Instruments

An electric guitar, a bass, a synthesizer, and drum machines are typical instruments that plug in to the interface and represent most of the input devices that you use in your studio. A synthesizer and drum machine can plug directly into the Line In inputs of your interface, whereas an electric guitar or a bass needs a direct box (or its equivalent) to plug in to first. (In the case of a Avid interface, you need to use one of the inputs that has a preamp.)

A direct box is an intermediary device that allows you to plug your guitar directly into a mixer without going through your amp first. (For more on direct boxes, see the upcoming section, “Deciphering direct boxes.”) Check out Figure 1-1 for an example of an instrument-input device.

Microphone

You use a microphone (mic) to record the sound of a voice or a purely acoustic instrument — sound sources that, last time I checked, can’t be plugged directly into the interface. A microphone converts sound waves into electrical energy that can be understood by the interface. I detail the several types of microphones in Book 3, Chapter 2. Check out Figure 1-2 for a look at a microphone.

Sound modules

Sound modules are special kinds of synthesizers and/or drum machines. What makes a sound module different from a regular synthesizer or drum machine is that these contain no triggers or keys that you can play. Instead, sound modules are controlled externally by another synthesizer’s keyboard or by a Musical Instrument Digital Interface (MIDI) controller (a specialized box designed to control MIDI instruments). Sound modules have MIDI ports (MIDI jacks) to enable you to connect them to other equipment.

Often sound modules are rack-mountable, meaning they have screw holes and mounting ears so you can put them into an audio component rack. Some controllers, however, are not rack-mountable; Figure 1-3, for example, shows a drum module that rests on a stand or tabletop.

Solid-state

Solid-state preamps use transistors to boost the level of the microphone or instrument. Top-quality (expensive) solid-state preamps are generally designed to produce a sound that’s clear and accurate (George Massenburg Labs and Crane Song brands, for instance). Solid-state preamps can also be designed to add a pleasing distortion to the music (Neve and Neve-clone preamps, for example). Many recording professionals prefer the clear and accurate sound of a solid-state preamp for acoustic or classical music or any situation where capturing a very natural sound is important. The preamps in nearly all audio interfaces are solid-state. Although not as high in quality as many of the more expensive external preamps, they are certainly serviceable for most purposes and allow you to create top-quality music when you use them correctly. (I show you how to do this in Book 3, Chapter 2.)

Tube

Since the beginning of the digital recording revolution, professionals have complained about the harshness of digital recording. As a result, many digital-recording pros prefer classic tube preamps because they can add warmth to the recording. This “warmth” is actually distortion, albeit a pleasing one. All-tube preamps are generally very expensive, but they’re highly sought after among digital recording aficionados because of their sound. Tube preamps work well with music when you want to add color to the sound (that is, not produce an accurate representation of the original source sound). No wonder they show up a lot in rock and blues — and they’re great for recording drums. You can also find tube preamps that are clean and open — much like the high-end solid-state preamps that I describe earlier — such as those made by Manley Labs.

Hybrid

A hybrid preamp contains both solid-state and tube components. Most of the inexpensive tube preamps that you find in the marketplace are actually hybrids. (These are also called starved-plate designs because the tubes don’t run the same level of voltage as the expensive tube designs.) These types of preamps are usually designed to add the classic tube warmth to your instrument’s sound. How much the sound is colored by the tubes — and how pleasing that colored sound is to the listener’s ears — will depend upon the quality of the preamp. Most hybrid preamps allow you to dial in the amount of character (pleasing distortion) that you want.

Your audio interface will come with a limited number of preamps, usually two or four. If you want to plug in more mics than the number of preamps you have — or if you want to be able to produce different sounds from your preamps — you need to buy one or more external preamps, such as the one shown in Figure 1-4.

Sampling rate

Sampling rate refers to the frequency at which the recorder samples the incoming sound source. When a recorder samples a sound, it actually takes a small snapshot of the audio signal. Typical sample rates for digital recorders are 44.1 kHz (the sample rate for audio CDs), 48 kHz, 88.2 kHz, 96 kHz (the sampling rate for DVD audio), 176.4 kHz and 192 kHz. The higher the number, the more samples are taken each second, and the closer the recorded sound is to the original.

The more times per second that a digital recorder samples the sound of the incoming signal, the more of the original sound it includes (which results in, among other things, a bigger sound file that takes up more space on your computer). These numbers are described in kHz (kilohertz) — that is, as taking place thousands of times per second. For example, when you record at 48 kHz, the sound is sampled 48,000 times every second. This sounds like an awful lot, but accurate sound requires lots of samples.

