Music Production

When enabled, the Elastic Time plug-in analyzes two dimensions of your audio in order to calculate the recorded performance’s tempo. It looks at the audio region’s duration in bars and beats, and it looks for transients that represent a regular periodic rhythm in the recording. In theory, if the rhythmic content of your recording is clear, with distinct transients, Elastic Time can figure out a performance’s tempo regardless of whether the audio region is trimmed to a perfect loop or not. This is really neat when it works, but it doesn’t perform miracles (nor should you expect it to, that’s not producing your own music).

Instead of relying blindly on Elastic Time to perform your beat matching (as in, crossing your fingers and hoping for the best), there are several simple steps that you can take ahead of time to prepare the audio and ensure perfect results every time. Performing this pre-processing, even though it’s a tedious task, helps to preserve the audio quality and the groove of the original performance after your tempo change.

The easiest way to demonstrate my pre-processing approach is to take you step by step through the process. This is the best way to understand not only the steps in the process but the logic behind my approach. (Try it a different way and you’ll find out just how quickly you can end up with a train wreck.) The DRM (Digital Rights Management) free stems being offered for your remixing pleasure by Radiohead, of their song “Nude”, on iTunes is perfect for this example. The stems are $0.99 each and you only really need two of the five available, the “Drum Stem” and the “Voice Stem”. But, it’s nice to have all of the stems in your session, even if you don’t use the “Bass Stem” and “Guitar Stem” they’re handy for finding the key of the original, and the “String FX Etc. Stem” contains several sounds that are perfect for a remix.

Remixing Radiohead

1. Create a new Pro Tools session and import all of the “Nude” stems. When prompted, select the import Destination as New Track with a Location of Session Start.


2. Arrange the tracks in the Edit window with the Drum Stem on top and the Voice Stem just below.


3. Mute all of the tracks except the Drum Stem. Use Tab to Transient to locate the very first downbeat in front of the vocals (approximately 1946229 samples in from the session’s start) and separate the regions.


4. Select the Edit Group, double click on the latter Drum Stem region and separate all of the regions.


5. Select the Shuffle Edit Mode and delete the first set of regions so that the downbeat of the second set of regions scoots to the beginning of the session.


6. Disable the Edit Group and using Tab to Transient find the downbeat every two bars in the Drum Stem and separate the regions. (The only region that I left as 4 bars is the drum break, having 2 bars before the break itself.) In some instances, you can make the separations every 4 bars, but with “Nude” there’s a lot of human tempo variation and every 2 bars will produce the most precise overall beat match.


7. Select the Edit Group and double click on each 2 bar Drum Stem section and separate the sections.


8. Select each 2 bar section and apply Identify Beat to generate a tempo map for each section. This will beat match and lock each 2 bar section to your session’s tempo grid. (You could use Beat Detective to generate a tempo map, but my method preserves the original performance’s groove every two bars, while simultaneously guaranteeing that the original’s downbeat is locked exactly to the grid every two bars. This way, your beat match never drifts.) At this stage, it’s also helpful to create a Click track in order to audition your tempo lock.


9. Change the Timebase selector for all of the tracks to Ticks. This will lock the regions to their relative bar and beat positions so that they stay in beat when you make your tempo change. Then, enable the Elastic Time plug-in for each track. Set the Drum Stem to Rhythmic, the Voice Stem to Monophonic, the String FX Etc. Stem to Polyphonic, the Bass Stem to Monophonic, and the Guitar Stem to Polyphonic. Give Pro Tools time to analyze the audio. (For now, leave the Elastic Time plug-in set to Real-Time Processing.)


10. In the Transport bar, disable the Conductor and enter a new tempo in the Current Tempo field, or experiment with the Manual Tempo Slider. Since the original tempo is about 42 BPM, it’s nice to bring the tempo up to about 62 BPM. At this tempo you can then record drums in double-time (124 BPM) and be at a dance tempo! (Once you decide on your remix target tempo, if your computer is running short on processing power, your can set the Elastic Time plug-in to Rendered Processing.)

