For one month only, take advantage of the full force of Studio One Artist with the Artist Booster Pack for 30% OFF!
This bundle includes five of our most popular add-ons for Studio One Artist:
VST and AU and Rewire Support
Studio One Remote Support
Acoustic Drum Loops Vol. 2
MP3 Converter Support
Ampire XT Metal Pack
A sampled drum sound can get pretty boring. There’s even a name for triggering the same sound repeatedly—“the machine gun effect.” Sometimes you want this, but often, it’s preferable to have a sound that responds to velocity and is more expressive.
There are two ways to address this with Impact XT, depending on whether you have multiple samples recorded at different intensities (i.e., softer and harder hits), or only one sample, which then means you have to “fake” it sounds like it was recorded with different intensities.
Multiple Drum Samples
This is the most common way to create expressive drum parts. Drum sample libraries often include multiple versions of the same drum sound—like soft, medium, and hard hits. The technique we’ll describe here works for more than three samples, but limiting it to three is helpful for the sake of illustration.
Impact XT makes it super-simple to take advantage of sounds recorded at different intensities because you can load multiple samples on a single pad. However, note that if a pad already contains a sample and you drag a new sample to a pad, it will replace, not supplement, the existing sample. So, you need to use a different approach.
Figure 1: Click on the + sign to load another sample on to a pad.
Figure 2: The splitter bar between samples can alter the velocity range to which a sample responds.
Now you’ll trigger different drum samples, depending on the velocity.
How to Fake Multiple Drum Samples
If you have a single drum sample with a hard hit, then you can use Impact XT’s sample start parameter to fake softer hits by changing the sample start time. (Starting sample playback later in the sample cuts off part of the attack, which sounds like a drum that’s hit softer.)
Figure 3: Click on the sample start line, and drag right to start sample playback past the initial attack. The readout toward the lower right shows the amount of offset, in samples.
Play the drum at different velocities. Tweak sample start times, and/or velocities, to obtain a smooth change from lower to higher velocities.
But Wait…There’s More!
Let’s add two more elements to emphasize the dynamics. These parameters affect all samples loaded on the pad, and are also effective with pads that have only a single sample.
Figure 4: Assigning velocity to Pitch and Filter Cutoff can enhance dynamics even further.
At the Pitch module, turn up the Velocity to Pitch parameter by around 0.26 semitones (Fig. 4). This raises the pitch slightly when you hit the drum harder, which emulates acoustic drums (the initial strike raises the tension on the head, which increases pitch slightly, depending on how hard you hit the drum).
Similarly, back off on the Filter Cutoff slightly, and turn up the Filter’s Vel parameter a little bit (e.g., 10%). This will make the sound brighter with higher velocities.
Done! Now go forth, and give your music more expressive drum sounds.
I sometimes record acoustic rhythm guitars with one mic for two main reasons: no issues with phase cancellations among multiple mics, and faster setup time. Besides, rhythm guitar parts often sit in the background, so some ambiance with electronic delay and reverb can give a somewhat bigger sound. However, on an album project with the late classical guitarist Linda Cohen, the solo guitar needed to be upfront, and the lack of a stereo image due to using a single mic was problematic.
Rather than experiment with multiple mics and deal with phase issues, I decided to go for the most accurate sound possible from one high-quality, condenser mic. This was successful, in the sense that moving from the control room to the studio sounded virtually identical; but the sound lacked realism. Thinking about what you hear when sitting close to a classical guitar provided clues on how to obtain the desired sound.
If you’re facing a guitarist, your right ear picks up on some of the finger squeaks and string noise from the guitarist’s fretting hand. Meanwhile, your left ear picks up some of the body’s “bass boom.” Although not as directional as the high-frequency finger noise, it still shifts the lower part of the frequency spectrum somewhat to the left. Meanwhile, the main guitar sound fills the room, providing the acoustic equivalent of a center channel.
Sending the guitar track into two additional buses solved the imaging problem by giving one bus a drastic treble cut and panning it somewhat left. The other bus had a drastic bass cut and was panned toward the right (Fig. 1).
Figure 1: The main track (toward the left) splits into three pre-fader buses, each with its own EQ.
One send goes to bus 1. The EQ is set to around 400 Hz (but also try lower frequencies), with a 24 dB/octave slope to focus on the guitar body’s “boom.” Another send goes to bus 2, which emphasizes finger noises and high frequencies. Its EQ has a highpass filter response with a 24dB/octave slope and frequency around 1 kHz. Pan bus 1 toward the left and bus 2 toward the right, because if you’re facing a guitarist the body boom will be toward the listener’s left, and the finger and neck noises will be toward the listener’s right.
