Instruments & Gear

Stephen Wise is your guide to home recording.

Another workhorse module in the recording studio is the compressor. It can be the most essential tool or the worst evil incarnate, depending on your point of view. Debates over compression can take on a religious fervor.

“Purists” despise compressors and compression may be totally absent from classical and other acoustic recordings. Today’s pop music, on the other hand, employs multiple levels of compression. If it sounds incredibly loud, then it’s probably compressed.

The compressor has an extra level of complexity above that of the effects that we’ve used thus far. It’s a little harder to set up and sometimes you just can’t get it quite right. Compressors also introduce audible artifacts that the trained ear can identify. A compressor has two main components: a variable amplifier and a level detector.

Before compressors, humans did the compressing! You’ve probably done it yourself. You listen to some music. When you hit a passage that is a bit too loud, you turn down the volume. When the loud section has passed, you turn the volume back up again. A compressor does exactly the same thing – automatically! A compressor can also work quickly and tirelessly, never missing anything.

The Audacity® compressor can be found under the “Effect” menu. Select “Compressor…” A basic compressor has four controls that let you determine how the compressor will behave: Threshold, Compression Ratio, Attack Time, and Release Time. For reasons unknown, the basic Audacity® compressor is missing a release time setting – a rather unfortunate situation.

“Threshold” lets you set the loudness at which the compressor will start to turn down the volume. Compression Ratio tells the compressor how much you will be turning down the volume. A ration of 2:1 (two to one) means that the compressor will be turning the volume down to one half of what it should be. Some compressors will let you set the ratio all of the way to “infinity”. At this point, you have a device known as a “peak limiter.” Attack Time tells the compressor how quickly to turn down the volume. Attack Time is very important and usually takes some tweaking before you get a smooth transition. Release Time, if you have it, is usually set longer than Attack Time. You want the volume to come back up slowly in classic compression.

Now you have it. What can you do with it? Suppose that you have a recording that has occasional loud sections. If you set the volume to a comfortable level, then the loud sections really hurt or cause distortion. You can set the volume lower, but then the quiet sections are too quiet. If you’re listening in your car, then you can’t hear the quiet sections over the road noise. The compressor can fix it at the expense of giving up some of the natural dynamics of the music. Everything that you hear on the radio is compressed and peak limited to keep the sound from getting lost in the static and to keep you from moving on to a “clearer” station.

If you compress a track or a mix, then boost it up to to maximum level, it sounds louder! That’s why commercials sound so loud even though the stations insist that they’re not any louder than the programming. The speaking voice can be compressed a lot. When the parts that should be soft are loud too, it just sounds loud, plus, you can hear every word over the noises in the kitchen! The “Normalize” check box on the Audacity® compressor does it for you automatically. The compressor is the main weapon in the “loudness wars”.

You may have noticed that today’s music sounds a lot louder than your old albums from the ’60s and ’70s. In fact, the digital remixes of those old albums sound louder too. That’s because today, “louder” means “better.” If an mp3 can only hold so many bits, make them all loud ones! Compress each track, then compress the final mix – more than once if necessary!

Although I dislike this approach to remixing old albums (it changes the character in a fundamental way), I have no objection to using compression as a creative tool. It’s a part of today’s sound just as “slapback echo” was a part of the Rockabilly sound.

Compressors all introduce a “pumping effect” into the sound. “Pumping” is descriptive of the effect of the volume being turned up and down. Play some music while rapidly turning the volume control up and down and you will hear the effect. Normally, you want to hide pumping as much as possible, but it can also be used as a creative effect. As an example, if you set up the compressor just so, you can get everything to “pulsate” with the drum beats, adding extra “drive” to the sound.

Since I have completed this blog, Leah has inadvertently done me a great service by posting the article “It’s Not That You’re Older, the Music Really is Louder,” saving me a lot of wind. Here is another great link on the subject: “What Happened to Dynamic Range?”

Stephen Wise has been designing electronic musical instruments since 1975. Steve specializes in realistic recreations of traditional instruments. He became interested in the field after hearing Walter/Wendy Carlos’ “Switched On Bach” and upon being introduced to the Allen Digital Computer Organ, the world’s first all digital musical instrument. Steve is currently designing instruments for Schulmerich Bells, maker of handbells, electronic carillons, and the breakout MelodyWave® instrument.

