a file off for mastering, sometimes the engineer will want two additional versions, one with the vocal up 1 dB and another with it down 1 dB. This is because during the course of mastering, the relationship of the vocal to the track might change. A more prosaic example is creating a mix without vocals for karaoke, and of course, remixes are common in EDM. You might also do an “unplugged” version with only the acoustic tracks.
After creating the alternate mix:
If your mix changes are relatively extensive, save it as a Version so you can recall it later. Choose File > Save New Version, and name it (Fig. 2). Use a name that corresponds to the name of the alternate mix file, and then you can choose File > Restore Version to recall the version corresponding to that particular mix.
Processing Song Sections with Alternate Mixes
Suppose you want to add flanging to only a solo section of a song. You can insert a flanger in the main bus, and then adjust automation to enable it and set the parameters as desired. However, another option is to create a mix, and bring it into your song as a track. Then, create an alternate mix you bring into the song. Now you can experiment on the alternate mix with the effect(s) you want to use, and when you have the sound you want, render it. Cut the solo section you want to flange from the main mix, and insert the rendered solo section from the alternate mix.
Better Virtual Instrument and Amp Sim Feel
Last week’s tip covered how to save CPU power with amp sims by bouncing and/or transforming the track with the sim. One user commented “I can see how doing [this] reduces CPU load, but only after I’ve finished choosing and using an amp sim patch. My frustration is the latency or CPU hit when actually playing my guitar through an amp sim, deciding on what I’m going to play, rehearsing it, auditioning amp sim patches and so on.” Alternate mixes to the rescue: make a premix of all tracks except the one with the amp sim, and then disable the tracks themselves. With the tracks now placing no load on the CPU because you’re listening only to the premix, you can throttle the latency way down when playing around with your sim and adding a new part. After recording the part, you can use the tips presented last week to reduce the CPU drawn by the amp sim.
Auditioning Different Mixes for Albums
The renewed interest in vinyl has had a corollary effect: an interest in albums and collections of songs, not just singles. As a result, Studio One’s synergy between the Song and Project page—where you can edit songs after hearing them in context with the master file, and update the master file with that song’s changes—is brilliant. But even if you don’t produce albums, Studio One’s mastering options make it easy to obtain consistency among songs. That way, if someone switches from one song of yours to another on YouTube or Spotify, there won’t be a jarring difference.
But you might do various mixes of songs, so you can choose the best one when assembling an album. If you update the version in the mastering file after changing a mix, to compare it to a previous mix you need to open the file with the other mix (which may take a fair amount of time if you have lots of effects and virtual instruments), update the mastering version, and repeat with any other mixes.
A simpler option is to create alternate mixes in your song as described above (remember to save each alternate mix as a separate version), and mute all tracks in the song except for the track with the mix you want to audition. Update the mastering file with that track. To audition a different mix, mute all tracks except for the different mix, update the mastering file, and hear that mix in context. Once you decide which mix you like best, open the Version containing that mix, and then you can make further tweaks to it.
There’s an old joke about guitarists:
“How many guitars does a guitar player need?”
“Just one more!”
…and sometimes I feel the same way about amp sims, because all of them are different. Ampire XT benefits from PreSonus’s “State Space” technology, and if you have no other amp sims, its collection of amps has pretty much all the essentials.
What’s more, you can load thrid-party cabinet impulse responses (IRs) that re-create the sound of various cabinets, mics, and mic positions. These go into the User Cabinet, whose unique feature compared to typical IR loaders is being able to load individual IRs for the three mics.
But you can take impulses even further by turning off an amp’s cabinet altogether, and following Ampire XT with the Open Air convolution processor. Although most people probably think of Open Air as a way to create a variety of reverb and other space-based effects, it’s also a flexible impulse response loader that plays nice with cabinet impulses.
There are many free cabinet impulses on the web to get you started. Admittedly, the sound quality varies—some are fine, some aren’t, but there’s also a middle ground where tweaking the Open Air controls can give the sound you want. http://cabs.kalthallen.de is a popular source for free impulses (click on the Free tab), but there are many other companies that offer free samples, or sell impulses commercially.
Create an FX Chain with Ampire XT followed by Open Air. The Impulse Responses are only for cabinets, so set up Ampire XT’s amp and effects however you want, but turn off the cabinet section (Fig. 1).
Figure 1: Click the cabinet bypass button (middle left, outlined in white) and the cabinet field will show None (upper right, outlined in white).
Follow Ampire XT with the Open Air, and start with its Default preset. Drag an impulse into the Open Air waveform display window (or click on the impulse name field to open the file selector, and then navigate to the impulse you want). Turn Mix to 100% so that you hear only the cabinet output, and none of the pre-cabinet amp sound (Fig. 2).
Figure 2: Make sure you set the Open Air Mix control to 100%, so that you don’t hear the pre-cabinet amp sound.