The send to bus 3 goes to the main guitar sound bus. Offset its highpass and lowpass filters a little more than an octave from the other two buses, e.g., 160 Hz for the highpass and 2.4 kHz for the lowpass (Fig. 2). This isn’t “technically correct,” but I felt it gave the best sound.
Figure 2: The top curve trims the response of the main guitar sound, the middle curve isolates the high frequencies, and the lower curve isolates the low frequencies. EQ controls that aren’t relevant are grayed out.
Monitor the first two buses, and set a good balance of the low and high frequencies. Then bring up the third send’s level, with its pan centered. The result should be a big guitar sound with a stereo image, but we’re not done quite yet.
The balance of the three tracks is crucial to obtaining the most realistic sound, as are the EQ frequencies. Experiment with the EQ settings, and consider reducing the frequency range of the bus with the main guitar sound. If the image is too wide, pan the low and high-frequency buses more to center. It helps to monitor the output in mono as well as stereo for a reality check.
Once you nail the right settings, you may be taken aback to hear the sound of a stereo acoustic guitar with no phase issues. The sound is stronger, more consistent, and the stereo image is rock-solid.
We recently had the opportunity to hear from Rich Mahan who is a guitarist, singer-songwriter, podcaster, and a PreSonus user! If you’re not familiar with Rhino, we’re excited to introduce you to it. It is important and very well-respected reissue label, the label home of Warner Music’s legendary catalog. Currently living in Nashville, Rich where he records his podcast titled the Rhino Podcast. This biweekly podcast dives into classic artist and albums, interviews with musicians and lots of behind the scenes stories about some of the most legendary music. Their latest episode discusses Prince and it’s very entertaining! Read all about his thoughts on the growing a creating a podcast, the industry, gear and the Quantum 2!
Tell us about your background. How long have you been in the audio industry?
I started recording in the mid 90’s on a Vestax 4 track cassette recorder, moved up to a Tascam 388 reel to reel, and then to computer-based recording around 2003, starting originally using Vegas. I’ve been working in Pro Tools now for about 15 years.
How has the Audio industry changed since your early days?
The gear keeps getting better and better. The quality you can capture in a home studio or out in the field is unreal, and digital editing has changed everything. You can repair audio problems now that you simply couldn’t before, problems that would necessitate re-recording a part.
I love Cocaine & Rhinestones, by Tyler Mahan Coe (No relation). It’s about 20th-century country music, it’s really well researched and produced, and I’ve learned a ton listening to it.
Tell us about your podcast. Where did the idea for your podcast come from? How does your first podcast compare to your most recent?
The idea for the Rhino Podcast came from both Rhino and me and my co-host Dennis Scheyer. We pitched the idea to them and they said, “We’ve been wanting to do a podcast…” so it came together pretty easily. The format since the 1st episode has changed here and there, but basically, it’s still the same. Every once in awhile Rhino will want to add or take something out, so it is a living, breathing thing that progresses as it goes.
There are so many podcasts these days. How do you stand out?
There are a number of reasons why the Rhino Podcast stands out. First and foremost, we are fortunate to be working with the greatest musicians and artists of our time. Rhino Entertainment is the catalog arm of Warner Music Group, so we cover any classic recordings from Atlantic, Warner Brothers, Reprise, Elektra, Sire… there’s a wealth of musical riches to explore, and it’s been thrilling to interview the artists who have created the soundtracks of our lives. On the production side of things, we hold ourselves to a high standard of audio quality; we fight hard to avoid using telephone audio for production purposes. If we can’t interview an artist in person, then we get them into a studio or send a recordist to them to capture high-quality audio for production, and just use the phone to talk with each other, everyone wearing a microphone headset or phone earbuds so there’s no monitor bleed into the mics. I spend a lot of time removing background noise and cleaning things up, removing lip smacks, getting fades perfect, and generally being a perfectionist. I don’t let anything go. If I hear an issue, I fix it.
What advice do you have for someone who wants to start a podcast?
Really learn how to record and edit well to make a professional sounding product. Record and edit as often as you can, and practice, practice, practice. Record your friends’ bands or your own music, interview your parents or grandparents and clean up the audio by removing ums, uhs, stutters and stammers, click and pops. Learn how the room you’re recording in affects the sound of your recording. Experiment with different microphones, and buy the best gear for the job that you can afford. Garbage in, garbage out. There’s a saying that you need to get 10,000 hours of experience to really start cooking, and there’s something to that.