Singer-songwriter Jonathan Butler may record one or two takes of a song for an album; others do several takes and combine them. Butler's sound engineer Paul Klingberg talks about his approach to recording Butler and others.

This post first appeared in a longer form in the Disc Makers blog.

In this post we take a look at the techniques used to create composite lead vocal tracks, referred to as “comping the lead vocal” by studio engineers. After a brief overview of the technique, we’ll speak with veteran engineer Paul Klingberg, who has recorded vocals with a wide range of artists including Earth, Wind & Fire, Jonathan Butler, Loreena McKennitt, Cheap Trick, Brian McKnight, The Simpsons, James Ingram and many others.

In an ideal scenario, you or your lead vocalist would nail the perfect studio performance of you new song in one continuous take.  However, once you put that vocal performance under the sonic microscope of the recording process, you’ll undoubtedly hear some elements of the lead vocal that could be improved. Maybe the phrasing is a bit rushed in one part, or a particularly long sustained note tends to lose pitch, or some other problem becomes apparent.

Rather than singing the track over and over from top to bottom, doing a number of solid takes of the song on separate tracks, then listening to and selecting the best parts of each take will eventually result in what will become a composite lead vocal track. When done correctly, this technique will give the illusion of a single, seamless performance when placed into the final mix of the song. Often the artist will also use the separate takes to experiment by slightly modifying their level of intensity, vocal placement, rhythms, etc.

Before you start recording
Before we get to the nitty-gritty of how to best put together comped vocal tracks, it’s important to note that the choice of vocal mic, mic preamp and recording set-up make a significant impact on the sound of your lead vocal tracks. As a general rule, large diaphragm condenser mics make excellent vocal mics, since they generally are more sensitive to the shadings and nuances that the human voice is capable of producing.

For the home studio, there are a number of excellent mics well suited to recording vocals in the under $500 price range. If you’re just starting out in home recording, you probably won’t be ready to invest in a separate mic preamp to enhance the sound of your vocal recordings. As your recording chops progress and your ears become more fine-tuned to the subtleties of recording, you’re likely going to want to invest in a high quality mic pre that will further enhance the warmth and richness of your vocal recordings.

It’s important to note that each specific mic and preamp will have its own sonic signature (or lack thereof) so when you’re ready to start shopping for either, it’s essential that you get yourself to a pro audio dealer and listen to the various models in your price range to see which ones best complement your voice. Many pro audio dealers will also let you try out a piece of gear at your home studio if you are a regular customer, which is the absolute best way to insure that any new gear will do what you expect it to at your own home studio.

Personally, I prefer to record vocals flat (meaning no EQ applied during recording) and with no compression or limiting during the tracking. If I want a brighter sound, I’ll try switching to a different mic which I know emphasizes the highs more than the first mic I tried. Once you have a sound, then it’s time to focus on helping guide the vocalist to getting that perfect take – compositely speaking, that is.

Many tracks end up being carved up down to the syllable or vowel level, which can be done quite readily using any of the popular digital recording programs available. How small you go depends on your level of attention, your patience, and what you hope to achieve with the finished track. However, it’s not uncommon for the lead vocal track of a pop song to include hundreds of sections, as well as a great deal of riding the gain of the vocal track (more on this in the next section.)

Sound engineer Paul Klingberg.

Vocal comping with Paul Klingberg
For many years now, Paul has been the go-to guy for the amazing vocal sound of Earth, Wind & Fire collectively, as well as solo albums by Maurice White and Philip Bailey. Among Paul’s latest projects was recording and mixing Jonathan Butler’s “So Strong,” released on Mack Avenue Records.

You’ve been at this for a while now. What’s changed about the way you approach vocals and composite tracking?
Essentially, vocal comping hasn’t really changed in practice since the ‘70s and ‘80s when we would use different tracks on a 16- or 24-track tape to record different sections of a vocal performance and then create a composite track, usually by bouncing the sections we wanted over to a new, final lead vocal track. In fact, back in the day, I had a friend of mine build me my own “comp box” which was essentially a switcher with two rows of eight buttons each and a cable harness to interface it with a studio patch bay. It had a fader right in the middle which allowed me to switch seamlessly between the vocal tracks using a little analog crossfade to smooth out the transition. That box made me a pretty popular engineer among studio vocalists at that time. I used it regularly on my early work with Earth, Wind & Fire.