Tweaking the Tone
The Kalthallen impulse shown in the screenshot above didn’t need tweaking to sound good, but you’ll find that with a lot of the free impulse responses, you’ll need to tweak the Gain and Frequency controls. Often the main problem is a “thin” sound and Fig. 3 shows some tweaks that help remedy this issue—pull back on the highs, and boost the low end for a bigger, beefier tone.
Figure 3: These EQ settings can help tame free impulse responses that sound too thin.
But the most dramatic tweaks come by enabling Shorten with Stretch and Stretch with Pitch, then varying the Length control. This can produce sounds that are similar to different mikings, or even cab sounds you’ve never heard before. The Predelay, ER/LR-Xover, and ER/LR controls can also affect the sound, although the differences aren’t as dramatic as stretching with the Length control.
Finally, although it’s great to have options, you don’t want to suffer from option overload (“maybe trying just one more impulse will give the sound I want…”). If you download a bunch of impulses, create a folder of favorites in a place where it’s easy to open it up, and drag-and-drop impulses into Open Air. If you find one you really like, save it as an Open Air preset for future use.
After doing some fairly “normal” tips for the last few Fridays, let’s go a little crazy—and explore some interesting sound design and rhythmic possibilities.
Open Air is a wonderful convolution processor, but it’s helpful to remember it can load any audio file, not just room and reverb impulses. I’ve said many times it’s more fun to ask “what if?” than “how do I?”, because “what if” is all about experimentation. So I asked “What if I’m using a drum loop, and also load that same loop into Open Air as an impulse?” You might not use the resulting sound all the time, but give this technique a try—you’ll hear an entirely new type of percussive effect.
Figure 1: Typical Open Air settings when modulating a drum loop by itself.
Remember that the drum loop is still acting like a reverb, so it will build up a bit over time until the level stabilizes, and the processed sound will have a tail as long as the loop.
Next, there are several ways to add variations. First, you don’t have to convolve a loop with itself—check out the audio example.
The first four measures are a drum loop convolved with itself. The second four measures convolve the original drum loop with a tom loop, while the final four measures convolve the original drum loop with a percussion loop.
Altering the Open Air Length can create interesting effects, especially when using a rhythmically related length—like half or 1/3 the length. With sparse loops, longer lengths can work too, like 1.33, 1.5, or 1.66 the length (get out your calculator, and work with the number that’s shown under the Length control). Additional EQ and processing can add even more interest.
And remember to experiment with other types of impulse as well—pads, voices, guitar chords, whatever! You never know what you’ll discover.
You’ll often see this kind of comment in forums: “There must be something wrong with Studio One! I can run only a couple amp sim instances before the program can’t handle any more!” But you’ll also see this comment about other DAWs—because the “problem” isn’t the DAW, it’s the amp sims and current computer technology. Fortunately, Studio One has anticipated these issues, and offers three effective solutions.
Remember, an amp sim is processing a dry guitar track in real time—not playing back processed audio. Amp sim sound quality has improved dramatically over the past few years, but the trade-off is the CPU power needed to do the serious number-crunching required for realistic amp sounds. Studio One’s CPU-saving options are great with virtual instruments, which can sometimes suck even more power than amp sims—but guitar players who are discovering the fun of amp sims need to know about these options, too.
The Old-School Fix
Although some people recommend the general-purpose, old-school fix of increasing latency to reduce stress on your CPU, that makes playing guitar much less fun. Another solution is to buy a much faster computer. Studio One’s solutions work at lower latencies, as well as older, slower computers.
Solution 1: Bounce to New Track
Select one or more Events. Right-click on any of them, and choose Event > Bounce to New Track (Fig. 1). This creates a new audio track that incorporates the sound created by the original track’s processing, but without any inserted plug-ins—the sound is “baked into” the new audio track. Audio tracks require far less CPU power than a track whose effects are being created in real time. Bouncing leaves the original track in place but mutes it, so you can unmute it to return to the original track’s audio and effects if needed.
Figure 1: If you use Bounce to Track (outlined in white) as much as I do, it will show up in the Recent Items section of your right-click context menu.
To conserve the CPU used by the original track’s effect(s), either turn off power to the effect(s), or right-click on the original track in the track column and choose Disable Track. To return the track to its initial status, right-click on the track in the Track column, and choose Enable Track.
Note that when signing off on a project, this is also an excellent way to “future-proof” the project against future operating system (or other) changes that may render a plug-in unusable. If the sound has been preserved as an audio file, you’ll at least be able to open the processed sound.
Solution 2: Transform to Rendered Audio
Right-click on the track in the Track column, or choose Track > Transform, and then choose Transform to Rendered Audio. This renders the effect sound so that it becomes part of the existing audio track. Unlike bouncing, this operation doesn’t create a new track, and it automatically disconnects the effect from the CPU to save power.
