* Studio Vocabulary
I can't tell you how many times I've gotten bogged down with a client over our choice of words. Discussing music and musical ideas is a lot like trying to dance "architecture". At some point words fail us. I've witnessed musicians who literally start speaking in tongues because they're not sure how to translate their musical ideas or thoughts into words. Maybe this will help.
Track, tracks, tracking - Simply put, a "track" is a contiguous line of magnetic impulses printed onto a reel of magnetic tape. The number of tracks that you can record onto a reel of magnetic tape depends primarily on the design of the tape deck and the width of the tape. The most common widths for analog reel-reel tape are 1/4", 1/2", 1" and 2". The standard cassette tape that you find in many home/car stereo systems can accommodate up to 4 tracks of audio (two tracks in one direction, two in the other). However, Tascam & Yamaha and a number of other companies were making these "porta-studio" units that could record up to 8 tracks of audio on standard cassettes. Alesis used to make digital audio recording decks called ADAT's that could record eight tracks of audio on standard VHS video tape. Tascam, not to be outdone by Alesis, developed a competing product that used the Hi-8 mm format video tapes.
And least we forget, back in the 70's, there was something called the "8 track". The 8 track was a plastic cartridge containing a single continuous loop of 1/4" tape. It was originally designed for car use but eventually found it's way into the home stereo market. Albums were released on both vinyl and 8 track formats. The problem with that old 8 track technology was that you were lucky to get more than a half dozen or so "plays" out of it before the tape would self-destruct. You can read all about the 8 track and it's history here. I was amazed to find out it was designed by guys from the Lear Jet company.
You can still find analog reel-reel decks in many of the bigger commercial recording studios that look sorta like this:
These refrigerator-size machines are capable of recording up to 24 simultaneous tracks of audio on 2" wide tape. The machines can be connected together to permit even more simultaneous tracking capability. These commercial decks are big, noisy and difficult to maintain. As a result, many studios installed these things in special sound-proof machine rooms located just outside of the main control booth. Which, of course, makes operating the machines a tad inconvenient. Usually, someone (called a tape op) has to baby-sit the machines during the course of a session.
One of the guys we have to thank for all this multi-track technology is a fellow by the name of Les Paul (yes, that Les Paul). Les Paul pioneered and championed the use of multi-track recording technology here in the United States. A little known fact about a guy who's name is synonymous with a certain brand of electric guitar.
The track concept technically doesn't apply to hard drive recording technology because computers store data on hard drives in a non-contiguous format (commonly called "files"). But we use the term anyways because it's easier to understand. Most hard drive recording programs render audio waveforms as individual tracks along a specific timeline. They're not really tracks in the old-school sense - they're audio files which are stored on the hard drive and displayed on your computer monitor as an analog waveform.
Tracking is the process of recording tracks.
Overdub, dubbing, dub - An overdub is a track (or tracks) of audio that is recorded "in addition to" another track (or tracks) of audio. For example, you might record an acoustic guitar track and then go back and overdub a vocal track. Now you have two tracks of audio that have to be mixed together. Maybe later you decide you want to add drums and bass and keyboards and all kinds of other stuff to your arrangement. So you go back and overdub all those elements. It's simply adding additional tracks to what's already there. The words, "dub" and "dubbing" are sometimes used to describe the process of copying audio from one format to another (like from audio cassette to CD-R or DAT).
Punch-in - this is an "old-school" recording term which refers to the process of recording a short section of audio "over" a previously recorded part. Say, for example, that you recorded a four minute bass guitar track that was perfect except for a few wrong notes here and there. You could go back and record a brand new bass guitar track or you could simply replace those few bad notes. Replacing a few bad notes takes less time than recording a whole new track which may, or may not, introduce more bad notes. In reference to analog tape, this is a destructive process because you are erasing what was there before and recording something new in it's place. Punching in on analog tape is always a tricky situation because you want to make sure that the punch-in start and stop times are perfect otherwise you might accidentally record over something that you don't want to replace. With digital audio programs, a punch-in is non-destructive which means you can easily revert back to whatever was there originally if you make a mistake. You don't lose the original audio. It's always there unless you were to permanently delete it for some reason.