Bit depth

When you start looking at digital recorders, you’ll hear jargon such as 16-bit (the bit depth of audio CDs), 20-bit, 24-bit, and so forth. This is the bit depth, which is described in terms of bits. (A bit, short for BInary digiT, is the basic unit of information in the binary numbering system used by computers.) The numbers 16 and 24 relate to the amount of digital information that can be contained in each of those sample rates that I describe in the previous section, “Sampling rate.” The higher the number, the more accurately the sound is represented; most professional audio gear now records at a 24-bit resolution. With that higher number, however, the sound takes up more of your digital storage space. This usually isn’t a big deal, though, considering the low cost of huge hard drives.

A/D and D/A converters

A third variable that affects how good a digital recorder will sound is the type of analog-to-digital (A/D) and digital-to-analog (D/A) converters that the recorder uses. The A/D and D/A converters are what actually transform the signal from an analog sound source into digital information and back again. They do this by taking small snapshots of the incoming signal (at whatever sampling rate the converter uses) and applying a number to that sample (based upon the bit depth).

For example, suppose you have a 24-bit converter with a 48-kHz sampling rate. When it senses an auditory signal — a vocal perhaps — the A/D converter takes a snapshot (measurement) of that signal 48,000 times per second (48 kHz). Each of these snapshots is given a value between –8,395,008 and +8,395,008 (24-bit resolution has 16,790,016 possible levels), which puts the vocal sound somewhere on a chart corresponding to a waveform shape. The recorder in turn reads these numbers, and the D/A converter translates these numbers back into an analog waveform again. (Pshew!)

Converting an analog signal into digital data and back again is a highly technical process. You can spend days or even weeks reading about the intricacies of audio conversion. (Check out Principles of Digital Audio by Ken C. Pohlmann, published by McGraw-Hill Professional; or The Art of Digital Audio by John Watkinson, published by Focal Press, if you’re into this techy stuff.) The most important thing to remember when considering A/D and D/A converters is how well you like the sound. If the converter sounds good to you, it’s a good converter.

You can buy separate A/D and D/A converters that can sound better than the one included in your audio interface. Depending on your ears, your engineering skills, and the style of your music, the improvement in sound of external converters can range from barely noticeable to mind-blowing. Before you spend any money on external converters, though, I highly recommend working on your skills as a songwriter, musician, and engineer.

Getting the sound in and out

After you have a computer with enough speed and muscle (see the previous section), you need the appropriate hardware to get the sound in and out of it. Traditionally (if there can be traditions in such a new technology), you needed a sound card — also called a PCIe card because it fits in the PCI (Peripheral Component Interface Express) slot in your computer. You also needed an audio interface, which allowed you to get the sound from your mixer or preamp to the sound card. This is not the case anymore.

Currently, audio can be brought in and out of a computer in several ways:

• Through a PCI card connected to your computer’s PCI slot: PCI slots use a sound card that installs in your computer and may or may not have a separate box that houses the converters, inputs, and outputs. Originally the standard for high-end audio interfaces, few computers offer PCI slots nowadays. However, if you’re old school and you have a computer with PCI slots, you can find some great options.
• Through an interface connected to the USB port: There are a ton of audio interfaces that employ USB connectivity. USB has proven to be a robust format, and USB 2 and 3standardshave transfer rates that are fast enough to handle all the data you need to run multiple inputs and outputs. Avid’s only current audio interface, the Eleven Rack (see Book 2, Chapter 2) employs USB 2 connectivity. USB is a good choice, but avoid interfaces that use the older USB 1.1 standard for recording.

• Through the Thunderbolt port: Thunderbolt is a versatile and super-fast protocol that allows for the transfer of both data and power at speeds beyond those available with USB. This is a relatively new connector type but has gone through a couple of iterations over the past few years. This ordinarily wouldn’t be a big deal. For instance, USB has grown from 1.1, to 2.0 to 3.0 to 3.1 and is largely backward compatible. This means if you have a computer with USB 3 capability, you can use a USB 2 device without any hassle or special adapters.

  Thunderbolt is a different story. Thunderbolt 2 and 3 have nothing in common except a name, so if you have a computer with a Thunderbolt 3 port, you won’t be able to use your Thunderbolt 2 device without a special adapter. Likewise, a Thunderbolt 3 device will not work with your Thunderbolt 2–equipped computer. With this in mind, you’ll find both types of Thunderbolt audio interfaces. Which to choose obviously depends on the port your computer has. Most Thunderbolt 2–equipped audio interfaces also have USB ports, so they’ll work if you don’t have a Thunderbolt 2 port on you computer — as long as you have USB.