When you choose Transform to Rendered Audio, a dialog box appears with two options (Fig. 2).
Figure 2: The Transform to Rendered Audio dialog box.
You can always undo if you change your mind, but Preserve Realtime State (which I highly recommend checking) preserves the original, real-time state so you can always return to the original track settings and effects. Preserve Realtime State also persists through saves and copies. To return to the original track, right-click on the track in the Track column, or choose Track > Transform, and then choose Transform to Realtime Audio.
The second dialog box option renders any effects tail, such as a long trail of echoes or delay, that extends past the length of the existing Events. You can choose Auto Tail, where Studio One detects how long the tail lasts and renders according, or specify a fixed tail of a particular length. (A fine point: Studio One fades out the Event over the tail’s duration, but it’s an editable envelope.)
Render Event FX
Event FX, as accessed through an Event’s Inspector, are invaluable. With Ampire XT (and many other amp sims), you can’t automate amp or cabinet changes—only parameters within amps and cabinets. So, if the verse’s guitar part is one Event and the chorus’s guitar part is a different Event, each can have its own amp sim sound.
Figure 3: Here, Ampire XT is an Event FX, and can be rendered to save CPU power. Note that you can also choose different amps and cabinets, see a tuner thumbnail, and turn the Stomps section on or off.
The trade-off is that more amp sims draw more CPU power. Fortunately, Event FX have a Render button (Fig. 3). Immediately upon rendering, the sound becomes part of the audio, the effect itself disconnects from the CPU, and the Render button changes to Restore. Similarly to transforming an audio track, you can revert to the original state at any time by clicking Restore.
Multiple Renders in One Operation
Suppose a track has two Events, each with their own Ampire XT inserted via an Event FX, and there’s a CPU-hungry reverb processing the entire track. If you apply Transform to Rendered Audio to the track, it will Render the Event FX and the Track effect automatically. But if you then need to make changes and transform the Track back to realtime audio, the Track and Event FX will be restored to their initial states.
Bounce to New Track with both Events selected will produce the same results in the bounced track, i.e., all the effects will be rendered. If you want to return from where you started, delete the bounced Track, and unmute the two Events in the original Track (which will still have its effects inserted).
Once you bounce or transform tracks and reclaim all that CPU power, you can continue going cRazY with amp sims—without stressing out your computer, or Studio One.
There are a lot of filter responses: notch, bandpass, peak, allpass, high pass, lowpass, shelf…now let’s add the “table” response to the collection.
Parametric EQs can add peaks and cuts that are broad, narrow, or anywhere in between, but they all have slopes on either side of the filter frequency. The table response described here can boost or cut over a range of frequencies, with a flat response over that range. This avoids having to dedicate several overlapping parametric stages, which still doesn’t achieve quite the same result. The key to this response is combining shelving EQs.
Table Response Boost
To boost a frequency range, set the low- and high-shelf frequencies to the lowest and highest frequencies in the range (Fig. 1). Use the Shelf setting to determine how quickly the boosted section returns to the flat response. I’ve found the 12 and 24 dB settings works well, because the Q control comes into play. This can provide additional modifications to the response, which we’ll cover later. However, for the gentlest effect, 6 dB is valid in many situations as well.
Figure 1: This table response, inserted before a high-distortion amp sim, gives greater sensitivity to midrange notes and also trims the highs and lows for a “tighter” sound.
But we’re not done quite yet. To provide an actual boost, increase the output Gain control for the desired amount of boost. For example, if you want the table response to boost +12 dB, set the high and low shelf Gain settings to -12 dB, and the output Gain control to +12.
Table Response Cut
Similarly to the boost option, set the low- and high-shelf frequencies to the lowest and highest frequencies in the range you want to cut (Fig. 2). Again, use the Shelf setting to determine how quickly the boosted section returns to the flat response; the same general comments about how the shelf slope works with boosting apply here too.
Figure 2: This table response for a drum loop cuts back on the midrange a bit to help emphasize the kick and the snap/sizzle of the share and high-hats; it also reduces any “midrange mud,” and makes space in that frequency range for other instruments.
Cutting requires an equal and opposite approach to what we did for boosting. If you want the table response to cut 4 dB, then boost the shelf controls by +4 dB. Then, set the output gain control to -4 dB. This restores the shelf boosts to flat, and adds the desired amount of cut for the specified frequency range.
When cutting with the low shelf or boosting with the high shelf, increasing resonance by turning up the Q control adds a peak just above the shelf’s corner frequency, and a dip below the corner frequency. When boosting with the low shelf or cutting with the high shelf, increasing Q adds a peak just below the shelf’s corner frequency, and a dip above the corner frequency. This emphasizes the extremes of the chosen frequency range, while also increasing the depth of the cuts near the corner frequency. Try adding resonance to the low shelf when using this technique for vocals, particularly narration (Fig. 3).
Figure 3: The table response adds a bit of a low-frequency boost (with Q) to give the “late night FM DJ sound,” but also cuts lower frequencies to reduce p-popping. Meanwhile, the high-frequency shelf emphasizes the voice’s articulation, while reducing extraneous highs, hiss, and sibilance.
Of course, the table response doesn’t replace a parametric. But sometimes, it might be just the response you need, and you’ll find it faster to dial in the right frequency range by moving the shelf controls than trying to make multiple stages of peak/boost EQ do what you want.
A physical guitar amp is more than a box with a speaker—it’s a box with a speaker being picked up by a mic in a room. Both the mic and room contribute to the overall sound. To better emulate the sound of a physical guitar amp, Ampire includes a Mic Edit Controls panel that allows making a variety of virtual mic adjustments.
Ampire doesn’t include room emulation, because you can emulate room sound with several of Studio One’s plug-ins—Room Reverb, Open Air Reverb, Mixverb, and Analog Delay. However, it’s best to avoid adding ambiance until most other tracks have been cut, so that the ambiance achieves the right balance. Too much ambiance can clutter the mix, or hog the stereo field.
The mics you choose, their levels with respect to each other, and whether you add delay can make a major difference in your amp’s sound. So, let’s investigate the Mic Edit Controls panel (Fig. 1).
Choosing the Mic Type
Many guitarists record with their amp cranked to really high levels, to get their “sound.” Dynamic mics are ideal because they can handle high levels, and the inexpensive Shure SM57 is the classic guitar cabinet mic—many engineers choose it even when cost is no object. Although dynamic mics may lack brightness compared to condenser mics (as modeled by Mic C), this doesn’t matter much with amp cabinets, which typically don’t have much energy above 5 kHz or so anyway. Mic A in the Mic Edit Controls panel has the SM57’s sonic character, and will likely be your go-to mic.
Mic B produces the sound associated with ribbon mics, which shows one of Ampire’s benefits: older ribbon mics tended to be fragile—but you can’t blow up a virtual mic. Ribbon mics have an inherently warm midrange. Royer’s R-121 mic is popular for miking cabs, and Mic B models its overall sonic character.
Mic C emulates the PreSonus PM-2 matched pair of condenser microphones. Condenser mics are often too sensitive for close-miking loud amps, but when moved a bit back from the cab, they can give a brighter, more “open” response that handles note attacks well. They’re also commonly used as room mics, which is why these two virtual mics are arranged in an X-Y miking configuration to give a stereo image.
Wait a Minute—Did You Say Stereo?
Guitars are mono signal sources, but taking full advantage of Ampire’s mics, as well as room ambiance plug-ins, requires a stereo signal. To convert the mono guitar into a dual mono signal (i.e., stereo, but with the same audio in the left and right channels), record the guitar with the Channel Mode set to Mono (one circle showing to the right of the Record Input selector). Although this means that any plug-ins will be in mono, that’s acceptable when tracking. After recording the track, change the Channel Mode to stereo (i.e., two circles showing to the right of the Record Input selector), select the event, and bounce it to itself (ctrl+B). Now the mono guitar is dual mono.
Mic Control Applications
Each mic has three controls: level, mute button (which makes it easy to evaluate what a particular mic contributes to the overall sound), and phase switch (the Ø button). Also, Mics B and C have Delay controls.
Often when miking a physical amp with more than one mic, you’ll vary their blend to find the right mix. The Mic Mix Link button toward the extreme left simplifies this process. When enabled, altering one mic’s level adjusts the levels of the other mics oppositely. For example, turning up Mic A turns down Mics B and C, or turning up Mic B turns down Mics A and C.
The Phase buttons and Delay controls can make major differences in the overall sound. There’s no right or wrong phase or delay setting; use whatever sounds best to you. Try the following to hear how these controls affect the sound. (Bear in mind that amp sims do a lot of calculations, so moving the controls will sound “choppy.” This is because Ampire has to recalculate constantly to reflect the changing settings.)
Now check out how Mic C creates a stereo spread. With Mic Mix Link off, adjust Mic A and/or Mic B for the desired sound. Bring up Mic C’s Level control slowly, and you’ll hear the stereo image bloom. Again, the Delay control and Phase reverse button make a big difference in the sound.
Clean Sounds, Too
One of my favorite mic applications is with clean guitar sounds (cabinet only, no amp). Mic C is particularly useful, because its brightness gives the cabinet’s tone a useful lift, and creates a stereo image. Finally, note that if you change the Channel Mode from mono to stereo (or the reverse), the sound may mute. Varying one of the Mic level controls restores the sound. Of course, it’s easy enough to call up an Ampire preset, and just start playing… but becoming proficient with the Mic Edit Controls opens up a wealth of possibilities.