Music Production

Using a motorized fader to write automation is so much better than using a mouse. Sure, you can get by using a mouse. But, for that really professional touch, when automating the levels in a mix, you need to be able to “ride the fader.” This is the technique of shaping the dynamics of a performance through a series of detailed fader moves. In essence, the mix engineer/producer is hand tailoring the drama of the performance to match the flow of the song. For example, on a vocal track, pushing the level of a sustained note up instead of allowing it to fade out naturally, bringing up the level of a word that was a little too quiet, pushing down a phrase that was too loud, removing a lip smack, fixing sibilance, or creating the perception of a crescendo where previously there was none.

After I’ve written an initial level across an entire vocal track, I then enter the Touch automation mode. This mode will only write the automation for as long as my finger is on the motorized fader. I carefully listen through to my vocal track, and as I hear words and phrases that require level adjustments, I simply touch the fader and write the adjustments. This process is infinitely faster than trying to program hundreds of automation breakpoints using your mouse. (Ever try riding a mouse? I don’t recommend it.) Plus, riding the fader forces you to really use your ears and listen to exactly how the part sits in your mix, versus depending on your eyeballs to make level adjustments that you’re only guessing will be correct. When you ride a fader you’re getting real-time feedback about the part’s volume. Mixing is about what sounds best over your studio speakers, not what looks best on your computer screen.

Here’s a typical example of an automation level on a vocal part, written using a motorized fader. Don’t try programming these sort of detailed automation moves with a mouse.

The two big excuses I hear for not getting a control surface with motorized faders are cost and desktop real-estate. Thanks to a couple of innovative products, neither of these excuses holds any water. The PreSonus FaderPort and Frontier Design Group’s AlphaTrack are both diminutive control surfaces featuring a single motorized fader, a handful of knobs and buttons, and both work with all the foremost DAW programs available today (from Pro Tools to Sonar, just to name a couple). Best of all, their street price, $149 and $199, respectively. (Check out Sweetwater.com for those prices.)

A common question that students ask is, “How do I use EQ? What’s the best way to EQ each instrument in my mix?” Unfortunately, there’s no simple answer to this question. Unlike a compressor plug-in, most EQ plug-ins don’t have presets—if only it where this simple. Instead, EQ curves vary from mix to mix, and from track to track. Even the same instrument, recorded during the same session, but in a different song, will be treated differently. This is because no one instrument is ever heard in a vacuum, and every arrangement is unique. Consequently, each mix requires its own set of individual EQ curves to make it sparkle and shine.

That said, here’s the rub. The secret to efficient EQ processing is twofold: an ability to hear the frequencies that you want to change, and a working knowledge of the EQ controls with which to do the job. Both of these orders are easily accomplished by themselves, it’s putting them together that can be challenging.

In order to accurately hear your music you must have a good monitoring system, professional studio monitors, and, preferably, more than one set of speakers. (See my earlier blog on Setting Up Multiple Monitors for Better Mixing.) Your monitors must be positioned properly in your room and your room should be tuned to achieve the best possible listening environment. (I’ll discuss how to tune your room in a future blog.) The bottom line is this, if you can’t hear what you’re working on because all you own for monitoring is a pair of headphones and computer speakers, you can’t expect to become an EQ master.

Next, you’ll need to understand all of an EQ processor’s parameters. For example, the difference between Frequency and Q controls, and when to use a high-pass filter versus a low-shelf EQ. Such details are explained very nicely in the PDF document that comes with Pro Tools 7.4, and can also be downloaded directly from the Digidesign Web site, the DigiRack Plug-Ins Guide (version “v74”). (Some versions of the DigiRack Plug-Ins Guide without the “v74” appear to be missing the DigiRack EQ plug-in chapter.) Consequently, I won’t waste space trying to explain all of these parameters here, just read the manual.