  All that said, Thunderbolt audio interfaces work great, especially if you want to chain a bunch together to get large numbers of input and outputs (for recording bands live, for example).

Software

Because you’re reading this book, I’m going to assume that your main choice in recording software is Pro Tools. Three lines of Pro Tools Software are available:

• Pro Tools First: This is Avid’s free entry-level program that offers a limited number of tracks (four simultaneous recording and 16 total audio tracks). This is an inexpensive way to get into Pro Tools; it works well for artists recording their own music.
• Pro Tools: Pro Tools is Avid’s host-based program (which means that Pro Tools relies on the power of your computer to run the program as intended), with 32 simultaneous and 128 total audio tracks. This version of Pro Tools is a full-featured program that can do anything you’re likely to want to do with your own (or others’) music.
• Pro Tools | Ultimate: This is the native solution that requires you to buy Avid HD hardware to provide the processing power to run Pro Tools. This version of the software also offers some extra functionality beyond what you get with the host-based options, which some pros may find useful, such as a maximum of 256 simultaneous audio tracks, 512 auxiliary tracks, and 64 video tracks (as opposed to just one for Pro Tools and none for Pro Tools First). If you choose Avid’s HDX hardware, Pro Tools expands your total simultaneous audio tracks to a maximum of 768.

Down the road, you may want to add some other software to your system. You can include (for example) programs such as Acid (for creating loops) or Logic (for more MIDI-intensive work). You can find an online discussion board on the Internet for each of the major audio-recording software programs. Before you buy, go to the sites of the systems that interest you and see what people say about the programs. Ask questions, explore the issues that the people are having with the programs, and look for comments that reflect how well the programs interact with Pro Tools and your Avid hardware. Doing so can save you lots of time dealing with bugs in your system, allowing you to record a lot more music. You can find these sites by using the product name as the keyword in your favorite search engine, or by checking out these Internet forums:

• www.gearslutz.com/board
• www.homerecording.com
• www.prosoundweb.com
• www.recording.org

Graphic EQ

Use graphic equalizers to choose a specific frequency to increase or decrease by a certain amount, generally up to 6 or 12 decibels (dB). Doing so enables you to eliminate an offending frequency from the signal or make other adjustments to the tonal quality of the source signal. The graphic EQ has a certain number of frequency bands that you can adjust. You’re limited to only those frequency bands that your EQ has available. Figure 1-7 shows a typical graphic EQ.

Shelf EQ

A shelf equalizer affects a range of frequencies above or below the target frequency. Shelf EQs are generally used to roll off the top or bottom end of the frequency spectrum. For example, you can set a shelf EQ to roll off the frequencies below 250 hertz (Hz) in order to reduce the amount of rumble (low-frequency noise) in a recording. You can see how this looks in Figure 1-8. Notice how the shelf EQ gradually reduces the amount of energy (sound) below the set point and then levels off — hence the shelf in its name.

Parametric EQ

The parametric equalizer enables you to choose the specific frequency that you want to affect as well as the range of frequencies to put around that frequency. With a parametric EQ, you dial in the frequency that you want to change, and then you set the range (referred to as the Q) — that is, the number of octaves that the EQ will affect. Check out Figure 1-9: The two diagrams show how the Q relates to the range of frequencies affected. (The higher the Q setting, the narrower the band of frequencies affected.)

The parametric EQ is one of your most useful tools when mixing all your individual tracks into a stereo pair. I describe this tool in detail in Book 6, Chapter 3.

Compressors/limiters

A compressor’s job is not only to limit how loud a note can be, but also to reduce the difference between the loudest and softest note (that is, compress the dynamic range) of the sound being affected. The purpose of the compressor is to eliminate transients (unusually loud notes) that can create clipping (digital distortion).

Compressors are used extensively on modern recordings; the device keeps transients at bay by gently reducing the highest level that goes through it. Compressors are also used in mastering to raise the overall volume of a song without creating distortion. The device does this by reducing the overall dynamic range; as a result, a compressor effectively raises the volume of the softer notes without allowing the louder notes to be too loud and distorted.

A limiter works much like the compressor except it severely limits the highest level of a sound source. The limiter is basically a compressor on steroids; it gives you beefed-up control over volume. Any signal above a certain level, called the threshold, gets chopped off instead of compressed (as it would be with the compressor). A limiter is a good choice in extreme situations, such as when you want a really in-your-face, snare drum sound. In this instance, the limiter essentially eliminates any dynamic variation from the drummer’s snare-drum hits and makes them all as loud as possible.

