Music Production

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!

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’s easy for me to advise you to finish all of your productions, no matter what (see my earlier blog, How to Become a Great Producer), but what exactly are the steps to getting your tracks done? Let’s address the all too common complaint of, “I start lots of cool tracks but can’t seem to finish any of them.”

In my experience, there are two prevailing reasons for not getting your tracks done, you are your own worst critic and never satisfied with your music, and you are uncertain as to exactly how the production process should progress. Either one of these reason’s would be enough to derail your train, put them together (as is often the case) and you have a roadblock that’s a serious challenge to overcome.

Learning the technical skills behind music production is easy to address. Take some courses at Berkleemusic.com and study hard. No matter the area in which you need to improve and acquire proficiency, we’ve got it covered—musicianship, songwriting, music theory, recording, sequencing, mixing, and mastering—it’s all here. Quelling your inner critic so that you can finish your music requires a distinctly different and, most often, a less obvious path. However, no matter the path you take to find your balance and harmony, embracing a clearly defined process (complete with a list of steps you need to follow to reach your goal) can help you to better steer your music productions from start to finish. For me, the process of writing rough drafts, lots and lots of rough drafts, is the key to bringing my productions to fruition time and time again.

A rough draft is a sketch of your intentions. It’s a process that helps to bring the music you’re hearing in your head into the physical world. It lets you find the balance between what you can imagine and what is actually possible given your skill level and the tools at your disposal. It is a wonderful way to quickly build the framework for a song, because without this framework you would have no structure on which to hang your musical ideas. It’s about getting the big picture recorded first, so that you can focus on the details later. Writing drafts has become a truly essential part of my creative process, and without it I would most certainly get stuck in the details, lacking a clear direction for my song.

Of course, writing rough drafts, like any other discipline, takes practice. You’ll need to accept that your first, and possibly second drafts will probably suck—mine certainly do. But, like any discipline, with regular practice writing rough drafts will become easier and more fluid over time.

Here’s a list of the rough drafts I write each time I do a track:

1) The Basics: A starting point for your song. This might be a beat, a bass line, some chords, a melody, or any combination of these ingredients. It might be two bars or sixteen. Get it down quickly and without second guessing yourself.
2) Put It In the Pot: Record every idea that you think might work. Throw it all into the mix, to be sorted out later. Again, do this quickly and without second guessing yourself. Remember, hard drive space is cheap and MIDI sequences require hardly any storage space at all.
3) Song Structure: Rough out a basic song structure. From all the parts that you put down in the last step, arrange a working song structure. Slide the parts around, mute regions, slice and dice, whatever works to create distinct song sections using the parts that you’ve recorded so far. Keep in mind that it’s easy to add or subtract bars later on should you need to alter the song structure.
4) Produce Transitions: Now it’s time to get detailed. Produce your song section transitions using techniques such as drum fills, synth rises, arrangement builds, musical crescendos, chord inversions, etc. Take your time and really work out the performances. You may find yourself making some significant song structure changes at this stage of the production.
5) Bells and Whistles: The final stage of the music production. This is the time for you to add those last production touches, the ones that will have listener’s saying, “Wow, that was cool!” This is also a good time to begin your rough mix, the first mix pass where you begin finalizing levels, panning, EQ, compression, and group effect settings. (After this step it’s onto the final mix down and mastering.)

It’s true that with a Mac Pro computer you can probably insert a D-Verb reverb plug-in on practically every audio track in your Pro Tools session. However, this is a clear cut case of, “Just because you can do it doesn’t mean you should.” Reverb should be applied as a group effect, on a bus, in the send/aux return position. There are a few reason’s why:

1) The purpose behind reverb is to mimic the ambiance that’s captured during the recording of a live performance. With multitrack recording, performances are recorded at different times, often in different spaces, and sometimes with no ambience at all (as in the case of a direct line input recording or a virtual instrument). Consequently, there is an absence of the cohesive ambiance that naturally occurs in a live recording situation. Placing a reverb on your mixer’s bus let’s you send an appropriate amount of signal from each performance in your session through the same virtual space, just as if you had recorded everything live, in one space.

2) Reverb creates a sense of depth in your mix, of front to back space. The more reverb a signal is blended with, the further back in the mix it will sound. Alternately, the less reverb a signal is blended with the closer it will sound to the listener. By using a channel’s send to add different amounts of reverb to a performance you can bring an instrument forward, or push it to the back of your virtual sound stage. For example, a lead vocalist can be made to sound in front of the stage by mixing her with only a little reverb, while a cello can be made to sound at the far back of the stage by mixing it with much more reverb.

Always keep your reverb’s dry/wet mix parameter set to a 100% processed return. The reverb effect is created by mixing the dry channel signal with the wet reverb effect return. If your reverb’s dry/wet mix parameter is less than 100% you’re just returning dry signal back into your mix, and this is not the object. Use a channel’s send to adjust the amount of reverb you want mixed with a track. The higher the send level the more reverb effect return you’ll hear for that track.

3) Reverb, used correctly, on a mixer’s bus, sounds superior in both clarity and depth than individual reverb plug-ins inserted on each mixer channel. When it comes down to it, that’s really what matters, it just sounds better.

In most modern DAW mixers you can send to a reverb in either mono or stereo. I prefer a mono send and a stereo return for most projects (such as dance, hip-hop, and alternative). However, a stereo send and a stereo return is equally acceptable and I often use this configuration when I want to achieve a higher degree of separation and fidelity (for example, a sparse acoustic recording or string quartet). Pictured below is a mono send/stereo return reverb configuration for Pro Tools (notice my choice of the “mono/stereo” version of the D-Verb reverb plug-in). I’m also attaching a Zip of a Pro Tools 7.3 session so you can dig around and see first-hand how it’s all set up.

Reverb 4 U Pro Tools Session File