Your productions tend to be eclectic, with everything from vintage jazz to soul to hip-hop to alternative rock showing up in your work.

I’ve always done more than just hip-hop. I’m from Boston — between Dorchester and Brookline — so as much as hip-hop was an influence at the time, rock and punk were huge influences on me as well, and I knew as much about them as I did about soul and hip‑hop.

What gear did you start on?

When I started creating beats and sounds, I started on things like the Ensoniq Mirage, Roland S-Series keyboards like the S-50, and an AKAI S900 sampler. All this was well before the MPC was even available, though I would graduate to an MPC-60 eventually.

But at that time, I’d use whatever was available; the school studio at Hampton University, for example, had a Kurzweil K2000S. But I really developed my ability to put tracks together on a rackmount Tascam 134B 4-track Syncaset that belonged to a bass player friend of mine.

In your recent work with A$AP Rocky, as with Wyclef Jean back in the day, you incorporate a nice blend of live instruments and sampled beats.

I suppose that if I’m known for anything, it’s the combination of samples and live instruments. A lot of my commercial work has that combination because that’s really what people come to me for.

When I came into the game, DJ Premier of Gang Starr was a huge inspiration. I was enamored with how he manipulated samples, because when I came up, it was the time when people used to just loop samples, and Premier was one of the first to go to the next level of chopping up samples, and processing them in more experimental ways.

Because I play instruments, I could incorporate them along with this new way of creating beats from samples. It enabled me to be a lot more free and experimental with the productions.

Though it’s easy to see samples as simply pre-recorded sounds and instruments, they’re almost a different medium entirely. What do they bring to the table in a production?

Samples are great because they have interesting textures. Something that was recorded in a dusty room in Memphis in 1966 is going to have a very unique texture and flavor to it — and that’s something that’s very hard to create with live instruments.

What are some ways you add texture to a sample?

Guitar pedals — I carry around at least 50 of them. You can use them to distort and disrupt clean-sounding signals in unexpected ways. In fact, I especially like pedals from the ’60s and ’70s, like an old Maestro Fuzz, for instance, for that very reason. They add so much character and texture to live sound, and get them much closer to the vibe you get from a really great vintage sample.

Are there dangers to the wide availability of commercial samples and beats, perhaps a certain sameness creeping in?

Sure. Someone recently commented that 95% of today’s beats sound the same, and I think that’s basically true. Don’t get me wrong; I’m someone who’s very aware of the evolution of music technology, and I care about what the kids are into, because as far as I’m concerned, everything begins with the present generation of kids, just like it did in my day.

Not all hip-hop is going in that direction however.

No. I mean, you think of an album like Kanye’s Yeezus and Life of Pablo — they are very different than the majority of rap records. I recently read a quote with 50-Cent talking about Jay-Z’s last album, 4:44, and his “criticism” was really more of a compliment. He said it was too grown-up, too mature. I thought, the man’s a 47-year-old rapper! He should be making mature albums. You could say the same thing about a Kendrick Lamar album — the incredibly high level of intelligence and sophistication — or J Cole’s new album,KOD, that everyone’s talking about.

Let’s talk about the Apollo Artist Session you did with the Preservation Hall Jazz Band in New Orleans. What were the particular challenges of recording in that space, completely live?

The biggest problem was separation. Normally, I’ll isolate things and baffle up different instruments in the studio, but in this live setting I didn’t have that luxury. We had one AKG C24 stereo room mic; we had the brass miked on the left; the reeds on the right, and we had an overhead Neumann U47 vocal mic on the brass side, too, because there’s a vocal part on the song.

The brass players are sort of playing into the horn mic, and then leaning up to the overhead mic when they do their vocal. The idea was to really stay away as much as possible from making choices that wouldn’t have been used in a session in the ’60s, and except for the fact that we used nine tracks, without any bouncing, I’d say we kept it pretty tight.

You opted for the UA 610-B Tube Preamp & EQ plug-in to get that period sound, right?

Yeah, we put a UA 610-B Tube Preamp plug-in in the Unison slot on every channel in Console to give us the equivalent of a Universal Audio 610 desk. In fact, the ability to create a custom Neve, API, or SSL desk is one thing I love about the Console app.

So each channel had the 610-B plug-in, Studer A800 Multichannel Tape Recorder, and Fairchild 660 Limiter plug-ins. Very simple and straightforward.

Did you need to add any ambience?