Gates

A gate is basically the opposite of the limiter. Rather than limiting how loud a note can get, the gate limits how soft a note can get. The gate filters out any sound below a certain setting (the threshold) while allowing any note above that threshold to be heard.

Gates are often used on drums to keep unwanted sounds from the cymbals from bleeding through to the tom-tom or snare drum mics, or on guitars by refusing to allow the noise generated by guitar effects to be heard when the instrument isn’t playing.

Expander

An expander is basically the opposite of a compressor — instead of attenuating (reducing the volume of) the loudest notes in a performance, an expander attenuates the softest notes. For example, if you have a singer whose breath you can hear in the mic and you want to get rid of that particular blemish, just set the expander to go on at a level just above the annoying breath sounds to subtly drop the offending noise.

Reverb

Reverb is undoubtedly the most commonly used effects processor. With reverb, you can make any instrument sound as if it were recorded in almost any environment. Reverb, a natural part of every sound, is the result of the sound bouncing around inside a room. The larger the room, the more pronounced the reverb. The purpose of a reverb in audio production is to make an instrument sound more natural (especially because most instruments are recorded in small, nonreverberant rooms) or to add a special effect. Reverb can make almost any recorded instrument sound better — if used correctly.

Delay

Think of delays as a recording studio’s version of an echo. The delay can be set to happen immediately after the original sound or be delayed much longer. Delay can sound natural or be used as a spacey special effect. You can have a single echo or multiple delays (very common with the snare drum in reggae music, for instance). Delays are commonly used on vocals and guitar, although you can hear them on just about any instrument, depending on the style of music.

Chorus

A chorus effect can make one instrument sound like several. Chorus effects add very slightly off-tune versions of the unaffected sound, which results in a fuller sound. You find chorus effects used on vocals, guitars, and lots of other melodic instruments.

Flanger

A flanger (pronounced flanj-er) effect is similar to a chorus effect in sound except that the flanger gets its sound from delaying part of the affected sound in relation to the original, rather than altering its pitch. Flangers are sometimes used on background vocals and solo instruments to add an interesting texture. This is a unique sound that you recognize almost immediately upon hearing it.

The flanger effect comes from the early days in recording. You can create the flanger effect the old-fashioned way by using a two-track tape recorder to record a duplicate track of the one you want to flange. You then play the two identical parts back at the same time, gently pressing against the edge of the two-track tape (the one with the duplicate part) while it’s running. This delays certain parts of the sound slightly and drastically changes the character of the instrument.

Nowadays, you can just choose the flanger patch (sound) on your effects processor to get this sound. Isn’t technology great?

Pitch correction

Pitch correction, like its name suggests, is used to correct an out-of-tune note. You can use this effect to help a singer or an instrument player sound better by fixing those notes that are slightly out of tune. Pitch correction (also called auto-tune) has gotten a bad rap lately (mainly from its overuse and potential for abuse with a singer who can’t sing on-key). When used sparingly and appropriately, pitch correction can make an otherwise-decent vocal performance really shine. Auto-tune can also be used to create some interesting effects, such as that robotic-vocal sound you hear on so many of the pop songs on the radio nowadays. The most easily distinguished example is the lead vocal on Cher’s “Believe.”

WAV and AIFF

WAV (Microsoft Wave format) and AIFF (Apple; Audio Interchange File Format) files are the formats for audio files found with most professional audio software. The advantage to saving your music to WAV or AIFF is that when you hand over a CD containing your WAV and/or AIFF files to a mastering house, the recorded sound is actually in a higher-quality format than that of the finished CD (provided that your recorder records in 24-bit, which most do). You can also take your music files to any other studio that supports these file formats and work with them there.

MIDI

A MIDI file is not an audio file; rather, it’s a data file that contains MIDI information that can be transferred from one computer to another; it tells the computer how to use a sound module to construct the music. An advantage to MIDI files is that they take up less room than an audio file. The disadvantage is that they contain only the MIDI information and no sound. To play a MIDI file, you have to have a sound module. And the sound you get from the file depends entirely on what sound source you use.

MP3

MP3 is a file format that has become quite popular on the Internet. Its advantage over audio CDs and other computer files is that it takes up less room. Its disadvantages are that the data is compressed and the sound quality is not nearly that of commercial CDs (contrary to what MP3 proponents claim). You can find out more about MP3s and audio quality in Book 8, Chapter 2.

AAC

AAC (Advanced Audio Coding) is the file format of choice for iTunes music and some other Internet music sites. This format is said to offer a higher quality of sound than MP3 for its size.