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.)

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.

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!

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)

It seems like the Pro Tools Accelerated Videos (http://www.digidesign.com/accelerated) are often overlooked as an educational resource. Sure, they’re clearly for promotional purposes, extolling the virtues of Pro Tools. But, at the same time, they are an excellent overview of Pro Tools centered music production techniques and features, both new and old. The videos aren’t nearly as in-depth as our twelve week courses here at Berkleemusic, but if what you need is a quick blast of information, an overview of a feature or technique, the Pro Tools Accelerated Videos are just perfect.

The Pro Tools Accelerated Videos can also be seen on YouTube. Here’s one on using Reason Adapted with Pro Tools and setting up the ReWire connections:

For starters, it’s a good idea to determine a general stylistic direction for your remix, because not every song will work with every style of beats. In general, the most important factor in making this choice is a song’s original tempo, because no matter how you slice, dice, and warp the audio, there’s always one constant; the less of a tempo change you make on the original the better it will sound in your remix. (This is, of course, assuming you want to hear the original song clearly in your remix. If not, all bets are off.)

Regardless of how great a particular tempo change process is, the further you stretch, compress, and slice up a piece of audio the more warbled, disjointed, and distorted it will become. Since most electronic music styles that are appropriate for a remix inhabit a select tempo range, it’s wise to choose a target remix style with a tempo that is as close as possible to the song’s original tempo. (See the chart below for a sampling of several popular dance music styles and their tempo ranges.)

Tempo changes that are less than +/-20 BPM are usually a safe bet, but this amount will vary depending on the types of sounds and performances you need to alter. For example, slowing down legato bass and vocal performances is trickier than speeding up staccato guitar and percussion parts. Speeding up a performance usually produces better sonic results, with fewer processing artifacts, than slowing it down. However, in some cases, slowing down the original by a few BPM so that you can double-time the remix tempo and reach a suitable dance music tempo can also work. So long as your tempo change doesn’t destroy the musical “feel” and audio quality of an artist’s recording, you’re in safe territory (especially if the remix has been commissioned by the artist or their record label).

To precisely determine the tempo of the original song I fly the stereo mix into Pro Tools, separate out a couple of bars and apply Identify Beat to the selected region. Make sure that the region you create is perfectly trimmed to the downbeat and loops seamlessly. Identify Beat is only as precise as your selection, so if your separation points are off the downbeats by even a little, your tempo calculation will also be off. The clear transient of a kick drum is one of the best landmarks for a downbeat, and the Tab to Transient function can make finding your kick transients an absolute snap.

Is Reason 4 Cool? Is Tony Hawk a good skateboarder? The answer is a resounding, “Yes!” But the question I inevitably hear is, “Why do I need to update Reason if I just use it as a virtual sound module.” Now, that’s an easy answer, the Thor Polysonic Synthesizer and the RPG-8 Monophonic Arpeggiator.

You’ve heard of Thor the Norse God of Thunder, right? Initially, I thought the guys at Propellerhead were just being silly, but this new synth is aptly named because it does produce some thunderous sounds. Take for example the Moog influenced Fat Boy bass patch, or the endlessly entertaining sequenced and rhythmic patches (such as Elastic Overdrive or Modular Funk Machine). Thor has a built-in step sequencer for some amazingly complex and funky sounds. And, if designing sounds isn’t your thing, the Reason 4 Factory Sound Bank comes with a dozen folders packed full of earth shaking Thor patches: Bass, Fx, Lead Synths, Pads, Percussion, Poly Synths, Rhythmic, Sequenced, Textures, Voice and Choir, Signature Patches (patches created by well-known artist and producers), and several uncategorized patches in the Thor Patches parent folder.

At the Reason Producer’s Conference in San Francisco, just before the Release of Reason 4, I distinctly heard Propellerhead product specialist James Bernard answer a question on the possibility of having an arpeggiator function in Reason with, “There will never be an arpeggiator in Reason.” Clearly, he was pulling our collective leg. The RPG-8 is dope! It is a comprehensive arpeggiator with a load of parameters for shaping arpeggios. For example, Hold, Octave Shift, Velocity, Gate Length, it can generate up to a four octave arpeggio, and there are controls for introducing random and sequenced variations (such as the Pattern Steps with which you can mute specific steps in the arpeggio). Better still, the RPG-8 can be controlled by your ReWire master application. For example, in your ReWire master, assign a MIDI track’s output to the RPG-8, send it a chord (or even a single note) and it will play an arpeggio using the instrument device that it is connected to in the Reason rack. Talk about cool!

For those of you with Pro Tools 7.3 and Reason 4, I’m attaching a ReWire session to show you how the RPG-8 can be controlled from an Instrument track in Pro Tools and sending an arpeggio to Thor at the same time. In the Zip folder you’ll find the Pro Tools 7.3 session file and Reason 4 song file. Remember to launch the Pro Tools session first, then the Reason song file. (For those of you without Pro Tools 7.3 and Reason 4, I’m also attaching an MP3 so you can at least hear what I’m going on about.)

1) Session Zip

2) MP3 Audio File