I didn’t add anything in terms of reverb because we’re recording in this old hall that was built in the 1700s, then burned down and rebuilt in the 1800s, so you really can’t get more character than that. Just really nice, rich old wood. Although the room was smaller than I’d anticipated.

You’ve got the resources and the skills to use pretty much any plug-in, so what makes Apollo and UAD plug-ins special for you?

When I discovered Apollo interfaces, it changed my life. I came up in the era of tape, recording on SSL and Neve boards, and tweaking real Lexicon reverbs, and being able to actually touch and control all that stuff. So, even as digital recording became the norm, and it was easy to get kind of “plug-in happy,” I wouldn’t say that there were ever very many of them that I really loved using.

Then I started using Apollo interfaces and UAD plug-ins in Console, and my productions sounded so much better than what I had before.

I mean, I’d even A/B the UAD plug-ins against the comparable hardware units when I was in big studios — recording through a vintage hardware Neve 1073, for example, then comparing it to the UAD Neve 1073, and they always sounded great against the “real thing.”

What would say to someone hoping to emulate your success as both producer/composer and business person?

You have to take control of your schedule! I can only speak for myself, but I really can’t mix being a producer and a businessman. If you’re following a musical idea in your studio, you’re focused, you’re locked in. And if you have to interrupt that with an email or a phone call, it’s completely disruptive to your creative process.

I did this thing a few years back I call “Taking Back Your Schedule,” where I made some firm decisions about, y’know, this is my phone call and email time. This is when I’m reachable. And this is when I go into creative mode without any interruptions, unless they’re urgent.

It’s tough. There have been times I’ve been working with artists, and it’s been very problematic when business keeps interrupting the flow. So I’ve been working more and more to make sure that separation really sticks.

In general, mornings up to about 2pm are for business calls and emails and all that; 3pm on is for creating music. Look, if you’re going to do both — and in many ways you especially need to do both these days — you need to craft a system that works for you. Doing so will make a significant impact on your ability to be productive in your creative and business life.

The Laws of Physics

Essentially, phase refers to sound waves — or simply put, the vibration of air. When we listen to sound, what we’re hearing are changes in air pressure. Just like the ripple of a stone in water, sound is created by the movement of air. And just as in water, those movements cause a rippling effect — waves comprised of peaks and troughs. Those waves cause our eardrums to vibrate, and our brains translate that information into sound.

When we record sound, the diaphragms in our microphones essentially replicate the action of our eardrums, vibrating in accordance with those waves. The waves’ peaks cause the mic’s diaphragm to move in one direction, while their troughs generate movement in the opposite direction.

The first illustration below shows what happens when we’ve got two channels of a signal in phase. When both channels are in phase, we hear the sound at the same amplitude level at the same time in both ears.

But if one side of the stereo signal is reversed, as shown in the second illustration, the signals will cancel each other out. In fact, if we were using a pure sine wave, combining both signals out of phase would result in silence, since the sounds would literally cancel each other out.

In the real world, we normally don’t listen to pure sine waves. Since most of the music we hear and the instruments we record are a complex combination of multiple waves and harmonics, the results of phase cancellation will be equally complex.

In the Studio

When recording, phase issues can quickly become complicated, usually becoming a problem when more than one channel is used to record a single source, such as stereo miking a guitar, multi-miking a drum set, or using a microphone/DI combo for bass. As sound waves of different frequencies reach different microphones at different times, the potential for one mic to receive a positive phase while another receives a negative is greatly increased, and the relationship between all of these waves’ phases can be unpredictable. In fact, the more mics in play, the more inevitable some sort of phase issues become.

Let’s look at a simple scenario, like a stereo recording of an acoustic guitar.

Most often, two mics will be set up, with one pointed toward the sound hole to pick up the lower frequencies, and the second mic pointed toward the neck and fingerboard to pick up the attack. Of course, the guitar’s frequency range covers several octaves, which means a wide range of different audio wavelengths. Since the mics are a fixed distance from the source, those different waves will arrive at the mics at different points.

Inevitably, one or more harmonics will end up sounding weaker than the rest. Your best practice would involve moving the mics very slightly — even a fraction of an inch can make a difference — until you achieve the best sound to your ears. Another solution would be to use a mid-side miking technique, which you can read about in our Mid Side (MS) Mic Recording Basics article.

Again, the more microphones used in a recording, the more potential for phase problems. In modern music recording, that usually points to the drum kit. Consider even a single snare drum, miked from above and beneath. Since the top and bottom heads of the drum are usually moving in directly opposing motion (when the top drum head is hit, it moves inward, causing the bottom head to move outward), the two mics will record signals that are directly out of phase.