How did you first hear of PreSonus?
I first heard of PreSonus when I was building my first ProTools rig. I wanted better preamps than my Digi002 offered, and I scored a PreSonus Eureka channel strip on the recommendation of a friend. It was great to get a good clean preamp, compression, and EQ all in a one-rack space unit. I liked the sound of it so much that I then bought two PreSonus MP20’s for tracking drums. That improved my recordings at that time dramatically.
What PreSonus products do you use?
Currently, I’m using a Quantum 2.
Why did you decide to go with Quantum?
There are a few reasons I went with the Quantum. Firstly because it has 4 mic inputs, along with a ton of other I/O options, which I need when interviewing multiple subjects simultaneously. Another huge feature is that the Quantum has 2 Thunderbolt ports which allow me to plug in my bus-powered thunderbolt drive into the Quantum, and then the Quantum into my Macbook Pro. Thirdly, I love the smaller footprint, its’ not as wide as a piece of rack gear, so it fits easily into my messenger bag making it really easy to carry onto an airplane. And last but certainly not least, it is dead quiet, and there’s plenty of gain in the preamps to drive a Shure SM7B, which is my VO microphone of choice. The Quantum sounds awesome.
What do you like about PreSonus? What caught your eye?
The folks at PreSonus really are the best to work with. If you have an issue or need to figure something out, you can get help and get back up and running quickly. But another great thing is their gear is intuitive and easy to use. It’s easy to get great tones with their gear.
Recent projects? What’s next for you?
I just released a new album entitled, “Hot Chicken Wisdom.” I was able to put the Quantum to use while tracking parts, especially when I was traveling and wanted to have friends lay down parts away from my studio. I think if you listen to the record you’ll hear that we got some great tones, besides it’s the perfect summer soundtrack!
Next up for me is some touring to support Hot Chicken Wisdom, and I have a second Podcast in pre-production that I can’t announce quite yet, but I’m really excited about it. Anyone who wants to keep up with me can check me out at richmahan.com.
Did you know? When you buy a qualifying StudioLive digital mixer, you get the Fat Channel Plug-in Collection Vol. 1 automatically added to your my.presonus.com account. The Fat Channel Plug-in Collection Vol. 1 adds 5 EQs and 6 compressor models to your mixer—and they can also be used in the Fat Channel XT Plug-in in Studio One!
This Plug-in collection would cost $499 USD if purchased independently. However, if you’ve purchased one of the mixers listed below after July 1, 2019, all you need to do is register your mixer and the Fat Channel Plug-in Collection Vol. 1 will appear in your my.presonus.com account, ready for download and installation.
Qualifying mixers include:
All Plug-ins are state-modeled to accurately produce the sound and response of the original hardware processors. The following plug-ins are included in this bundle:
This EQ offers the world’s most popular EQ curve. Using gently sweeping treble and bass EQ shelves, it allows you to make subtle, yet effective, changes over wide swaths of the frequency spectrum.
Comp 160 Compressor
With simple controls, yet capable of extreme compression traits, the Comp 160 provides VCA character with a personality all its own. Try it on drums—you’ll be glad you did!
Everest C100A Compressor
Based on a classic design focused on gentle, natural-sounding gain reduction, the Everest C100A helps control dynamics while still letting the signal breathe.
The smooth character of this compressor allows you to create transparent or extreme color changes to your audio, making it a workhorse for just about any application.
Vintage 3-band EQ
With its distinct filter shaping, sheen, and bite, this three-band active EQ includes both high and low shelving filters, providing enhanced tone-shaping possibilities.
Tube P1B Compressor
In general, the response time of optical compressors tends to soften the attack and release, which can smooth out uneven volume fluctuations. Emulating an all-tube, optical design, the Tube P1B compressor delivers musicality, preserving the clarity of the signal even at the most extreme settings.
Tube Midrange EQ
This midrange EQ is based on a passive, all-tube design for ultra-smooth and musical equalization, making it ideal for any midrange source material.
This model of an iconic compressor/limiter of the 1950s imparts an unmistakable silky warmth on just about any signal.
Capturing the unique sound of a twin VCA gain-reduction amplifier design, the Brit Comp is ideal for taming piano dynamics or adding punch to drums and percussion.
