Dubspot Producer’s HandbookWelcome 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 RecordingLet’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 RecordingsThe 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 RecordingMagnetic 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 RecordingThe 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 RecordingIn 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 & EngineersHere 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 & EffectsIf 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 4An 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 LIVEThe 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 choicefor 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 8This 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 AudioDigital audio recording works by recording, or sampling, an electronic audio signal atregular intervals (of time). An analog-to-digital (A/D) converter measures and storeseach sample as a numerical value that represents the audio amplitude at that particularmoment. Converting the amplitude of each sample to a binary number is calledquantization. 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 determiningthe quality of a digital audio system.

Sample RateThe 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 electronicsnapshots made of the sound wave per second. Higher sample rates result in highersound quality because the analog waveform is more closely approximated by thediscrete samples. Which sample rate you choose to work with depends on the sourcematerial you’re working with, the capabilities of your audio interface, and the finaldestination of your audio.For years, the digital audio sample rate standards have been 44,100 Hz (44.1 kHz) and48 kHz. However, as technology improves, 96 kHz and even 192 kHz sample rates arebecoming common.

Bit DepthUnlike analog signals, which have an infinite range of volume levels, digital audiosamples use binary numbers (bits) to represent the strength of each audio sample. Theaccuracy of each sample is determined by its bit depth. Higher bit depths mean youraudio signal is more accurately represented when it is sampled. Most digital audiosystems use a minimum of 16 bits per sample, which can represent 65,536 possiblelevels (24-bit samples can represent over 16 million possible levels).To better understand bit depth, think of each digital audio sample as a ladder withequally spaced rungs that climb from silence to full volume. Each rung on the ladder isa possible volume that a sample can represent, while the spaces between rungs arein-between volumes that a sample cannot represent.

Sample RatesWhen a sample is made, the audio level of the analog signal often falls in the spacesbetween rungs. In this case, the sample must be rounded to the nearest rung. The bitdepth of a digital audio sample determines how closely the rungs are spaced. The morerungs available (or, the less space between rungs), the more precisely the originalsignal can be represented.

Quantization errors occur when a digital audio sample does not exactly match theanalog signal strength it is supposed to represent (in other words, the digital audiosample is slightly higher or lower than the analog signal). Quantization errors are alsocalled 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-digitalconverter rounds this to 1 volt because this is the closest bit value available. Thisrounding error causes noise in your digital audio signal. While quantization noise maybe 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 audiosamples 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 silenceor full volume, and nothing in between. Any audio sample that falls between theserungs must be rounded to full volume or silence. Such a system would have absolutelyno subtlety, rounding smooth analog signals to a square-shaped waveform.

When the number of bits per sample is increased, each sample can more accuratelyrepresent the audio signal.

To avoid rounding errors, you should always use the highest bit depth your equipmentsupports. Most digital video devices use 16- or 20-bit audio, so you may be limited toone of these bit depths. However, professional audio recording devices usually support24-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.