Now factor in the hi-hat mic, a pair of overheads, at least one kick drum mic and one on every tom, not to mention the relationship to ambient mics, and you’ve got a sonic soup that’s ripe for phase problems. That’s why many microphones, as well as mic preamps and consoles, offer a phase flip switch. It's also why a lot of “old school” recording engineers wax nostalgic about the days when they recorded a kit with only two or three mics!

There are plenty of other “gotchas” that can introduce phase problems into your recordings. For example, a bass track recorded direct (DI) can be too clean sounding, so putting a mic on the bass amp cabinet and mixing the two sounds can give the extra “oomph” it needs — but it can also introduce phase problems.

Even certain delay settings, including pre-delays within a reverb patch, can create a delay of your original signal that ends up being out of phase

Check Your Speakers

Phase cancellation can also occur by simply wiring speakers incorrectly, inadvertently reversing the polarity of one channel. It’s surprising how many home stereos — and even project studios — have their monitors wired out of phase. In some circumstances, it may not even be apparent without careful listening. Though this is commonly referred to as “out of phase wiring,” technically-speaking it’s an issue of polarity. That said, the audible effect of this polarity reversal is the same as you get with phase cancellation.

The easiest way to check your speakers is to sum your mix to mono (more on this later). Many stereos and most mixing consoles allow you to do this, but even in stereo, there are some telltale signs of phase problems.

What does a phase problem sound like? Since phase cancellation is most apparent in low frequency sounds, the audible result of out of phase monitors is typically a thin-sounding signal with little or no bass sound. Another possible result is that the kick drum or bass guitar will move around the mix, rather than coming from a single spot.

Another common artifact of out-of-phase stereo mixes is where signals panned to the center disappear, while sounds panned hard to one side remain. Often this will be the case with a lead vocal or instrument solo — the main part will vanish, leaving only the reverb. In fact, this is how many of those old “remove the lead vocal” karaoke boxes work — they flip the phase of one side of the stereo mix, relying on the assumption that in most commercially recorded tracks, the lead vocal is panned dead center.

So what’s the fix?

As with most things, the answer is “it depends.” Assuming you identify a phase problem during the recording process, a fix is as easy as moving a mic or flipping the phase on a mic or its input channel.

When attempting to capture ambience, there's also a quick cheat: the 3:1 Rule of Mic Placement. Put simply, when using two microphones to record a source, try placing the second mic three times the distance from the first mic, as the first mic is from the source. So if the first mic is one foot from a source, the second mic should be placed three feet from the second mic. Using this simple 3:1 rule can minimize phase problems created by the time delay between mics.

Of course, if the problem doesn’t show itself until you’re mixing, it’s often possible to pull the tracks up in your DAW, zoom in close on their waveforms, and slightly nudge one track just a bit. You’d be amazed what a difference just moving a track by one or two milliseconds can make. There are also some very effective phase alignment plug-ins on the market that can really clean things up — and even serve as great creative tools — one of which is the UAD Little Labs IBP Phase Alignment Tool Plug‑In.

Sum It Up

We’ve only scratched the surface, but the bottom line is that phase issues are a fact of life, and practically unavoidable.

The first order of business is to identify the problem. Most phase problems will not show themselves in stereo, and will only appear when you collapse your mix into a single summed channel. That’s why it’s critically important, as you build your mixes, to check them regularly in mono. Don’t wait until you’ve got a completed mix to sum it into mono. Check the basic tracks, especially drums and bass, early on in the process when the arrangement and the mix are less dense and fewer things are going on. And check it again every time you add a few more instruments, or change a track’s EQ, or add reverb.

As with many things, the sooner you catch a phase problem, the easier it will be to fix. Happy mixing!

Watch as they detail how to find the perfect song to infuse with a ‘60s funk/soul vibe, as well as the challenges presented by recording a live band in a space built in the 1800s. Then learn about the recording and microphone techniques used in the ’60s and how Che used Apollo’s realtime plug-in processing to emulate an analog UA 610 console with Fairchild compressors on every track to create a finished sounding session that made an instant impression on the band.

Live Tracking

Exclusively using Apollo 8p and Apollo Twin Mk II Audio Interfaces, Che and engineer Eric Corson created a virtual Universal Audio 610 mixing board by running all of the mics through Apollo's Unison-enabled UA 610-B Tube Preamp & EQ plug-in. A Fairchild 660 compressor plug-in also resided on each channel along with the plug-ins. Top and bottom-end EQ on the 610-B was boosted slightly on the drums, and a top boost was added to the brass section. Only nine tracks were required to record the entire ensemble into Pro Tools.

Che leaned heavily on vintage mics that would be right at home on a classic ’60s funk/soul session: a single Coles 4038 on the drum kit; a Telefunken U-47 on the upright bass; a Telefunken U-48 on the Hammond B3; a Coles 4038 on the brass section (with a U47 for that section's vocals); an RCA 44BX on the reeds (with a U67 for their vocals); and an AKG C-24 stereo room mic which captured the tone, essence, and space of historic Preservation Hall, deep in the heart of New Orleans’ French Quarter.

The Mix

Inspired by the sound of Duke Ellington’s classic “Perdido,” Che kept plug-ins to a minimum on the final mix, relying on the Ampex ATR-102 Tape Recorder plug-in for vintage analog tape warmth and saturation. As for reverb, Che initially experimented with some plate reverb on individual tracks, but concluded that the room’s natural ambience was hard to beat. “We recorded right in Preservation Hall, which was built in the 1800’s,” says Che. “I mean, you can’t get more character than that.”

Hear the results and watch the Preservation Hall Jazz Band performance in Part 1 of this series.

Bonus: behind the scenes photos from the session

Live Tracking

Exclusively using Apollo 8p and Apollo Twin Mk II Audio Interfaces, Che and engineer Eric Corson created a virtual Universal Audio 610 mixing board by running all of the mics through Apollo's Unison-enabled UA 610-B Tube Preamp & EQ plug-in. A Fairchild 660 compressor plug-in also resided on each channel along with the plug-ins. Top and bottom-end EQ on the 610-B was boosted slightly on the drums, and a top boost was added to the brass section. Only nine tracks were required to record the entire ensemble into Pro Tools.

Che leaned heavily on vintage mics that would be right at home on a classic ’60s funk/soul session: a single Coles 4038 on the drum kit; a Telefunken U-47 on the upright bass; a Telefunken U-48 on the Hammond B3; a Coles 4038 on the brass section (with a U47 for that section's vocals); an RCA 44BX on the reeds (with a U67 for their vocals); and an AKG C-24 stereo room mic which captured the tone, essence, and space of historic Preservation Hall, deep in the heart of New Orleans’ French Quarter.

The Mix

Inspired by the sound of Duke Ellington’s classic “Perdido,” Che kept plug-ins to a minimum on the final mix, relying on the Ampex ATR-102 Tape Recorder plug-in for vintage analog tape warmth and saturation. As for reverb, Che initially experimented with some plate reverb on individual tracks, but concluded that the room’s natural ambience was hard to beat. “We recorded right in Preservation Hall, which was built in the 1800’s,” says Che. “I mean, you can’t get more character than that.”

Welcome to the first installment of “Producer Presets Unpacked,” where we’ll focus on a UAD plug-in preset and its designer – learning how they created it, why it works, and how pros use presets in their everyday workflow.

To get us started, we turned to Manhattan audio guru Chuck Zwicky, a platinum-selling producer, engineer and mastering ace who’s worked with Prince, Soul Asylum and Nine Inch Nails, Zwicky has helped create numerous UAD presets for the MXR Flanger/Doubler, Lexicon 224 Digital Reverb, Pure Plate Reverb, and, today’s focus, the acclaimed Manley VOXBOX Channel Strip plug‑in, which Zwicky played a key role in refining during its early stages.

Do you use presets in your own sessions, and if so, how?

I like to say that a preset is like showing someone a guitar lick. It’s a really neat exercise, and though it’s probably not going to work in your song the exact way I’m playing it, it’ll give you a starting point for what you might do.

So I think of my presets very much as jumping-off points. There’s no way I can expect anyone to simply call up one of my presets and be done.

I’ve never used a preset in my life. On anything. Not on a synthesizer, not on a multi-effects processor, hardware reverbs, nothing. Sure, I have chains of effects that I may like to use on a vocal, but not presets within that. But even if you do use presets, ultimately everything you do still needs to be tailored for the application you’re going to use it on.

Let’s say you have something like my “Warm Male Vocal” preset: Is that for more of a Michael Jackson vocal sound or is it Barry White? Obviously, everyone’s got a different register and timbre. That said, if you go through some presets, and arrive at something that sounds like it’s working pretty well, look at what it’s doing, learn from it, and adjust it as you need to.

But it’s understandable that many of us may want to “defer to the experts” and trust your settings as a starting point rather than starting with a blank canvas themselves.

Sure. The potential danger with presets, though, is that you can endlessly scroll through them looking for something that perfectly suits your preferences, rather than having a vision in your head of what it is you want to go for, and then dialing in settings in that direction.

You only become an expert through repetition. If I’m working on an outboard console, there are no presets. It’s all intuition, knowing when something’s not translating, and having an instinct about which EQ knob to grab to help improve it.

But if you look at the presets I created for the Manley VOXBOX, you can ask yourself, “why did he put that knob at this setting?” That’s the real benefit to presets — they can be really educational.

What was your goal with the “Warm Male Vox” preset on the Manley VOXBOX Channel Strip.

I wanted a preset that pushes a bit of warmth and distortion. You can see that I have the tube preamp gain set at the maximum, +60dB, while the input level is turned way down.

Here’s what is going on behind the scenes with the Manley VOXBOX circuit: the Gain switch adjusts the gain of the circuit by changing the amount of negative feedback around the amplifier. That’s why they call it “feedback-controlled.” The more negative feedback you apply to an amplifier, the lower the distortion goes.

So, when you’re adding, say, 20dB of negative feedback, you’ve just reduced the gain from 60dB down to around 40dB. It’s going to be much more linear, it’s going to be extremely clean, right up to the point where the amplifier clips.

But if you reduce the negative feedback, there’s going to be more lower order harmonics, second harmonic and third harmonic, in the signal. And obviously if you’re getting 20dB less negative feedback, the gain is up 20dB and you can attenuate the input to still get a bit of a warmer sound. If you want a cleaner sound, you just add more negative feedback by turning the gain down and the input level up.

The way I have the gain stage set in “Warm Male Vox” is optimized for a certain amount of lower order distortion and hitting the input level too hard is just going to push it into clipping. You’ll notice that the output knob is also trimmed back to -6dB. The reason the output is tuned down a bit is to compensate for the fact that there’s a lot of gain at the amplifier itself which I did here for tonal reasons.

You’re basically trying to balance the gain structure?

Well, you want to find a sweet spot with distortion. By turning up the input even just a bit more, you can hear that you start to go into heavier distortion. So, this is set up in a place where, hopefully, you’re close to a sweet spot where it’s hitting that amplifier in a way that’ll give you just enough distortion to cause a certain amount of warmth, but not enough to blow you out.

By the way, in this respect, the bypass button is your best friend. Because you’re able to A/B and hear exactly what the differences are between your original signal and the effected signal, you’ll really know what it is you’re hearing. Don’t be afraid to use it.

In your "Warm Male Vox" preset, the VOXBOX's compressor section is set to the fastest attack and release. Why?

Because, in the Manley VOXBOX, the gain reduction element is a Vactrol opto-isolator, a passive light-dependent clamping resistor, driven by an LED — so the user has very little control over the speed of the compressor.

You can’t really do anything other than slow it down from the natural dynamic envelope. A lot of compressors that use similar technology — like the LA-2A and LA-3A, although technically they’re driven by an electro-luminescent panel rather than an LED — they’re all fairly slow, and I don’t like a compressor that’s any slower than that.

So even at the fastest setting, the natural physical state of that gain reduction element is what you’re using. That setting, for me, is the Manley VOXBOX’ compressor’s most natural-sounding state.

You’ve got the De-Esser set to 3K, and yet you’re simultaneously boosting 3K in the EQ section. Why?

Yeah, I’m boosting the EQ section a little bit at 3K, and if you bring up that region, you’re naturally going to hear it a lot more, and that’s going to allow that top EQ band to be more apparent when the vocals are softer, when things are a little less intense. The de-esser is set to that same region to make sure that nothing rips your head off if the signal increases. Basically, the De-Esser is really just a limiter that’s going to kick in for those particular frequencies when it’s needed.

Interestingly, I’m also adding some very low frequencies, around 70Hz, and dipping around 200Hz, and that’s just to give it a bit more chestiness without a lot of murk or boom; it would be great for a male spoken vocal, too, or any male voice that’s a bit lower in register. If a male rock singer is singing an octave up, you may not notice much impact at all with that, but with, say, a radio announcer, they would have a lot more voice in that range. Still, to take out any clutter, you’d want to dip just above that, in that 200Hz range. In terms of sculpting that low-end response, having the low-end roll-off in the preamp section at 80Hz, rather than 120Hz, just prevents any sub-sonic stuff from getting in there.