Click Track - A click track is a single track of audio which contains some sort of simple rhythmic clicking or ticking sound (like a metronome) that matches the tempo and timing for a musical performance. Once the click track has been recorded, it becomes the timing reference for the entire song. The musicians listen to the click track in their headphones while they are performing their parts. This keeps everyone in time, even the drummer. The most common type of click sound is the sound of two drum sticks hitting together. But in reality, the click sound can be anything you want it to be. Sonar has a built-in metronome which can be configured to play back any sort of "single-hit" audio sample including stick hits, cowbells, congas, hi hats, finger-snaps, burps, farts, etc. Any sort of quick, percussive sound is a good choice for a click track.
The point of playing to a click track is to make sure the song maintains a steady and even tempo from start to finish. A consistent tempo is important for certain styles of music (rap, industrial metal, dance, etc). In some cases, however, you don't want the tempo to be consistent. Jazz and Blues and R&B are good examples of musical styles that need to breathe and swing a bit more.
One of the main editing advantages to using a click track is that you can copy sections of music from anywhere in the song, and paste them anywhere else in the song, and the tempo will match. The music will "line-up". If you don't play to a click track, it's unlikely that the tempos will match (from start to finish), making editing difficult or impossible.
Fade in/Fade out - pretty much like it sounds, this refers to the lowering or raising of the volume of the audio at certain points (usually the beginning or ending of a song or clip). Most digital audio recording programs give you the ability to create automated fade curves at the beginning and ending of audio clips. It simulates the raising and lowering of faders on a mixing console (or simply turning the volume knob up and down).
DAW - stands for Digital Audio Workstation. Describes any sort of PC-based or hard drive equipped recording system including desktop computers.
Condenser microphone - refers to microphones that need +48 volts DC phantom power (or some sort of external power source) in order to function. Phantom power is usually supplied by the mixing console or microphone preamp. It's called "phantom power" because the 48 volts travels backwards through the mic cable towards the microphone without causing any interference to the audio signal which is coming from the microphone and moving towards the mic preamp. So you've got electricity moving in two different directions along the same wire. Pretty sneaky huh? Phantom power can usually be turned on and off at the mixer because some microphones can be damaged by the presence of phantom power (such as ribbon microphones).
A condenser is really just a fancy word for a capacitor. A capacitor is an electrical device used to store and release energy. A capacitor consists of a pair of closely spaced metal conductors or plates. When a positive voltage is applied to one of the plates, an equal (but opposite) charge appears on the other plate. This electrical phenomenon basically explains how a condenser mic works. The diaphragm inside a condenser mic is a thin piece of metal-coated plastic, positioned in front of a charged plate. When air waves hit the diaphragm, these vibrations result in a change in distance between the diaphragm and the charged plate causing small changes in electrical current which are sent to the mixer or preamp.
Condenser mics usually have a wider frequency range than your garden-variety dynamic mics. As a result, they tend to produce a much crisper, more realistic sound. They are much more sensitive to sounds within a given space. Every little breath noise or rustle gets picked up. As a result, condenser microphones are typically not handled or held during a performance. They are mounted to a mic stand.
Dynamic microphone - These are the typical hand-held style microphones that most performers use on stage. Dynamic style mics don't need an external power source and are not affected at all by the presence of phantom power. The Shure SM58 is one of the most popular dynamic mics in the world.
Pop Filter - A pop filter is a piece of acoustically transparent material (sort of like nylon stocking) which is stretched across a circular hoop or frame and suspended in front of a microphone by way of a flexible gooseneck attachment. In the pictures below you can see examples of commercial pop filters and how they are positioned in front of a microphone. Basically, all a pop filter does is deflect some of the breath noise caused by hard consonants like "P's" or "B's" or "T's" which can create an annoying popping noise in the vocal track.
Compressor/limiter - these are standard audio engineering tools used to control the dynamics (wide swings in volume) of a musical performance or audio recording. In a nutshell, a compressor makes soft sounds louder and loud sounds softer. The amount of volume increase or decrease is user adjustable. A limiter basically limits loud sounds. Think of it as a sort of a protection device that is used to tame wild transients that might overload the input of a mic preamp or cause damage to a loudspeaker. Compression and limiting are closely related, so you'll often find both things incorporated into a single device. The effects of limiting and compression are very evident on radio broadcasts. Have you ever recorded a song off the radio and compared it to the actual album version? The radio version will sound kinda flat and lifeless compared to the album. That's because radio stations use heavy-handed compression and limiting to improve the sound quality of their broadcasts in cars or transistor radios or low-end playback equipment. You can read more about all this here.