The 1960s-vintage EQ provides consistent, repeatable equalization using three overlapping bands, divided into seven fixed frequency points, each with five steps of boost or cut. Its selectable peaking or shelving filters for the high and low band, along with an independently insertable bandpass filter, provide an easy path to creating acoustically superior equalization.
Solar 69 EQ
The sound of classic British EQ is absolutely legendary and has enhanced many a great recording. Emulating this classic British design, the Solar 69 EQ adds definition to kick drums, shapes electric guitars, and adds shimmer to acoustic guitars and vocals without sacrificing body.
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.
Mid-side (M-S) processing encodes a standard stereo track into a different type of stereo track with two separate components: the left channel contains the center of the stereo spread, or mid component, while the right channel contains the sides of the stereo spread—the difference between the original stereo file’s right and left channels (i.e., what the two channels don’t have in common). You can then process these components separately, and after processing, decode the separated components back into conventional stereo.
Is that cool, or what? It lets you get “inside the file,” sometimes to where you can almost remix a mixed stereo file. Need more kick and bass? Add some low end to the center, and it will leave the rest of the audio alone. Or bring up the level of only the sides to make the stereo image wider.
The key to M-S processing is the Mixtool plug-in, and its MS Transform button. The easiest way to get started with M-S processing is with the MS-Transform FX Chain (Fig. 1), found in the Browser’s FX Chains Mixing folder.
The upper Mixtool encodes the signal so that the left channel contains a stereo file’s center component, while the right channel contains the stereo file’s side components. This stereo signal goes to the Splitter, which separates the channels into the side and center paths. These then feed into the lower Mixtool, which decodes the M-S signal back into stereo. (The Limiter isn’t an essential part of this process, but is added for convenience.)
Even this simple implementation is useful. Turn up the post-Splitter gain slider in the Center path to boost the bass, kick, vocals, and other center components. Or, turn up the gain slider in the post-SplitterSide path to bring up the sides, for the wider stereo image we mentioned.
Fig. 2 shows a somewhat more developed FX Chain, where a Pro EQ boosts the highs on the sides. Boosting the highs adds a sense of air, which enhances the stereo image because highs are more directional.
In addition to decoding the signal back to stereo, the second Mixtool has its Output Gain control accessible to compensate for any level differences when the FX Chain is bypass/enabled. Also, you can disable the MS Decoder (last button, lower right) to prevent converting the signal back into stereo, which makes it easy to hear what’s happening in the center and sides.
And of course…you can take this concept much further. Add a second EQ in the center channel, or a compressor if you want to squash the kick/snare/vocals a bit while leaving the sides alone. Try adding reverb to the sides but not the center, to avoid muddying what’s happening in the center. Or, add some short delays to the sides to give more of a room sound….the mind boggles at the possibilities, eh?
Spoiler alert: We’ll get into some rocket science stuff here, which probably doesn’t affect your projects much anyway…so if you prefer something with a more musical vibe, come back next week. But to dispel some of the confusion regarding an oft-misunderstood concept, keep reading.
You pan a mono signal from left to right. Simple, right? Actually, no. In the center, there’s a 3 dB RMS volume buildup because the same signal is in both channels. Ideally, you want the signal’s average level—its power—to have the same perceived volume, whether the sound is panned left, right, or center. Dropping the level when centered by 3 dB RMS accomplishes this. As a result, traditional hardware mixers tapered the response as you turned a panpot to create this 3 dB dip.
However, there are other panning protocols. (Before your head explodes, please note you don’t need to learn all this stuff—it’s just to give you an idea of the complexity of pan laws, because all we really need to do is understand how things work in Studio One.) For example, some engineers preferred more of a drop in the center, so that audio panned to the sides would “pop” more due to the higher level, and open up more space in the center for vocals, kick, and bass. You could accomplish the same result by adjusting the channel level and pan, but the additional drop was sort of like having a preference you didn’t need to think about. To complicate matters further, some mixers lowered the center signal compared to the sides, while others raised the side signals compared to the center. If a DAW does the latter, when you import a normalized file and pan it hard left or hard right, it will go above 0 and clip.
But wait! There’s more. Some engineers didn’t want equal power over the panpot’s entire travel, but a slightly different curve. Others wanted a linear change that didn’t dip the signal at all.