Here’s what the final results sound like, using all of the stems at 62 BPM, some effect processing, and backed by Digidesign’s Strike virtual drum instrument playing in double-time at 124 BPM:
Nude – The Strike Remix (by Erik Hawk)

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.)

With the introduction of Elastic Time, in Pro Tools 7.4, it’s clear that Digidesign is taking a proactive stance against competing programs boasting easy to use, time compression/expansion based, automatic beat matching algorithms. Though it’s tough to proselytize that Elastic Time is as easy to apply as similar functions in competing software applications, I can say, without a doubt, that it kicks butt in terms of its audio quality and the level of direct control you have over how the algorithm is processing your audio. It’s deep and offers a variety of ways to fine tune the processing, from multiple Elastic Audio “plug-ins” (Polyphonic, Rhythmic, Monophonic, and Varispeed), to individual control panels for each plug-in type, and the ability to freely edit Warp markers directly in the Edit window. Talk about control! Due to the simple fact that you have such a high degree of control over the Elastic Time processing, it requires a bit of homework to master, but once you do, you can achieve absolutely stunning results.

It’s important to note that Digidesign calls Elastic Audio a “plug-in”. However, it’s not applied in the way you’d normally think of a plug-in, as an insert on a mixer channel or an offline AudioSuite processor. Instead, it’s applied directly to a track in the Edit window, with its plug-in menu located just below the track’s Timebase selector.

Beat Matching Made Simple
Elastic Time is at its most intuitive when browsing for loops in the Workspace. Simply turn on the Audio Files Conform to Session Tempo button (it looks like the ticks metronome) in the Workspace, select one of the four plug-in types (they’re pretty self-explanatory), and when you click on a loop’s play button (the little speaker) the loop will be automatically analyzed and played back at your session’s tempo. You can even do this while your session is playing, allowing you to hear immediately whether the loop you’ve selected is a bang or a bust. Talk about optimizing your workflow! When you find the loop that you want, simply drag it into the Edit window and it will be deposited on a newly created ticks based audio track, and conformed to your session’s tempo. It doesn’t get much easier than this.

For those of you who are used to working with REX loops, it should be noted that with the introduction of Elastic Time, the Processing Preference setting, Import REX Files as Region Groups is off by default. So, if you’ve recently updated to Pro Tools 7.4 and you’re wondering why when you drag in a REX file it no longer comes in as a region group, mystery solved. To return to your old way of working with REX loops, make sure that the Processing Preference for Import REX Files as Region Groups is checked, as well as the REX and Acid Files Only selection for Drag and Drop from Desktop Conforms to Session Tempo. From the Workspace, you can also audition REX loops at your session tempo when the Audio Files Conform to Session Tempo button is on. And, if when you drag a REX loop from the Workspace into the Edit window, you want the REX loop to automatically conform to your session’s tempo, again, the Audio Files Conform to Session Tempo button must be enabled.

There’s a saying, “Copying is the highest form of flattery.” Copying is also one of the best ways to hone your production skills. Taking the time to pick apart and recreate a song done by your favorite producer is almost like being an intern for that producer. You’re getting the benefit of dissecting the techniques used to produce their unique sound without the hazards of being an actual intern (you’ll never be shackled to the coffee maker nor asked to clean up after the band).

When selecting a song to copy, make sure that it is full bandwidth audio (like what you find on an audio CD), not a compressed audio file format (such as MP3 or AAC). You need to be able to hear every nuance of the original production, and a 128 kbps MP3 file just isn’t going to cut the mustard, there’s just too much audio content missing. You can audition MP3 files to find the song that you want to copy, but when you’ve identified the song, you should buy the audio CD to ensure that you’re listening to the best quality audio available.

The object of copying a song is to get as close to the original sound as possible. However, even though this is a great bar to shoot for, it’s not usually possible from a technical standpoint. For example, the producer used a $20,000 Lexicon 960L reverb unit, and all you have to work with is D-Verb (the Pro Tools LE factory reverb plug-in). Obviously, their sound isn’t going to compare. Fortunately, simply going through the process of copying the song as closely as you can is practice enough. Even if your copy isn’t a dead ringer, you’ll still be going through the steps and experiencing the techniques required to create the producer’s sound. Of course, ultimately, the idea isn’t to become a clone of your favorite producer, it’s to learn a variety of techniques and then to apply them in your own unique ways.