Rack, rack mount - Many audio equipment manufacturers build equipment that is designed to be mounted into an equipment "rack". In this industry, the standard equipment rack is 19" wide (some example pics here). The height of a rack, or a piece of rack-mountable gear, is usually measured in "spaces". A single rack mount space is considered the smallest unit of measurement. Most gear is designed to be either one or two rack spaces high - that is - the gear occupies one or two rack spaces inside the rack. Some racks are designed to be permanently installed in a recording studio or rehearsal space. Other racks are designed to be portable (for live performances). Companies like Anvil make custom rack enclosures which are designed for many different purposes and types of equipment. Racks simply protect the gear and provide a neat way to organize the gear for everyday use.
Speakers, monitors, cans - In a recording studio, speakers are called monitors and are sometimes referred to as near-field or mid-field. Near-field monitors are meant to be positioned close to the engineer - usually sitting on top of the mixing console or desk. They are typically smaller bookshelf style speakers. Mid-field monitors are usually positioned farther away from the engineering position (usually soffit-mounted into the front wall).
Reverb, delay, echo - these are "effects" that are added to audio tracks - usually through some sort of electronic device called an "effects unit". These effects simulate sound bouncing around inside a room or an enclosed space. It gives the music depth. Without reverb or delay or echo, a musical performance sounds very dry and unflattering (which can be a good thing sometimes). This is a vague generalization I suppose, but reverb, delay and echo are sort of defined by the amount of time it takes a signal to bounce off of something and return to your eardrums. Reverb has very short return times and there is usually a lot of "smearing" of the sound on it's way around a room and back to the listener. Like singing in the shower. Delay has slightly longer return times which means there is more definition to the returned signal - like singing in a gymnasium or auditorium. Echo - well, I think we all know what that is right? It's just a copy of the original signal, delayed by many seconds which bounces off of something and comes back to your ear - like singing in the Grand Canyon. The amount of time it takes for a sound to bounce around and come back is usually measured in milliseconds - as a result, effects units allow reverb, delay and echo adjustments to be variable from zero up to say 1000 milliseconds (one second). Or more.
Reverb is probably the most commonly used effect in the history of recording. Almost everything sounds good with at least a little reverb dialed-in. Reverb is used to give the music more of a realistic impression that you are standing in the same room with the musicians while they are performing. Delay and echo are used more as musical embellishments.
Balanced/Unbalanced - These words are used to describe two types of common professional audio connections. I'll start with balanced technology first because it's the more complicated of the two terms. A balanced connection is designed to remove noise from an audio path. Balanced technology works by incorporating two simple electrical design principles called polarity inversion and phase cancellation.
When you take an audio signal and invert it 180 degrees, that's called polarity inversion as shown in Fig 1 below.
Fig. 1
This representation shows two identical audio waveforms which are 180 degrees out of phase with each other. As the top waveform goes positive, the bottom waveform goes negative and vice versa. I don't know if you remember your 8th grade math but when you have two numbers which are equally positive and negative and add them together, it equals zero. They basically cancel each other out. Well, it's like that with audio signals. When you combine two identical audio signals that are 180 degrees out of phase with each other (equally positive and negative), they cancel each other out completely. No audio. Dead silence. That is called phase cancellation. The ability to invert an audio signal and then combine two identical out of phase signals like this is the fundamental explanation for how balanced technology works.
In the wacky world of sound engineering, you sometimes need to run audio signals over very long distances. Consider a live outdoor concert for example. The front-of-house soundman is usually located somewhere out in the middle of the audience area which can be a hundred feet or more from the stage. In order to control the sound coming from the stage, there has to be audio connections from the mixer position to the stage. The problem with running cables over such a potentially long distance is that there is a good chance you will pick up interference from external electrical sources (such as police radios, fluorescent lights, motors, air conditioners, power lines, you name it). Microphones generate very weak, low amplitude signals and those weak signals are very likely to be damaged or adversely affected by cable noise. Balanced technology to the rescue.