Fortunately, Studio One has a rational approach to pan laws, namely…
THE “WHAT-THE-HECK-DO-PAN-LAWS-DO” TEST SETUP
To see how the different panning laws affect signal levels, I created a test setup (Fig. 1) with a mono track fed by the Tone Generator set to a sine wave. Two pre-fader sends went to two buses, each with a Dual Pan inserted and linked for mono. That way, one bus’s Dual Pan could be set for hard pan and the other bus’s Dual Pan for center, to compare what happens to the signal level.
Figure 1: Test setup to determine how different pan laws affect signal levels.
In all the following test result images, Track 1 shows the mono sine wave at 0 dB, Bus 1 shows the result of panning the Dual Pan full left, and Bus 2 shows the result of panning the Dual Pan to center.
Fig. 2 uses the -3dB Constant Power Sin/Cos setting for the Dual Pans. Note that the centered version in Bus 2 is 3 dB below the same signal panned full left. This is the same setting as the default for the channel panpots. However, if you collapse the output signal to mono, you’ll get a 3 dB center-channel buildup. (A fine point: setting the Main bus mode to mono affects signals leaving the main bus; the meters still show the incoming signal. To see what’s happening when you collapse the Main out to mono, you need to insert a Dual Pan in the Main bus, click on Link, and set all controls to center.)
Figure 2: -3dB Constant Power Sin/Cos pan law.
Fig. 3 uses the -6 dB linear curve. Here, the centered signal is -6 dB below the signal panned hard left. Use this curve if the signal is going to be collapsed to mono after the main bus, because it keeps the gain constant when you collapse stereo to mono by eliminating the +3 dB increase that would happen otherwise.
Figure 3: The -6 dB linear curve is often preferable if you’re mixing in stereo, but also anticipate that the final result will end up being collapsed to mono.
Fig. 4 shows the resulting signal from the 0dB Balanced Sin/Cos setting. There’s no bump or decrease compared to the centered signal, so this acts like a balance control with a constant amount of gain as you pan from left to right.
Figure 4: 0dB Balanced Sin/Cos acts like a balance control.
Sharp-eyed readers who haven’t dozed off yet may have noticed we haven’t covered two variations on the curves described so far. -3dB Constant Power Sqrt (Fig. 5; Sqrt stands for Square Root) is like the ‑3 dB Constant Power Sin/Cos, but the curve is subtly different.
Figure 5: -3dB Constant Power Sqrt bends the curve shape slightly compared to the other Constant Power curve.
In this example, the panpot is set to 75% left instead of full left. Bus 1 shows what happens with -3dB Constant Power Sin/Cos, while Bus 2 is the Sqrt version. The Sqrt version is in less of a hurry to attenuate the right channel as you pan toward the left. Some engineers feel this more closely the situation in a space that’s not acoustically treated, so there’s a natural acoustic center buildup.
Finally, Fig. 6 compares the 0 dB Balance variations, sin/cos and linear.
Figure 6: Comparing the two 0 dB Balance pan law options.
The difference is similar to the Constant Power examples, in that the basic idea is the same, but again, the linear version doesn’t attenuate the right channel as rapidly when you pan left.
I HAVEN’T FALLEN ASLEEP YET, SO PLEASE, JUST TELL ME WHAT I SHOULD USE!
The bottom line is when using the channel panpot with a mono track, if you live in a stereo mixdown world the above is mostly of academic interest. But if you’re mixing in stereo and know that your mix will be collapsed to mono (e.g., for broadcast), consider using the Dual Pan in mono channels, and set it to the -6 dB Linear pan law.
For stereo audio, again, a channel panpot works as it should—it acts like a balance control. However if the output is going to be collapsed into mono, you might want to leave the channel panpot centered, and insert a Dual Pan control to do the panning. It should be set to -6 dB Linear, controls unlinked, and then you move both controls equally to pan (e.g., if you want the sound slightly right of center, set both the left and right panpots to 66% right). Now when you pan, the mono levels in the main bus will be constant.
ONE MORE TAKEAWAY…
And finally…I’m sure you’ve seen people on the net who swear that DAW “A” sounds better than DAW “B” because they exported the tracks from one DAW, brought them into a second DAW, set the channel faders and panpots the same, and then were shocked that the two DAWs didn’t sound identical. And for further proof, they note that after mixing down the outputs and doing a null test, the outputs didn’t null. Well, maybe that proves that DAWs are different…but maybe what it really proves is that different programs default to different pan laws, so of course, there are bound to be differences in the mixes.
This offer ends July 31 – act fast!
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MVP Loops provide some of our most popular, top-selling loops and their whole collection is half off right now! Some of MVPs greatest hits include: