You are on page 1of 16

Dubspot Producer’s Handbook

Welcome to Dubspot’s Electronic Music Production Program! Soon you will be making your way into
the world of beats and production, creating your own original sounds and recordings.

Music production is a vast topic, with years of history, techniques, and technological developments.
Provided in this workbook is some general information, technical descriptions, and terminology to
benefit all new and aspiring producers.

Read this workbook and familiarize yourself with the information; the terms and concepts presented
here will undoubtedly cross your path time and time again as your progress through your courses at
Dubspot and into your future music projects!

History of Music Recording


Let’s begin with a look at some important chapters and milestones in the
history of music recording that have carried us to the point we are today.

Early Recordings
The earliest methods of recording sounds involved recording of a live
performance directly to the recording medium. One of the earliest
recording devices was the phonograph, invented by Thomas Edison in
1877. The phonograph was a device with a cylinder covered with a soft
material such as tin foil, lead, or wax. The sound of the performers was
captured by a diaphragm with the cutting needle connected to it, the
sound caused the stylus to draw grooves into the surface of the
recording material. The depth of the grooves made by the stylus
corresponded to change in air pressure created by the original sound.
The recording could be played back by tracing a needle through the
groove and amplifying, through mechanical means, the resulting
vibrations, resulting in what soon became a very popular means of
recording and distributing music.

This changed with the advent of the gramophone (phonograph in


American English), which was patented by Emile Berliner in 1887. The
gramophone imprinted grooves on the flat side of a disc rather than the
outside of a cylinder. Instead of recording by varying the depth of the
groove (vertically), as with the phonograph, the vibration of the recording
stylus was across the width of the track ( horizontally).
Tape Recording
Magnetic tape recording, developed by the Germans during the Second
World War, was used both in radio broadcasts and for the deciphering of
intercepted code messages.

The poor sound quality obtained on that early equipment soon improved with
the introduction of commercial recorders and new tapes by Telefunken in
Europe and Ampex in the United States. These machines were mono, full
track (one 1/4" wide track).

It was discovered early on that you could overdub by playing back a


previously recorded tape through a mixer, blending that with live mics, and
sending the composite signal to a second recorder. In the history of record
production, this early form of multi-tracking rivaled the invention of the wheel.

Unfortunately, the pretaped music lost a bit of sound quality and gained a bit
of extra noise. Nevertheless, this technique did allow artists to add layers of
new music. Such mono-to-mono copy-overdubbing was the standard in pop
music production up until 1962. Until that time, pop and rock records were
made to sound good on AM radio-in highly compressed mono.

Multitrack Recording
The next major development in magnetic tape was multitrack recording, in
which the tape is divided into multiple tracks parallel with each other.
Because they are carried on the same medium, the tracks stay in perfect
synchronization. Basic tracks could be laid down on one track, some
instrumental overdubs (perhaps even horns or strings) on the second track
(recorded while these musicians heard a headphone playback of track 1),
then both tracks mixed onto a second machine.

The next development in multitracking was stereo sound, which divided the
recording head into two tracks. Stereo quickly became the norm for
commercial classical recordings and radio broadcasts, although many pop
music and jazz recordings continued to be issued in monophonic sound until
the mid-1960s.

Much of the credit for the development of multitrack recording goes to


guitarist, composer and technician Les Paul, who also helped design the
famous electric guitar that bears his name. In 1955, Les Paul commissioned
Ampex Corporation to build a custom recorder, with eight parallel tracks to be
recorded onto special 1" wide tape.

The first Ampex 8-track was delivered the next year, and Les proceeded to
make a string of Top 10 hits on it. He developed almost all the techniques
and tricks that later became standard in multitrack sessions--headphone or
cue mixing, overdubbing, bouncing tracks, prelaying effects and delays, and
special varispeed operations.
Multitrack recording was immediately taken up in a limited way by
Ampex, who soon produced a commercial 3-track recorder. These
proved extremely useful for popular music, since they enabled backing
music to be recorded on two tracks (either to allow the overdubbing of
separate parts, or to create a full stereo backing track) while the third
track was reserved for the lead vocalist.

Three-track recorders remained in widespread commercial use until the


mid-1960s and many famous pop recordings, including many of Phil
Spector's so-called "Wall of Sound" productions and early Motown hits,
were taped on Ampex 3-track recorders.

The next important development was 4-track recording. The advent of


this improved system gave recording engineers and musicians vastly
greater flexibility for recording and overdubbing, and 4-track was the
studio standard for most of the later 1960s.

Many of the most famous recordings by The Beatles and The Rolling
Stones were recorded on 4-track, and the engineers at London's Abbey
Road Studios became particularly adept at a technique called
"reduction mixes" in the UK and "bouncing down" in the United States,
in which multiple tracks were recorded onto one 4-track machine and
then mixed together and transferred (bounced down) to one track of a
second 4-track machine. In this way, it was possible to record literally
dozens of separate tracks and combine them into finished recordings of
great complexity.

4-track tape also enabled the development of quadraphonic sound, in


which each of the four tracks was used to simulate a complete 360-
degree surround sound. A number of albums including Pink Floyd's
Dark Side of the Moon and Mike Oldfield's Tubular Bells were released
both in stereo and quadrophonic format in the 1970s, but 'quad' failed
to gain wide commercial acceptance. Although it is now considered a
gimmick, it was the direct precursor of the surround sound technology
that has become standard in many modern home theatre systems.

In a professional setting today, such as a studio, audio engineers may


use 24 tracks or more for their recordings, utilizing one or more tracks
for each instrument played.

source:
Middleton, Richard (1990/2002). Studying Popular Music. Philadelphia: Open University Press
Digital Recording
In the 1990s, digital systems were introduced and began to prevail.
Within a few years after the introduction of digital recording, multitrack
recorders were being commonly used in professional studios. In the
early 1990s, relatively low-priced multitrack digital recorders were
introduced for use in home studios; they returned to recording on
videotape. The most notable of this type of recorder is the ADAT.
Developed by Alesis and first released in 1991, the ADAT machine is
capable of recording 8 tracks of digital audio onto a single S-VHS video
cassette. The ADAT machine is still a very common fixture in
professional and home studios around the world.

As hard disk capacities and computer CPU speeds increased at the


end of the 1990s, hard disk recording became more popular. Hard disk
recording is available in two forms. One is the use of standard desktop
or laptop computers, with adapters for encoding audio into tracks of
digital audio. These adapters can either be built-in soundcards or
external, connecting to the computer using internal audio interface
cards, USB, or Firewire cables.

The other form of hard disk recording uses a dedicated recorder unit
which contains analog-to-digital and digital-to-analog converters as well
as one or two removable hard drives for data storage. Such recorders,
packing 24 tracks in a few units of studio rack space, are actually
single-purpose computers, which can in turn be connected to standard
computers for editing.

Notable Producers & Engineers


Here is a short list of some of the countless great producers who have contributed to the wide range of
music that shaped the world we live in.

Tom Dowd (Atlantic Records) Produced early hit record for Ray Charles, Aretha Franklin, Charlie
Parker, and dozens of other famous jazz, soul, and rock artists.

Berry Gordy (Motown) Founder of legendary label Motown, his artists included the Supremes, Stevie
Wonder, and Smokey Robinson to name a few.

Ken Scott (UK) From Abbey Roads Studio, Ken was behind the boards for the Beatles, Pink Floyd, and
David Bowie, and other successful acts.

King Tubby (Jamaica) Legendary Jamaican producer and one of pioneers of dub and reggae music as
we know it today, developing innovative ways of using effects and the mixing console as an instrument.

J-Dilla (Detroit) Highly influential hip hop producer, responsible for the beats behind countless hits for
Tribe Called Quest, Busta Rhymes, De La Soul, the Pharcyde, Common, and many more.

Next time you are listening to your favorite albums, be sure to check the liner notes and
find out who was behind the scenes making the music sound so amazing!
What is the difference between audio and MIDI?
A seemingly simple question that perplexes many newcomers to audio and recording...

When you record an audio signal, then the acoustic or electronic


waveform that the instrument produces is captured directly. The
recording is a representation of the sound the instrument actually made,
and will be different according to whether the instrument was, say, a
violin or a trumpet. An audio signal is recorded on an audio track of a
digital audio workstation software.

A MIDI (Musical Instrument Digital Interface) signal is normally


generated by a playing on a keyboard, and it contains information about
which keys are being pressed. The MIDI signal can be recorded on a
MIDI track of a digital audio workstation. Only the data about which keys
were pressed, plus other associated data, is recorded. So the MIDI
signal doesn't sound like a violin or a trumpet, it is merely a list of which
keys were pressed and when.

The advantage of using MIDI is that you can change the instrumentation
the MIDI notes you recorded will play later.

Regular audio isn't as flexible: Your recording of a violin will always sound
like a violin. You can EQ it, but it will still sound like a violin.

However, you might have had your keyboard set to a violin sound when you recorded your MIDI
signal, but when you play it back you can set your keyboard to any sound you like. So what was
once a violin can now very easily become a trumpet, etc.

MIDI has further advantages: You can edit the MIDI data more flexibly than audio. For instance
you can correct the timing of notes, or how forcefully they were played. You can correct wrong
notes, transpose parts, change the tempo, and more!

Since MIDI's introduction in the music industry, most electronic


instruments sold have built in MIDI ports to connect them to a
sequencer or other MIDI instruments. The ports use special MIDI
cable to communicate performance data. With the popularity of
desktop or laptop studios, there are also a wide variety of keyboards,
drum pads, and controller devices that transmit MIDI information to
the computer thru a USB cable to allow hands-on performance and
control of software instruments and devices.

Audio signals from an instrument or microphone need to be routed into the audio inputs on digital
music interface or mixer using the appropriate cables. Typically, microphones use XLR mic
cables, whereas instruments such as synths or drum machines use 1/4” audio cables.
A Guide To Hardware Devices vs. Virtual Instruments & Effects
If you have been recording or producing music, you may have heard the terms hardware and VST. What is the
difference between hardware and VST and how does it figure in to playing, recording, and editing music?

What is Hardware?
Hardware refers to any physical piece of music equipment. This includes mixing consoles, synthesizers, samplers,
drum machines, rack mount effects processors, and more.
The advantage of having dedicated hardware equipment is the reliability of a dedicated piece of gear (no software
crashes!), the unique sound qualities and properties each unit possesses, and the hands-on "tweak-ablility" of the
dedicated knobs and sliders on a physical piece of gear. The downside is the high-cost (certain hardware synths
exceed $3000 and high-quality mixing consoles can exceed $25,000!) as well as the limits of physical space
available for mountains of equipment in a small home or professional studio.

Hardware instruments are recorded by connecting audio cables from the outputs of the instruments to the mixer or
recording device being used. They may also be synched by the use of MIDI cables to play together with other
devices, control another device, or be controlled by a sequencer.

Prior to advent of cheap and powerful computers on the market, and the powerful music software programs we take
for granted now, nearly every studio and producer was using all hardware devices to make and record their music.
Even with today's music technology, many players still prefer the sound and feel of physical equipment, and still
swear by certain classic hardware gear as part of their recording process.

What is VST?
VST, Virtual Studio Technology, is an interface standard for connecting synthesizers and effects to audio editors and
recording systems. Invented and developed by Steinberg, makers of the Cubase audio recording programs; VST
replaces traditional audio recording hardware with software equivalents. The advantage of VSTs is there is a wide
range of virtual synths, high-quality effects, samplers, and drum machines capable of emulating and in some cases
surpassing its much more expensive hardware counterparts, giving you an tremendously expanded palette of
sounds and production tools right on your existing computer!

There are 2 different types of VST plug-ins, with thousands of different varieties, making it the most widely used
plug-in type. The 2 main types are:

VST instruments – Also know as a VSTi, they take the form of synthesizers, samplers, drum machines and other
instruments. They can be played in real time, or also used in conjunction with MIDI for live performance. They are
basically a software version of a piece of hardware. Lugging the soft synth version of the B3 Organ around on a
laptop is a whole lot easier than having to carry around the hardware version!

VST effects – VST effects are used to process audio, like any other type of audio effect. There are VST equivalents
to every type of audio effect available as hardware, including many rare and expensive rack mount hardware
processors previously only available to a handful of producers in a few of the world's top-notch recording studios.

VST instruments and effects must be used in conjunction with a VST host.
A VST host is usually a software application, such as Cubase, Ableton Live, Logic, or others.

Audio Units (AU) - Similar to VST are Audio Unit or AU plug-ins. They are a system-level plug-in architecture
provided by Core Audio in Mac OS X developed by Apple Computer. It may be thought of as Apple's architectural
equivalent to the other popular plug-in format, Steinberg's VST.

AU are used by Apple applications such as GarageBand, Soundtrack Pro, Logic Express, Logic Pro, Final Cut Pro
and most 3rd party audio software developed for Mac OS X.
Music Software: “Which Do I Choose?”
With so many great software options to make electronic music currently on the market, it can be confusing
what the different programs do. Many new producers might feel discouraged just trying to figure out
which programs to pursue.

In the end, some producers may choose to focus on just one program to create their music, while others
will find using a combination of several programs together will yield the best sounding results. The most
important point with music production to remember, though, is that it’s not about the software you use, but
ultimately making great songs that you and others enjoy listening to!

At Dubspot, we teach many of the most popular music programs used by today’s producers; here is a list
of some of the main programs we teach and a description of each.

REASON 4
An easy to learn all in one compositional tool for midi-programming containing a
big rack of instruments and effects and an intuitive sequencer layout. This
program will appeal to the hardware lovers as it mimics the functionality of a
real hardware rack with cables for customized wiring. ideal for learning the
basics of synth and midi programming. huge sound library with easy pre listen
and load features. No audio recording but configurable for hands-on live
performance, Reason 4 offers an ideal introduction to music production with
midi instruments.

ABLETON LIVE
The current market leader for electronic music production due to it’s unique and
easy interface that allows you to build tracks in a improvisational way both with
audio and midi clips. This is a all in one package suitable equally for live, studio
and DJ uses It is particularly well equipped for mixing and mashing audio
content due to it’s automatic tempo matching capabilities (warping). Recording
and editing midi can be done in or out of timeline. Live is the software of choice
for performing electronic musicians and Djs alike, but it also excels in the studio
for mixing with its innovative production tools and endless sound and
automation possibilities. The Suite ships with synthesizers and a big Library.

LOGIC 8
This great Legacy Software by Apple is particularly strong for studio uses with
great overall sound quality, a nice collection of midi instruments and the unique
Ultra beat Drum sequencer. This app has a lot of detail and depth for producing
great sounding tracks and songs in and is equally suited for the recording of
live instruments and midi sequencing. Logic 8 comes with a big sound library of
both midi and audio content. Composition is done in a straightforward timeline
format only and offers a lot of control for editing, automation and mixing but the
the overall structure makes it less suitable for improvisational live performance.
Digital Audio
Digital audio recording works by recording, or sampling, an electronic audio signal at
regular intervals (of time). An analog-to-digital (A/D) converter measures and stores
each sample as a numerical value that represents the audio amplitude at that particular
moment. Converting the amplitude of each sample to a binary number is called
quantization. The number of bits used for quantization is referred to as bit depth.
Sample rate and bit depth are two of the most important factors when determining
the quality of a digital audio system.

Sample Rate
The sample rate is the number of times an analog signal is measured—or sampled—
per second. You can also think of the sample rate as the number of electronic
snapshots made of the sound wave per second. Higher sample rates result in higher
sound quality because the analog waveform is more closely approximated by the
discrete samples. Which sample rate you choose to work with depends on the source
material you’re working with, the capabilities of your audio interface, and the final
destination of your audio.
For years, the digital audio sample rate standards have been 44,100 Hz (44.1 kHz) and
48 kHz. However, as technology improves, 96 kHz and even 192 kHz sample rates are
becoming common.

Bit Depth
Unlike analog signals, which have an infinite range of volume levels, digital audio
samples use binary numbers (bits) to represent the strength of each audio sample. The
accuracy of each sample is determined by its bit depth. Higher bit depths mean your
audio signal is more accurately represented when it is sampled. Most digital audio
systems use a minimum of 16 bits per sample, which can represent 65,536 possible
levels (24-bit samples can represent over 16 million possible levels).
To better understand bit depth, think of each digital audio sample as a ladder with
equally spaced rungs that climb from silence to full volume. Each rung on the ladder is
a possible volume that a sample can represent, while the spaces between rungs are
in-between volumes that a sample cannot represent.
Sample Rates
When a sample is made, the audio level of the analog signal often falls in the spaces
between rungs. In this case, the sample must be rounded to the nearest rung. The bit
depth of a digital audio sample determines how closely the rungs are spaced. The more
rungs available (or, the less space between rungs), the more precisely the original
signal can be represented.

Quantization errors occur when a digital audio sample does not exactly match the
analog signal strength it is supposed to represent (in other words, the digital audio
sample is slightly higher or lower than the analog signal). Quantization errors are also
called rounding errors because imprecise numbers represent the original analog audio.
For example, suppose an audio signal is exactly 1.15 volts, but the analog-to-digital
converter rounds this to 1 volt because this is the closest bit value available. This
rounding error causes noise in your digital audio signal. While quantization noise may
be imperceptible, it can potentially be exacerbated by further digital processing.
Always try to use the highest bit depth possible to avoid quantization errors.
The diagram on the far right shows the highest bit depth, and therefore the audio
samples more accurately reflect the shape of the original analog audio signal.

For example, a 1-bit system (a ladder with only two rungs) can represent either silence
or full volume, and nothing in between. Any audio sample that falls between these
rungs must be rounded to full volume or silence. Such a system would have absolutely
no subtlety, rounding smooth analog signals to a square-shaped waveform.
When the number of bits per sample is increased, each sample can more accurately
represent the audio signal.

To avoid rounding errors, you should always use the highest bit depth your equipment
supports. Most digital video devices use 16- or 20-bit audio, so you may be limited to
one of these bit depths. However, professional audio recording devices usually support
24-bit audio, which has become the industry standard
GLOSSARY OF TERMS
ANALOG-TO-DIGITAL (A/D) CONVERTER: A circuit that converts an analog audio signal into a stream of digital data (bit
stream).

ASSIGN: To route or send an audio signal to one or more selected channels.

ATTACK: The beginning of a note. The first portion of a note's envelope in which a note rises from silence to its maximum
volume.

ATTACK TIME: In a compressor, the time it takes for gain reduction to occur in response to a musical attack.

AUTOMATED MIXING: A system of mixing in which a computer remembers and updates console settings. With this
system, a mix can be performed and refined in several stages and played back at a later date exactly as set u previously.

BANDPASS FILTER: In a crossover, a filter that passes a band or range of frequencies but sharply attenuates or rejects
frequencies outside the band.

CHANNEL: A single path of an audio signal. Usually, each channel contains a different signal.

CHORUS: 1. A special effect in which a signal is delayed by 15 to 35 milliseconds, the delayed signal is combined with the
original signal, and the delay is varied randomly or periodically. This creates a wavy, shimmering effect. 2. The main
portion of a song that is repeated several times throughout the song with the same lyrics.

COMB-FILTER EFFECT: The frequency response caused by combining a sound with its delayed replica. The frequency
response has a series of peaks and dips caused by phase interference. The peaks and dips resemble the teeth of a comb.

COMPRESSION: 1. The portion of a sound wave in which molecules are pushed together, forming a region with higher-
than-normal atmospheric pressure. 2. In signal processing, the reduction in dynamic range or gain caused by a
compressor. 3. In computing, data compression reduces the number of bytes in a file without losing essential information.

COMPRESSION RATIO (SLOPE): In a compressor, the ratio of the change in input level (in dB) to the change in output
level (in dB). For example, a 2:1 ratio means that for every 2 dB change in input level, the output level changes 1 dB.

COMPRESSOR: A signal processor that reduces dynamic range or gain by means of automatic volume control. An
amplifier whose gain decreases as the input signal level increases above a preset point.

CONDENSER MICROPHONE: A microphone that works on the principle of variable capacitance to generate an electrical
signal. The microphone diaphragm and an adjacent metallic disk (called a backplate) are charged to form two plates of a
capacitor. Incoming sound waves vibrate the diaphragm, varying its spacing to the backplate, which varies the
capacitance, which in turn varies the voltage between the diaphragm and backplate.

CROSSOVER: An electronic network that divides an incoming signal into two or more frequency bands.

CUE SYSTEM: A monitor system that allows musicians to hear themselves and previously recorded tracks through
headphones.

DAW: Abbreviation for digital audio workstation.

dB: Abbreviation for decibel.

DECAY: The portion of the envelope of a note in which the envelope goes from maximum to some midrange level. Also,
the decline in level of reverberation over time.

DECIBEL: The unit of measurement of audio level. Ten times the logarithm of the ratio of two power levels. Twenty times
the logarithm of the ratio of two voltages.
DE-ESSER: A signal processor that removes excessive sibilance ("s" and "sh" sounds) by compressing high frequencies
around 5 to 10 kHz.

DELAY: The time interval between a signal and its repetition. A digital delay or a delay line is a signal processor that
delays a signal for a short time.

DIGITAL AUDIO: An encoding of an analog audio signal in the form of binary digits (ones and zeros).

DIGITAL AUDIO WORKSTATION (DAW): A computer, sound card, and editing software that allows you to record, edit and
mix audio programs entirely in digital form. Stand-alone DAWs include real mixer controls; computer DAWS have virtual
controls on-screen.

DIGITAL RECORDING: A recording system in which the audio signal is stored in the form of binary digits.

DIGITAL-TO-ANALOG CONVERTER: A circuit that converts a digital audio signal into an analog audio signal.

DIRECT BOX: A device used for connecting an amplified instrument directly to a mixer mic input. The direct box converts
a high-impedance unbalanced audio signal into a low-impedance balanced audio signal.

DISTORTION: An unwanted change in the audio waveform, causing a raspy or gritty sound quality. The appearance of
frequencies in a device's output signal that were not in the input signal. Distortion is caused by recording at too high a
level, improper mixer settings, components failing, or vacuum tubes distorting. (Distortion can be desirable--for an electric
guitar, for example.)

DOUBLING: A special effect in which a signal is combined with its 15 to 35 millisecond delayed replica. This process
mimics the sound of two identical voices or instruments playing in unison. In another type of doubling, two indentical
performances are recorded and played back to thicken the sound.

DRUM MACHINE: A device that plays samples of real drums, and includes a sequencer to record rhythm patterns.

DRY: Having no echo or reverberation. Referring to a close- sounding signal that is not yet processed by a reverberation
or delay device.

DSP: Abbreviation for Digital Signal Processing, modifying a signal in digital form.

DYNAMIC RANGE: The range of volume levels in a program from softest to loudest.

ECHO: A delayed repetition of a signal or sound. A sound delayed 50 milliseconds or more, combined with the original
sound.

EFFECTS: Interesting sound phenomena created by signal processors, such as reverberation, echo, flanging, doubling,
compression, or chorus.

EFFECTS LOOP: A set of connectors in a mixer for connecting an external effects unit, such as a reverb or delay device.
The effects loop includes a send section and a receive section. See Effects Send, Effects Return.

EFFECTS RETURN (EFFECTS RECEIVE): In the output section of a mixing console, a control that adjusts the amount of
signal received from an effects unit. Also, the connectors in a mixer to which you connect the effects-unit output signal.
They might be labeled "bus in" instead. The effects-return signal is mixed with the program bus signal.

EFFECTS SEND: In an input module of a mixing console, a control that adjusts the amount of signal sent to a special-
effects device, such as a reverberation or delay unit. Also, the connector in a mixer which you connect to the input of an
effects unit. The effects-send control normally adjusts the amount of reverberation or echo heard on each instrument.

ENVELOPE: The rise and fall in volume of one note. The envelope connects successive peaks of the waves comprising a
note. Each harmonic in the note might have a different envelope.

EQUALIZATION (EQ): The adjustment of frequency response to alter the tonal balance or to attenuate unwanted
frequencies.
EQUALIZER: A circuit (usually in each input module of a mixing console, or in a separate unit) that alters the frequency
spectrum of a signal passed through it.

EXPANDER: 1. A signal processor that increases the dynamic range of a signal passed through it. 2. An amplifer whose
gain decreases as its input level decreases. When used as a noise gate, an expander reduces the gain of low-level
signals to reduce noise between notes.

FADE-OUT: To gradually reduce the volume of the last several seconds of a recorded song, from full level down to silence,
by slowly pulling down the master fader.

FADER: A linear or sliding volume control used to adjust signal level.

FILTER: 1. A circuit that sharply attenuates frequencies above or below a certain frequency. Used to reduce noise and
leakage above or below the frequency range of an instrument or voice. 2. A MIDI Filter removes selected note parameters.

FLANGING: A special effect in which a signal is combined with its delayed replica, and the delay is varied between 0 and
20 milliseconds. A hollow, swishing, ethereal effect like a variable-length pipe, or like a jet plane passing overhead. A
variable comb filter produces the flanging effect.

FREQUENCY: The number of cycles per second of a sound wave or an audio signal, measured in hertz (Hz). A low
frequency (for example, 100 Hz) has a low pitch; a high frequency (for example, 10,000 Hz) has a high pitch.

GAIN: Amplification. The ratio, expressed in decibels, between the output voltage and the input voltage, or between the
output power and the input power.

GRAPHIC EQUALIZER: An equalizer with a horizontal row of faders; the fader-knob positions indicate graphically the
frequency response of the equalizer. Usually used to equalize monitor speakers for the room they are in. Sometimes used
for complex EQ of a track.

HEADROOM: The safety margin, measured in decibels, between the signal level and the maximum undistorted signal
level. In a tape recorder, the dB difference between standard operating level (corresponding to a 0 VU reading) and the
level causing 3 percent total harmonic distortion. High-frequency headroom increases with analog tape speed.

HERTZ (Hz): Cycles per second, the unit of measurement of frequency.

HIGHPASS FILTER: A filter that passes frequencies above a certain frequency and attenuates frequencies below that
same frequency. A low-cut filter.

HUMAN HEARING: Human Hearing is limited from 20Hz to 20kHz. Any frequency below 20Hz is called subsonic; any
frequency above 20K is ultrasonic. The international hi-fi norm is from 30Hz to 16Khz. The human communication sounds
(speech) reach from about 65Hz to 10 kHz.

INPUT: The connection going into an audio device. In a mixer or mixing console, a connector for a microphone, line-level
device, or other signal source.

LEVEL: The degree of intensity of an audio signal--the voltage, power, or sound pressure level. The original definition of
level is the power in watts.

LIMITER: A signal processor whose output is constant above a preset input level. A compressor with a compression ratio
of 10:1 or greater, with the threshold set just below the point of distortion of the following device. Used to prevent distortion
of attack transients or peaks.

LINE LEVEL: In balanced professional recording equipment, a signal whose level is approximately 1.23 volts (+4 dBm). In
unbalanced equipment (most home hi-fi or semipro recording equipment), a signal whose level is approximately 0.316 volt
(-10 dBV).

LOWPASS FILTER: A filter that passes frequencies below a certain frequency and attenuates frequencies above that
same frequency. A high-cut filter.
MASTER FADER: A volume control that affects the level of all program buses simultaneously. It is the last stage of gain
adjustment before the 2-track recorder.

MIC LEVEL: The level or voltage of a signal produced by a microphone, typically 2 millivolts.

MIDI: Abbreviation for Musical Instrument Digital Interface, a specification for a connection between synthesizers, drum
machines, and computers that allows them to communicate with and/or control each other.

MIDI CHANNEL: A route for transmitting and receiving MIDI signals. Each channel controls a separate MIDI musical
instrument or synth patch. Up to 16 channels can be sent on a single MIDI cable.

MIDI CONTROLLER: A musical performance device (keyboard, drum pads, breath controller, etc.) that outputs a MIDI
signal designating note numbers, note on, note off, and so on.

MIDI IN: A connector in a MIDI device that receives MIDI messages.

MIDI INTERFACE: A circuit that plugs into a computer, and converts MIDI data into computer data for storage in memory
or on hard disk. The interface also converts computer data into MIDI data.

MIDI OUT--A connector in a MIDI device that transmits MIDI messages.

MIDI THRU--A connector in a MIDI device that duplicates the MIDI information at the MIDI-In connector. Used to connect
another MIDI device in the series.

MIX: 1. To combine two or more different signals into a common signal. 2. A control on a delay unit that varies the ratio
between the dry signal and the delayed signal.

MIXDOWN: The process of playing recorded tape tracks through a mixing console and mixing them to two stereo
channels for recording on a two-track tape recorder.

MIXER: A device that mixes or combines audio signals and controls the relative levels of the signals.

MIXING CONSOLE: A large mixer with additional functions such as equalization or tone control, pan pots, monitoring
controls, solo functions, channel assigns, and control of signals sent to external signal processors.

MONITOR: A loudspeaker in a control room, or headphones, used for judging sound quality.

MONO, MONOPHONIC: 1. Referring to a single channel of audio. A monophonic program can be played over one or
more loudspeakers, or one or more headphones. 2. Describing a synthesizer that plays only one note at a time (not
chords).

MONO-COMPATIBLE: A characteristic of a stereo program, in which the program channels can be combined to a mono
program without altering the frequency response or balance. A mono-compatible stereo program has the same frequency
response in stereo or mono because there is no delay or phase shift between channels to cause phase

NOISE GATE: A gate used to reduce or eliminate noise between notes.

OCTAVE: The interval between any two frequencies where the upper frequency is twice the lower frequency.

OUTBOARD EQUIPMENT: Signal processors that are external to the mixing console.

OUTPUT: A connector in an audio device from which the signal comes, and feeds successive devices.

OVERDUB: To record a new musical part on an unused track in synchronization with previously recorded tracks.

PARAMETRIC EQUALIZER: An equalizer with continuously variable parameters, such as frequency, bandwidth, and
amount of boost or cut.

PEAK: On a graph of a sound wave or signal, the highest point in the waveform. The point of greatest voltage or sound
pressure in a cycle.
PHANTOM POWER: A DC voltage (usually 12 to 48 volts) applied to microphone signal conductors to power condenser
microphones.

PHASE: The degree of progression in the cycle of a wave, where one complete cycle is 360 degrees.

PHASE CANCELLATION, PHASE INTERFERENCE: The cancellation of certain frequency components of a signal that
occurs when the signal is combined with its delayed replica. At certain frequencies, the direct and delayed signals are of
equal level and opposite polarity (180 degrees out of phase), and when combined, they cancel out. The result is a comb-
filter frequency response having a periodic series of peaks and dips. Phase interference can occur between the signals of
two microphones picking up the same source at different distances, or can occur at a microphone picking up both a direct
sound and its reflection from a nearby surface.

PHASE SHIFT: The difference in degrees of phase angle between corresponding points on two waves. If one wave is
delayed with respect to another, there is a phase shift between them of 2[pi]FT, where [pi] = 3.14, F = frequency in Hz, and
T = delay in seconds.

PHASING: A special effect in which a signal is combined with its phase-shifted replica to produce a variable comb-filter
effect. See also Flanging.