Now, let’s jump to the chase, how to go about finding a particular set of frequencies in your signal that you can hear needs help (and you can hear this because you have properly set up monitors and a fine listening environment). My favorite technique is to insert a parametric EQ, and to use it like an EQ magnifying glass in order to find my troublesome frequencies. This is a technique that has been in use ever since the invention of parametric EQ, so I’m sure it has several other names, but I call it the “magnifying EQ trick.” Here’s a video I made on how the process works.

Do you think you’re set with one pair of professional studio monitors? If so, think again. Most home-studio owners purchase one good pair of powered reference monitors and that’s it. However, to truly hear how your music will translate to the outside world, the real world beyond the four walls of your comfy studio, you should be working on at least two sets of speakers: your main near-field monitors and a set of small, inexpensive desktop computer speakers (minimonitors). This dual monitor approach will let you hear how most listeners will be hearing your tracks—over a cheap home stereo system, a television, or computer speakers—instead of the precise, accurate, and “flat” sound of your pricey studio reference monitors.

Of course, chances are that your audio interface only has one set of monitor outputs (a pretty standard design). This begs the question, “Where do I connect another set of speakers?” The solution is to add an analog monitor control box to your system. The stereo mix coming out of your audio interface’s monitor output is then connected to this box and split (multiplied) into several monitor output paths, each of which can be sent to its own monitor destination (including headphones). Two of the most popular solutions on the market are the PreSonus Monitor Station ($400) and the Mackie Big Knob ($390). Each of these units is designed to sit on your desktop and provide ergonomic monitor control, making it easy to switch between monitors while you’re mixing without losing your “sweet spot” (the listening position between your monitors that sounds the best).

The point of near-field monitoring is to remove as much room coloration from your listening position as possible. Though it’s no less important to tune your room for better acoustics (a topic for another blog), a proper near-field setup can reduce much of the room tone that you would normally hear if you were seated farther away from your monitors, outside the sweet spot. Today, most well-designed studio reference monitors feature frequency fine-tune controls for tailoring a speaker’s response to best fit your listening environment. For example, to compensate for a room that adversely emphasizes low frequencies, you could roll off your monitor’s low-frequency response by a couple of dB. Foam speaker-isolation wedges (such as the Auralex MoPAD) are also an option and allow a monitor to be decoupled from what it sits on, preventing the speaker from transferring sound to the surface in a way that might adversely affect what you hear.

Now, for the speakers. Keep in mind that the size of the low-frequency drivers (the woofers) determines your monitors’ low-frequency output. The larger the woofer, the more bass you’ll hear in your mix. Consequently, you should stick with a 6-inch woofer or larger for your main monitors. In my opinion, the best enclosure-size-to-bass-output ratio for your dollar comes from monitors with 8-inch woofers (such as the Mackie HR824 or Event Studio Precision 8). Of course, you can add a subwoofer to augment monitors with small woofers, but for most music-production applications, having the bass in your face is preferable to having it under your seat or to the side of your workspace. By comparison, your minimonitors should have a 3- to 4-inch woofer (such as the Edirol MA-7A or M-Audio StudioPro3). And, for the sake of quality and convenience, the minimonitors should be self-powered, just like your main monitors.

How your speakers are set up is also crucial for good monitoring. For the best near-field monitoring possible, make sure that your speakers are upright and level with your head. When seated in the sweet spot between your speakers, your head and the two speakers should comprise the three points of an equilateral triangle. You can place the minimonitors just to the inside of your main monitors. Make sure that the speakers are as far away as possible from any walls to avoid potential low-frequency interaction with your room’s physical structure. Pushing your speakers against the wall or shoving them in a corner is never a good idea. Remove any impediments that might interfere with a clear line of sound from the speakers to your ears (such as plastic figurines and stuffed animals—really, I’ve seen it done). And, watch out for possible reflective surfaces just beneath the monitors (such as a large mixer or laminated tabletop) that may cause high-frequency reflections to bounce off and sully your sweet spot.

Between your main monitors, a pair of minimonitors, and a studio-quality pair of headphones (such as the Sony MDR-7509 or Ultrasone Proline 750), you can construct a clear picture of how your mix will sound in the real world without ever leaving your studio. Plus, with a good monitor controller you’ll never need to move from your sweet spot to switch monitors. Now, with all of this control at your fingertips, the only trick is to remember to get up and use the restroom every so often. Though, seriously, if that’s not inspiration enough, many an award winning mix engineer has been known to walk outside the studio, and down the hall, in order to hear how their mix sounds from a completely different perspective—for real, it really does work.

A mashup (AKA bootleg) is taking two songs and beat-matching them together to create a new blended mix of both songs. For example, the classic mashup of Kylie Minogue’s “Can’t Get You Out of My Head” and New Order’s “Blue Monday.” It’s often done using full stereo mixes (with vocals), or, alternately, an a cappella and a stereo mix (possibly an instrumental). To hear a variety of well crafted mashups, check out Party Ben.

Mashups became such a hit on the dance-floor that some producers (such as Richard X) went on to remake parts of the original songs in order to clear the entire mashup for commercial release. For example, the 2002 UK hit by the Sugababes, a combination of the lyrics from Adina Howard’s “Freak Like Me” and the music of Gary Numan’s “Are Friends Electric?”

The point behind my little history lesson is, you don’t always have to play a traditional instrument, or even record a track, in order to be wonderfully creative with music. I have the privilege of working with music production students at all levels of experience, some are seasoned musicians while others are just starting piano lessons. Obviously, for our production project in class, I expect students to create their own tracks, one way or another. It’s a snap for experienced players to record a performance, but a serious challenge for students just beginning an instrument to record something decent. As an alternative, I encourage the use of MIDI files, a cappella mixes, and sampling. (For educational purposes only, of course.) These resources can provide a signal and a musical performance with which to practice your production chops whether you play an instrument or not.

However, if you have never worked with samples or imported a MIDI file, taking advantage of these resources can be intimidating. One of the best ways I know to explain the whole process is to show you in a song. So, without playing a darn thing, just using my ears and production skills, I produced a mashup in Reason 4 using an a cappella, a MIDI file of a cover tune, and a sample of the original tune — all items I found for free on the Web. This mashup features Tone-Loc’s “Funky Cold Medina” and Kraftwerk’s “The Model.”

You can download the Reason 4 song file below (it’s about 8 MB) and explore the production, from its samples to its mix. To download an MP3 of the mashup, visit my myspace page.

Cold Medina Mashup

Of course, just because you don’t understand compression doesn’t mean that you’re mentally challenged. As a rule, the compressor, and how it controls a signal’s dynamics, is one of the more challenging processors to grasp. Learning how to effectively apply compression in your mix can take a significant amount of study time, patience, and good old fashioned experience.

Now, I could explain what each parameter of a compressor does and how it affects the signal. I could even give you some compression presets to get you started. But, this approach would be old hat and does nothing to help you actively hear compression and how each of its components work. You see, without the ability to hear in your mind how compression colors a signal, and to then know which parameters on a compressor to reach for in order to achieve your sound, you’re just fumbling blindly.

The skill necessary to properly operate a compressor is comparable to the ability you developed as a toddler to recognize and apply colors. You learned to visualize what color you wanted to apply to the flower in your coloring book, and you learned which color to reach for in your box of crayons to achieve your objective. The trick with compression, as with any type of processing or synthesis used in music production and sound design, is to know, instinctively, which parameters to reach for in order to create the sound you’re hearing in your head. It’s a deceptively simple process because it’s so easy to quantify, but as we all know from experience, it’s tough to put into practice.