The practice really isn’t any different today, except of course that with DAW’s like Pro Tools, Logic, etc., we have a graphical interface to see the waveforms as we edit and lots of crossfading options to arrive at a seamless vocal track. Although I primarily work in Pro Tools, it’s a good idea to develop a basic understanding of the various platforms so you can work with whatever [program] the artist is using on a particular record.

For instance, Jonathan Butler’s latest album was done primarily at his home studio and he works in Logic. Jonathan is a very talented singer, so he would sing it straight down, rarely doing more than two takes of any song. We’d just use the best portion of each take to make up the final vocal. With a group like Earth, Wind & Fire, on the other hand, with multiple takes, we’d end up with a big matrix of vocal performances in Pro Tools, allowing nearly endless possibilities for the final vocal track.

Personally, I’d rather not copy and paste the lead vocal track together from any number of earlier takes. Instead, what I prefer to do is record the best sections together onto a new master composite lead vocal track. The producer and I will listen to each section of the song and decide which of the multiple vocal takes offers the best performance. Today, most everyone is using some type of pitch correction software, so I assign that plug-in on each vocal track as an insert. As I get ready to record that section of the vocal to my newly created composite track, I can make whatever tuning adjustments might be needed.

For example, assume I have three complete takes of a lead vocal on three separate tracks, each one a little different. Working in the graphic mode of the pitch correction software, I’ll assign the output of each track to bus 1; then open up a fourth track to record all the sections I want to include in the final vocal recording. The input on this new fourth track will be bus 1’s output. This way, I’m really doing two things at once: I’m making the choices on which individual sections I want to use and I’m fine-tuning them with the pitch correction software. Once I have the sound just right, I’ll print it to the new track, then move on to the next section and repeat the process. This method really helps ensure proper intonation and better continuity, meaning the final composite sounds like a single performance.

When I’m done, I listen critically and confirm that the new lead vocal is the best composite performance given the performances available. That’s what it’s all about with regard to the process; I want to end up with one continuous track from top to bottom.

What pitch correction software do you prefer?
I’ve been using the Antares Auto Tune program for years. It comes in a number of different versions and is a very powerful and flexible solution to tuning problems.

What advice would you give to new artists or those new to recording at home doing vocals?
When I’m mixing an album for a new artist, I often find that there are not enough takes to get the job done. Of course, it depends on how savvy they are about how technology can help them capture a good vocal performance. When you are faced with a tough passage, be sure to do at least three or four takes and then assemble a comp of that section to see if it works as well as you want it.

Remember, more takes equals more options when it comes to creating the master composite vocal track. You can go overboard and have too many choices, but since you have the luxury of time when you’re working at home, do enough takes so you can try out your own composite and be confident it will work.

The other thing is that there is no substitute for working with an experienced audio engineer. Some people are willing to experiment and spend the many, many hours necessary to learn how to get good results recording at home. However, I’ve often been called in to mix an album and find that what has been recorded is just horrible. When your album is playing, no one knows how or where it was recorded, they only judge the recording by its overall quality.

Everything from the choice of the vocal mic and preamp to the space around the vocal (room ambience) becomes part of your sound. Unfortunately, often musicians may stumble trying to simultaneously perfect their music and learn the art of recording. In a case like that, consider working with an experienced engineer at your home studio, as it’s sure to result in a better sounding record. You really do want to end up with an album that will be competitive with the sound of what everyone else is listening to at any given time. Using the proper equipment and when you can, a professional studio, will make an audible difference to anyone listening to your record.

Back to vocal comping, what do you listen for when critiquing the master composite track you’ve just created?
It’s the subtle things that make a difference. I try to put myself in the shoes of a listener hearing the song for the very first time. Especially the dynamics, from loud to soft. And making sure there is midrange clarity so that every little nuance can be heard. I really perfected the techniques I use by working closely with Maurice White (of Earth, Wind & Fire). He would focus on the little trails, the endings of a word or a phrase and have me ride up the volume to wring every last bit of emotion out of the artist’s delivery. I may do it by bringing up the fader, or using some EQ to emphasize a particular frequency, or pump up the vocal with a compressor… whatever it takes to have exceptional clarity. Maurice taught me that really bringing out all these subtleties has an emotional consequence, it makes a real difference.

This post, by Keith Hatschek, first appeared in the Disc Makers blog. Hatschek has been a musician, educator, recording engineer, producer and marketing executive. He is the author of  The Golden Moment: Recording Secrets from the Pros and How to Get a Job in the Music Industry. Hatschek currently teaches Music Management at the University of the Pacific.