A typical balanced mic or instrument cable has three conductors as shown in Fig 2.
Fig.2
The two inner conductors are designated as positive and negative and constitute what's known as a "twisted pair". In case you didn't know, twisted pair wiring provides a type of noise reduction by canceling out external electromagnetic interference and cross talk. The third conductor is a braided shield which completely surrounds the two inner conductors. The braided shield is connected to ground and is the first line of defense against external noise.
The negative and positive signals are 180 degrees out of phase with each other. If noise is introduced along the length of the cable, and manages to get past the braided shield, it will usually show up equally in both connections (in phase). At the mixer or preamp side, the two out of phase audio signals are inverted and combined. When you invert these two out of phase signals, they are now in-phase. You can combine them and nothing happens because they are identical. You have your original audio signal. But the noise (which was previously in-phase) is now 180 degrees out of phase and is 100% canceled out. Make sense? Probably not. The point is, noise is reduced or eliminated before it gets to our ears thanks to the electrical circuitry inside the mixing console or mic preamp or whatever audio device you are using that incorporates balanced audio connections.
Unbalanced cables are simply two conductor cables - an inner wire surrounded by a braided shield as shown in Fig 3.
Fig. 3
The shield, which is grounded, offers some level of protection but there is still a possibility that noise might get in there and get passed along. Generally speaking, unbalanced cables are kept shorter for this reason.
So if balanced is better than unbalanced, why not use it all the time? Why don't equipment manufacturers just design all their stuff for balanced connections? Good question. The answer to that is simple - economics. It's cheaper to produce unbalanced equipment. A designer will decide up front how and where this piece of equipment is most likely to be used. A home stereo system, for example, uses unbalanced connections (although some of the higher end stuff uses balanced connections). A lot of pro audio equipment incorporates both balanced and unbalanced connections because you never know what type of equipment you will be required to interface with. If you have the option of running balanced vs. unbalanced always choose balanced, even for short cable runs. How do you know if a connection is balanced or unbalanced? Another good question. Usually that information is printed near the connector area or is indicated as such in the user instructions. You can also sometimes tell by the type of connectors used as described in the following sections.
XLR - XLR refers to the industry-standard, three conductor "balanced" microphone connectors that you find on your typical microphone cables. Originally designed by Canon which is why they are sometimes referred to as "canon plugs". The letters 'XLR" have an actual meaning and if you're curious, you can read about it here. These connectors are designed for quick, robust connectivity between mics and mic preamps or any equipment which is designed for balanced connections. Available in both male and female versions as shown in the picture below.
TRS - stands for "tip, ring, sleeve" and refers to instrument plugs which have three separate conductors. As shown in the picture below, there is a tip, a ring and a sleeve on the metal barrel end. The tip and ring carry positive and negative balanced audio signals and the sleeve is ground. If it's a headphone feed, then the tip is left side and the ring is right side signal. The other common plug standard is referred to as TS or "tip/sleeve" because it only carries two conductors and doesn't have the little metal ring beneath the tip (like your guitar cables for example). Those types of connectors are said to be unbalanced.
These types of plugs are designed for quick insertion and removal. I suspect that the origins of these plug designs dates back to the early days of telephony where you had telephone operators constantly plugging cables into a telephone patch bay all day long. They needed to work quickly and these types of connectors facilitated that type of work.
RCA - RCA plugs are commonly found on your garden-variety consumer electronics devices such as home stereos, VCR's, DVD players, cassette decks, video cameras, etc.
These are always considered unbalanced connections. The connectors are often color coded to indicate left (white), right (red) or video (yellow). These types of plugs provide good long-term connectivity for A/V devices. They are better for more permanent wiring solutions (especially if there is video involved) because they tend to provide greater metal-to-metal contact than the TRS or TS version plugs. They are a little harder to insert and remove though which makes them not so good in places where you need frequent access.
Snake - a snake is a bundle of wires that carries audio signals. Snakes come in many varieties and configurations and are basically used to route signals over long distances. The most common usage for a snake is with live sound systems. A microphone snake is used to connect all the stage mics to the front-of-house mixing console which is usually located out in front of the band somewhere. A snake makes connecting microphones and other audio gear more convenient (and ultimately safer) because now you have a single multi-conductor cable as opposed to many individual cables.