With all this in mind, I’ve cooked up an interactive compression lesson to help you better hear compression, and learn to associate compression colors with specific compressor parameters. It’s a Reason song file full of MClass Compressors, with each Compressor adjusted slightly differently, but applied to the same snare drum signal. Each compressor’s label reflects its parameter change (such as “More Attack” or “Less Attack”), so that you can easily identify the Compressor’s parameter that you’re hearing, in relation to a base compression setting (the “Basic Compression” device). And, since a sound is rarely heard on its own, but, instead, always with accompaniment, I’ve included the rest of the drum mix as a stereo stem on Channel 12 of the mixer.

Here’s How You Work It
Press Play to start the drum pattern, then, to hear each compression setting, solo each snare drum signal on the mixer (Channels 1 to 10), one channel at a time. Leave the drum mix on Channel 12 in solo mode so that you can hear how the different compression settings make the snare “sit” in the drum mix.

Many of the changes to the snare drum’s sound are subtle and a challenge to hear, especially if you’re new to this sort of critical listening. Accurate monitors are also key in being able to hear the differences in the drum’s sound. So, if you’re not hearing the differences right out of the gate, not to worry, below is a description of what you’re listening for in each compression setting.

Channel 1: “No Compression”
This is the snare drum dry, with no compression processing.
Channel 2: “Basic Compression”
This is a decent snare drum compression setting. It is the starting point from which a single parameter is changed in the following Compressors. For example, on the “More Attack” Compressor, all the parameters are identical to the “Basic Compression” settings except the Attack parameter.
Channel 3: “Less Threshold”
Increasing the Threshold means that less of the incoming signal will be compressed. Another way of putting it is that the threshold at which the signal will begin being compressed is higher.
Channel 4: “More Threshold”
Decreasing the Threshold means that more of the incoming signal will be compressed. Another way of putting it is that the threshold at which the signal will begin being compressed is lower.
Channel 5: “Less Ratio”
There’s no easy way to explain the compression ratio. It’s math, there’s no getting around it. Ratio sets the amount of input signal necessary to cause a 1 dB increase in output signal. For example, with a ratio of 4:1, an 8 dB increase in input will produce a 2 dB increase in the output. So, less Ratio means that an increase in input signal will sound louder at the output, less compressed compared to the original “Basic Compression” setting.
Channel 6: “More Ratio”
With more compression ratio applied, more input signal will be required to produce a 1 dB increase in output signal. Consequently, the output signal will sound more compressed when compared to the original “Basic Compression” setting. At high compression ratios, limiting occurs, where, at the most extreme settings, the output level stops increasing no matter how loud the input level becomes (referred to as “brickwall” limiting). In situations where the output level is very low in volume, you can use the Compressor’s Output Gain control to turn it up.
Channel 7: “Less Attack”
The Attack parameter sets how quickly the compression will begin. So, turning the Attack up means that less of the signal’s initial transient (the very beginning of its waveform) will be compressed. This is good if you want to retain the crack and pop of the waveform’s start.
Channel 8: “More Attack”
Turning the Attack down means that more of the signal’s initial transient will be compressed. This is good if you want to diminish the crack and pop of a waveform’s start.
Channel 9: “Less Release”
The Release parameter determines how long it will take for the compression effect to fade out. So, less Release equals a shorter release time and the signal’s waveform will be compressed for a very limited duration. This is good if you want to retain the natural decay of a waveform.
Channel 10: “More Release”
Turning the Release up means that the time it takes for the compression effect to fade out will be longer. This is good if you want to compress the natural decay of a waveform, like increasing the volume as the signal fades out.

After you’ve listened carefully to each compression setting, try describing the changes in the sound that you hear. This will connect what you’re hearing to a concrete idea in your mind. And, ultimately, help you to associate a compression color with a specific compression parameter. Once you master hearing what each compression parameter can do on its own, then you will begin to hear how all of the compression settings work together to create a variety of compression effects and sonic colorations.

Here’s the Reason song file. Remember to press Play before you begin soloing each snare drum signal, and only audition one snare signal at a time.

Compression Lesson (Reason 3 Song File)