Stephen Wise is your guide to home recording.

Audacity® supplies a handy tool for analyzing sounds called a Spectrum Analyzer. What the Spectrum Analyzer does is simple, but how it does it is not. In fact, enough scholarly papers have been written on the subject to fill a small town library.

In the previous post, I discussed the use of equalization. Sometimes, applying equalization doesn’t have the desired effect or seems to have no effect at all. Often, the problem is that you are trying to act on something that isn’t there or, at least, is not where you thought that it was.

When you look at a track in Audacity®, you are seeing a time domain representation of a sound – that is, the amplitude of the sound over a period of time. Through the use of a mathematical process known as Fast Fourier Transform, the time domain can be converted into the frequency domain. The useful thing about the frequency domain is that it lets you see a representation of the individual frequency components that make up a sound. Now you know where to place your filters!

If you are looking to replicate a certain sound, it can be useful to compare their individual spectra, applying filters or adding components until you have a match. The Spectrum Analyzer can also help you locate extraneous sounds for removal.

In Figure 1, I have taken the spectrum of a “sawtooth” wave created by the Audacity® “Generate” function. A sawtooth is one of the “geometric” waveshapes common to the original analog synthesizers. It’s a bright, buzzy sound and has the characteristic spectrum shown. All of the partials are present out to infinity and the levels of the partials decrease with increasing frequency.

By contrast, Figure 2 is a heavily low-pass-filtered square wave, another common analog synthesizer tone. You can see by the spectrum that the partials are lower in amplitude and that the even numbered ones are attenuated. A spectrum such as this one belongs to a bright but hollow sounding flute.

The controls on the Spectrum Analyzer no doubt seem cryptic. The controls are there because spectrum analysis is somewhat imprecise math. How you accomplish the Fast Fourier Transform (FFT) depends on what you want to see and what your source material is. The first box selects “Spectrum” along with some other functions for which I have yet to find a use.

To the right of that is drop-down list of numbers, each double the number above. This number is the number of sample points that will be used in the analysis. More points gives you a better picture but, you also get more averaging. More points is great for a steady tone such as an organ, but useless for something such as a drum hit.

Since a drum hit varies tremendously over time, you want to analyze small pieces of the whole and you can’t do that with a large number of sample points. You could, however, re-sample the drum to a much higher sample rate first.

The lower left window is perhaps the most cryptic. Since you are chopping out a chunk of sound to analyze, the actual sample points included in the chunk are somewhat arbitrary. If you use a “Rectangular” window, then you are just analyzing the raw, chopped out chunk. The abrupt “edges” on this chunk cause a lot of errors, so mathematicians have devised means to minimize the error.

Not surprisingly, most of these “window” functions have been named after mathematicians. If you really want to know the hows and whys of these window functions, there are plenty of papers on the web. My advice is to use what works best for you. Sometimes, the rectangle is best. Of the windows available in Audacity®, I prefer the “Hanning” (actually Hann) window as it generally gives me the most useful result.

The final window gives you a choice of linear or log scale. Always use log scales for audio. Log handles the wide ranges a lot better. You can get precise numbers for amplitude and frequency by moving the mouse over the graph. If you do the “mouse over” with the different window functions, you will see how the results vary from one function to the next. Note that in the case of amplitude (dB), it is the relative amplitude that is important, not the actual amplitude.

The Spectrum Analyzer is probably more useful for music synthesis. It is quite invaluable for creating electronic instrument sounds, but it also has uses in creating a clear audio mix. When mixing many instruments, some of the instruments can become “lost in the mix.”

You can often “find” these instruments by determining which other tracks are occupying the same frequency space and filtering those tracks to create a “hole” for the lost instrument to fit into. The ear/brain happily fills in these holes and also hears the formerly lost instrument. This trick works well with the snare drum, allowing you to keep the level of this drum high without its beats obliterating everything else. Filter the snare so that comes through the holes where the other instruments are not.

Stephen Wise has been designing electronic musical instruments since 1975. Steve specializes in realistic recreations of traditional instruments. He became interested in the field after hearing Walter/Wendy Carlos’ “Switched On Bach” and upon being introduced to the Allen Digital Computer Organ, the world’s first all digital musical instrument. Steve is currently designing instruments for Schulmerich Bells, maker of handbells, electronic carillons, and the breakout MelodyWave® instrument.