And the word "snake" is a good word to describe these things. I was working with a band one time that had a 24 channel, 150 foot snake. This thing was almost too heavy for a single person to lift and was definitely a two or three man operation when it came time to install and remove it during each gig. I literally felt like a snake-handler every time I had to uncoil this massive wad. :-)
Mixer, mixing console - the nerve center of a recording studio. Mixers (also called mixing boards, mixing consoles, mixing desks, etc) come in many shapes and sizes from simple tabletop models costing around a hundred bux...
to large desks costing a quarter million dollars or more and occupying more than one zip code...
A mixing console is an electronic device that allows you to connect a bunch of microphones (and other audio devices like drum machines, tape decks, synthesizers, outboard effects units, etc.) and "mix" the audio from all these sound sources. Audio coming from many different sources has to be mixed together in order to create a pleasing blend (otherwise known as a "hit song"). A mixing console contains X number of "channel strips" which is where you make volume and tone adjustments for all the microphones and outboard gear connected to the mixer. The number of inputs you can connect at any one given point in time depends, of course, on how many channel strips the board is designed to handle. The smaller tabletop models that you can pick up and carry under one arm usually accommodate anywhere from 4 to 16 channels. The big "desks" that you find in many commercial recording studios can come equipped with a hundred or more channels. The number of channels you need usually depends on the size and function of the studio. A modest project studio designed for recording rock bands or rap artists might only need 8 to 24 channels of mixing capability. A studio designed for film scoring (like in the picture above) might need 72 or more channels. These things look pretty daunting right? And they can be. But in reality, a typical mixing desk is just a bunch of identical channel strips. Learn one channel strip and you've learned a lot about the board in general.
Most of these large format mixers (and even a few specialized smaller boards) have what are called "flying faders". Which is just a fancy term to describe the fact that the individual volume faders are automated. You can program the position of each fader for a particular point in time and then instantly change their positions with the push of a button. Each fader has a small motor behind it which will automatically move the fader for you. This makes it easy for an engineer to set up and recall a variety of fader positions for each song. Faders don't usually remain in one place for the entire duration of a mix so this feature makes it easy for one person to mix an entire song. If you have 60+ tracks of audio, having automated faders can be a good thing :-)
Channels - People sometimes mistakenly use the word "channel" when they really mean "track" or vice versa. So I thought I would say something about the word "channel" and what it means in a studio environment. Channels usually refer to the inputs on a mixing board (as described above). For example, you might hear someone say, "this is a sixteen channel mixer". That means you can connect up to sixteen different signal sources to this device and mix or route those signals however you see fit. Mixing boards are typically defined by the number of input channels they have. Channels are electrical paths used to route audio signals within a particular piece of gear.
Which leads us to...
Preamps - We can't really talk about channels without also talking about preamps since the two things are often interrelated. A preamp is an electronic device (or special circuit) used to raise the level of an incoming audio signal from a microphone or musical instrument of some sort. The audio signal that comes traveling down a mic cable or a guitar cable is a very weak, low amplitude signal. In order to do anything meaningful with these signals, it needs to be amplified or "brought up" in level so that the signal can be effectively "processed" further on down the signal chain. That's one of the main purposes of a channel strip - to boost the incoming signal level so that we can apply EQ and other effects later on. Why exactly do we need to do this? Well, weak, low amplitude audio signals are prone to being damaged by noisy components or RF interference (even the connecting cables can add noise to the signal). Have you ever picked up police transmissions or ham radio signals or cell phone calls through your gear? There's a lot of garbage floating around the atmosphere. Transducers (such as those found in microphone elements and guitar pickups) aren't capable of producing very strong electrical signals. A preamp's main purpose in life is to boost the signal so the audio is well above the noise floor of the gear that it eventually comes in contact with. Otherwise all those resistors and capacitors and transistors and inductors would simply degrade and ruin the original signal.
I'll add more as time permits.
Some internet reference materials to recording and recording studios that you might find interesting:
http://www.audioed.com.au/secure/Platinum/modules/Studio%20Intro/mod_intro1_1.html