Neither is it necessary to copy an entire song, from start to finish. It’s fine to copy just a short section of the song. For example, the chorus, the bridge, or simply the intro beat. The production elements that you’re wanting to emulate are, more often than not, contained in only a few bars of the music. Copying just a section makes it convenient to loop the part, then beat match your session’s tempo to the loop. This also makes comparing your copy to the original song, right in your session, a total snap. Plus, with your session beat matched to the original, it becomes possible to extract the loop’s groove (using a tool like Beat Detective in Pro Tools) and apply it to your own tracks.

Here are some of the questions you should ask yourself when you copy a production:

Attached is a Pro Tools session in which I’ve imported and looped a short drum loop from Missy Elliott’s “Sockit2me” (produced by Timbaland). Then, I’ve used Xpand! and the Pro Tools LE stock plug-ins to copy the song’s basic production sound. It’s not perfect because of the limited palette of sounds I had to work with, but it certainly captures the flavor of the original beat. In fact, I even picked up an interesting production trick along the way: hard panning a gated reverb return to the left speaker, and then hard panning the original dry signal to the right speaker. See if you can hear this effect in the original loop and then find how I recreated it in my mix.

Copycat Cool Pro Tools

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.

The stereo master fader in your DAW’s virtual mixer is not for controlling the output level of your studio speakers. This is an all-too-common mistake, but an easy one to make. I mean, it does turn down your speaker level, right? Yes, but think about what’s happening to your mix down level, the waveforms that you’re printing to disk. If your master fader is turned way down in order to keep your speakers low, and your neighbors oblivious to your beats, the mix you’re printing to disk is also going to be low in volume. We’re talking itsy-bitsy waveforms here, a potentially bad signal to noise ratio, and just a plain old poorly executed recording, It’s this exact result that most often leads the inexperienced producer to the puzzling question, “Why the heck are my mixes so low in volume?” (And, incidentally, gain normalization is not the remedy here, because in normalizing a very low waveform you’re also turning the recording’s background noise way up.)

The purpose of the master fader is for setting your mixer’s main output level. Alternately, a monitor-level knob, such as the type found on a well-designed audio interface (for example, MOTU’s 896HD or Digidesign’s 003), is used to control the output level of your studio monitors. This design allows you to set your virtual mixer’s main output to an ideal level, having peaks just below digital zero, and than, independently, adjust the level of your control room speakers. I’m sure you’ll agree that after a long night of mixing the same song over and over and over again, the ability to turn your speakers down while recording your completed mix to disk is a godsend.

In traditional analog mixing consoles, the master fader and the monitor-level knob were both built into the mixer, in adjacent master output and monitor control sections. This made the signal path from the master fader to your mix down deck, or the monitor-level knob to your control room speakers pretty easy to follow. Today, you’ll find these two vital controls living in completely different worlds, the virtual world of your DAW, and in the real-world, on your audio interface. As if understanding signal flow wasn’t already enough of a challenge, now you need to bridge alternate realities!

To make matters even more confusing, most popular MIDI control surfaces give you direct control over your DAW’s master fader. Case in point, the Mackie Control Universal features a physical master fader that’s tied exclusively to your virtual mixer’s master fader. Nice feature, but there you go again, reaching for the master fader to turn down your speakers when you should be reaching over, or under, or around, or wherever your audio interface is stuck in order to grab its monitor-level knob. Clearly, this sort of stretching is great for yoga class but undesirable for mixing, because it moves your head out of the “sweet spot” between your speakers. The solution is to invest in a control surface that features a monitor control section (such as Digidesign’s Command|8 or C|24), or add to your system a dedicated monitor control sidecar (such as Mackie’s Big Knob or the PreSonus Monitor Station).

In the meantime, pull your audio interface to within an easy reach, and the next time you need to adjust your speaker level take a hold of its monitor-level knob and not your DAW’s master fader.