PHONE PLUG: A cylindrical, co-axial plug (usually 1/4-inch diameter). An unbalanced phone plug has a tip for the hot
signal and a sleeve for the shield or ground. A balanced phone plug has a tip for the signal hot signal, a ring for the return
signal, and a sleeve for the shield or ground.

PHONO PLUG: A coaxial plug with a central pin for the hot signal and a ring of pressure-fit tabs for the shield or ground.
Also called RCA plug.

PICKUP: A piezoelectric transducer that converts mechanical vibrations to an electrical signal. Used in acoustic guitars,
acoustic basses, and fiddles. Also, a magnetic transducer in an electric guitar that converts string vibration to a
corresponding electrical signal.

PITCH: The subjective lowness or highness of a tone. The pitch of a tone usually correlates with the fundamental
frequency.

PITCH SHIFTER: A signal processor that changes the pitch of an instrument without changing its duration.

PLUG-IN: Software effects that you install in your computer. The plug-in software becomes part of another program you
are using, such as a digital editing program.

POLYPHONIC--Describing a synthesizer that can play more than one note at a time (chords).

RELEASE: The final portion of a note's envelope in which the note falls from its sustain level back to silence.

RELEASE TIME: In a compressor, the time it takes for the gain to return to normal after the end of a loud passage.

REMIX: To mix again; to do another mixdown with different console settings or different editing.

REVERB: Natural reverberation in a room is a series of multiple sound reflections which makes the original sound persist
and gradually die away or decay. These reflections tell the ear that you're listening in a large or hard-surfaced room. For
example, reverberation is the sound you hear just after you shout in an empty gymnasium. A reverb effect simulates the
sound of a room--a club, auditorium, or concert hall--by generating random multiple echoes that are too numerous and
rapid for the ear to resolve. The timing of the echoes is random, and the echoes increase in number with time as they
decay. An echo is a discrete repetition of a sound; reverberation is a continuous fade-out of sound.

SAMPLING: Recording a short sound event into computer memory. The audio signal is converted into digital data
representing the signal waveform, and the data is stored in memory chips, tape or disc for later playback.

SEQUENCER: A device that records a musical performance done on a MIDI controller (in the form of note numbers, note
on, note off, etc.) into computer memory or hard disk for later playback. A computer can act as a sequencer when it runs a
sequencer program. During playback, the sequencer plays synthesizer sound generators or samples.
SOLO: On an input module in a mixing console, a switch that lets you monitor that particular input signal by itself. The
switch routes only that input signal to the monitor system.

SOUND CARD: A circuit card that plugs into a computer, and converts an audio signal into computer data for storage in
memory or on hard disk. The sound card also converts computer data into an audio signal.

SOUND MODULE (SOUND GENERATOR): 1. A synthesizer without a keyboard, containing several different timbres or
voices. These sounds are triggered or played by MIDI signals from a sequencer program, or by a MIDI controller. 2. An
oscillator.

SOUND PRESSURE LEVEL (SPL)--The acoustic pressure of a sound wave, measured in decibels above the threshold of
hearing. The higher the SPL of a sound, the louder it is. dB SPL = 20 log (P/P ref), where P = the measured acoustic
pressure and P ref = 0.0002 dyne/cm[superscript]2[end superscript]. [ref is subscript]

SOUND WAVE: The periodic variations in sound pressure radiating from a sound source.

STEREO, STEREOPHONIC--An audio recording and reproduction system with correlated information between two
channels (usually discrete channels), and meant to be heard over two or more loudspeakers to give the illusion of sound-
source localization and depth.

STEREO IMAGING: The ability of a stereo recording or reproduction system to form clearly defined audio images at
various locations between a stereo pair of loudspeakers.

SURROUND SOUND: A multichannel recording and reproduction system that plays sound all around the listener. The 5.1
surround system uses the following speakers: front-left, center, front-right, left-surround, right-surround, and subwoofer.

SUSTAIN: The portion of the envelope of a note in which the level is constant. Also, the ability of a note to continue
without noticeably decaying, often aided by compression.

SYNTHESIZER: A musical instrument (usually with a piano-style keyboard) that creates sounds electronically, and allows
control of the sound parameters to simulate a variety of conventional or unique instruments.

TRACK: A single channel of audio or MIDI.

WAVEFORM: A graph of a signal's sound pressure or voltage versus time. The waveform of a pure tone is a sine wave.

WAVELENGTH: The physical length between corresponding points of successive waves. Low frequencies have long
wavelengths; high frequencies have short wavelengths.

You might also like