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

There’s a lot that goes into producing a convincing drum track, especially when your drummer is a software sampler (such as Reason’s Redrum and NN-XT, Native Instrument’s Battery, or MOTU’s MachFive). Indeed, the shear number of techniques employed to create a great drum track would keep me writing blogs for months to come. But, rather than go on and on about how to produce realistic sounding drums, let’s cut to the chase and look at how it’s done in a Reason song file.

Using Reason 4, I’ve cooked up a song file that demonstrates how to produce and mix realistic sounding drums. I’m using only samples found in Reason’s Factory Sound Bank and Redrum as the sample playback device. You can do much more with the NN-XT in terms of the shear number of samples and velocity zoning. However, since most beginners reach for Redrum first, I decided to hold off on the NN-XT. The mix is not mastered (there’s no Mastering Combinator or Maximizer in the rack) so that you can see and hear how your drum levels should be hitting before mastering. (Mastering should be used to make a great mix sound awesome. Unfortunately, mastering is too often used to make a poor mix sound passable. But, that’s a subject for another blog.)

If you have Reason 4, you can open up this song file and explore the connections and settings. Of course, your drum tones and compression levels will vary with each individual mix, in relation to the other instruments in your song. For example, you might want your snare to have less compression on the initial attack of its waveform, for greater snap, or your kick to exhibit more mid frequency pop around 8 kHz. Fine adjustments such as these are easily accomplished when your devices are properly set up and routed, as they are in this song file. Alternately, if your drums aren’t properly routed, fine tuning your drum mix can be an exercise in frustration. Many of the techniques employed in this drum mix are the sorts of things that I teach in my Berkleemusic course, Producing Music with Reason.

Here’s a list of the production techniques used to produce this drum track:

· Compression and parametric EQ inserts
· Parallel compression
· Group effects
· Individual outputs
· Gesture sampling
· Proper levels and gain structure
· MIDI performance sample (a drum sequence created by a real drummer)

Turn your speakers up and have fun exploring this song file!

Redrum Drum Mix Demo

During this season of gift giving and gear lust I would like to remind us all (myself included) that it’s not the gear that makes great music. You’re the one that writes, plays, and produces the music, not the equipment. In fact, I bet, if you put your mind to it, you could write a cool beat with a kazoo and some cardboard boxes, and nary an AC outlet in site. But, instead, you’ve embraced high-tech music gear as your recording instrument of choice.

Sure, quality recording products give you the tools to make excellent sounding recordings. Assuming, of course, that you have some basic knowledge of music and recording engineering. However, ultimately, it’s the quality of the song that matters the most, because no matter how well a bad song is produced, it’s still a bad song. By comparison, a well crafted song that is also well produced has the potential to be a hit.

So, when purchasing your next amazing high-tech music making gizmo, keep in mind that it will not be able to magically make your songs better. If you already own all the equipment to record your tracks (for example, a computer, DAW software, audio interface, MIDI controller, and quality studio monitors), and you imagine that this next piece of gear will make your productions that much better, think twice before you spend your hard earned cash. Sometimes, it’s not a lack of gear that’s holding you back, but a lack of knowledge about how to use the gear you already have to its absolute fullest potential.

Fortunately, there is today a wealth of resources available to help you improve your craft, on both technical and creative fronts. If you can play an instrument and have a decent grasp of music theory and notation, you’re ready to jump right into production classes (such as Pro Tools 101 and Producing Music with Reason). If you’re shaky on the fundamentals of music and have not mastered an instrument, than you should start at square one and study music theory and an instrument (such as Music Theory 101 and Berklee Keyboard Method). If you’re not currently in a position to take Berkleemusic online courses, to start you on your way, treat yourself to a couple of books from Berklee Press (such as Instant Bass and Producing in the Home Studio with Pro Tools).

Lots of enthusiasm for making music is excellent, and most certainly a necessary ingredient to being successful in this industry. However, along with all that enthusiasm you’ll also need to know which end of a microphone boom to sing into. (See the YouTube video below.)

Happy Holidays! Have a safe and musical New Year!