A to D Converter or Analogue to Digital Converter is a stand alone piece of hardware or a computer interface that will digitize an analogue audio signal. The quality of the A to D converter will affect how accurately the digital audio compares to the original analogue signal.
AAC - see Compressed files
Ableton Live is a software music sequencer and digital audio workstation for OS X and Windows. In contrast to many other software sequencers, Live is designed to be an instrument for live performances as well as a tool for composing,recording, arranging, mixing and mastering. It is also used by DJs, as it offers a suite of controls for beatmatching, crossfading, and other effects used byturntablists, and was one of the first music applications to automatically beatmatch songs.
AIFF - AIFF (Audio Interchange File Format) is Apple Inc.'s uncompressed digital audio file format, and along with WAV's are currently the most commonly used uncompressed digital audio file formats.
An acronym for After Fade Listen, which is another way of saying post-fader solo function.
This is a type of distortion caused during the analog-to-digital conversion process. If the frequency of the analog signal exceeds one-half the sampling rate, spurious signals and harmonics not present on the original signal may be created (see Nyquist Theorem). Careful design and ﬁltering before the sampling stage can reduce this aliasing to a minimum.
In sound mixers, assign means to switch or route a signal to a particular signal path or combination of signal paths.
To reduce or make quieter.
Short for Auxiliary.
In sound mixers, supplemental equipment or features that provide additional capabilities to the basic system. Examples of auxiliary equipment include: serial processors (equalizers, compressors, limiters, gates) and parallel processors (reverberation and delay).
A mixer bus output designed to send a signal to an auxiliary processor or monitor system.
A mixer input (sometimes a pair of inputs) with limited control capabilities, intended for bringing the output of an auxiliary processor or other line-level source into the main mix bus. Aux returns can sometimes be assigned to other buses in the mixer.
Audio Codecs can be either hardware or, more commonly, software tools that can compress and decompress digital audio data to a given audio file format such as MP3, WMA, AAC, Ogg Vorbis, and FLAC. See Compressed files.
An input consists of two leads, neither of which is common to the circuit ground. This is a “differential pair”, where the signal consists of the difference in voltage between the two leads. Balanced input circuits can offer excellent rejection of common-mode noise induced into the line.
In a classic balanced audio circuit, the output is carried on two leads (high or + and low or -) which are isolated from the circuit ground by exactly the same impedance.
A symmetrical balanced output carries the same signal at exactly the same level but of opposite polarity with respect to ground. A special case of a balanced output carries the signal on only one lead, with the other lead being at zero voltage with respect to ground, but at the same impedance as the signal-carrying lead. This is sometimes called impedance balanced.
The band of frequencies that pass through a device with a loss of less than 3 dB, expressed in Hertz or in musical octaves. Also see Q.
The smallest component of a digital word, represented by either a one or a zero.
Bit Depth is the number of bits contained in each sample of digital audio. Standard audio CD's are 16 bit (with a Sampling Frequency of 44.1kHz) while other formats such as DVD-A can contain 24 bit samples (at 96kHz sampling rates). As a general rule the lower the bit depth (and sampling frequency) the lower the quality of (PCM) digital audio.
A binary number containing 16 0's or 1's is equivalent to 216 or a range of 65536 decimal states per sample while a binary number containing 24 0's or 1's is equivalent to 224 or a range of 16777216 decimal states per sample. Therefore files with higher bit depths have higher file sizes and require greater storage capacity.
Bit Rate is the rate at which the bits of a digital audio file are transferred or processed per unit of time. The bit rate is measured in bits per second (bps) and generally prefixed with the SI standard kilo (kbps) or Mega (Mbps). Files with higher bit rates are of a greater quality, have higher file sizes and therefore require greater storage capacity.
BLUE Book is the technical term for an Enhanced CD (ECD) a 'multi-session' CD containing audio data which plays in a CD player and video/data content which is accessible via one's computer.
Blu-Ray is a relatively new optical disc storage media. It has the same physical size as a standard CD or DVD though a single layer Blu-Ray disc can store up to 25GB of data.
An electrical connection common to three or more circuits. In mixer design, a bus usually carries signals from a number of inputs to a mixing ampliﬁer, just like a city bus carries people from a number of neighborhoods to their jobs. It comes from the British “omnibus”.
A manufacturer of electrical connectors who ﬁrst popularized the three-pin connector now universally used for balanced microphone connections. In sound work, a Cannon connector is taken to mean a Cannon XLR-3 connector or any compatible connector. You can tell an audio geezer because he refers to this connector as “Cannon”. Today the term “XLR” is more common.
A functional path in an audio circuit: an input channel, an output channel, a recording channel, the left channel and so on.
The physical realization of an audio channel on the front panel of a mixer; usually a long, vertical strip of controls.
A time-based effect available in some digital delay effects units and reverbs. Chorusing involves a number of moving delays and pitch shifting, usually panned across a stereo ﬁeld.
Checksum is a way to determine the integrity of data sent across a network or via media. A checksum is generated at source and transferred with the subject data while another check is carried out upon delivery and any discrepancies with the source checksum indicates the data has become corrupted. If the sums are the same the data has been transferred successfully.
A form of severe audio distortion that results from peaks of the audio signal attempting to rise above the capabilities of the ampliﬁer circuit. Seen on an oscilloscope, the audio peaks appear clipped off. To avoid clipping, reduce the system gain in or before the gain stage in which the clipping occurs. Also see headroom.
This is a dynamics processor used to smooth out any large transient peaks in an audio signal that might otherwise overload your system or cause distortion. The amplitude threshold and other parameters such as attack time, release time, and tire pressure are adjustable.
Another term for the electronic component generally known as a capacitor. In audio, condenser often refers to a type of microphone that uses a capacitor as the sound pickup element. Condenser microphones require electrical power to run internal ampliﬁers and maintain an electrical charge on the capacitor. They are typically powered by internal batteries or “phantom power” supplied by an external source, such as a mixing console.
Compressed files are files that have been reduced in size to make them easier to store, copy, email, etc. There are generally two types of file compression; 'lossless' and 'lossy'.
FLAC and MLP are examples of lossless data compression where the uncompressed WAV or AIFF can be reduced in file size by 50% with no degradation in audio fidelity. Similarly Zip/Unzip/.SIT are forms of lossless data compression which we use at QS Sound Lab Online Mastering to facilitate the upload of multiple titles in one hit.
MP3 and AAC are examples of lossy compressed files where "unwanted" data from the uncompressed WAV or AIFF is discarded sometimes reducing file sizes to 10% of the original (subject to the encoded file's Bit Rate) resulting in significant degradation of audio fidelity. They may be quicker to upload however due to this compromise in audio fidelity we don't accept lossy compressed files at QS Sound Lab Online Mastering.
Another common file compression type, developed by Microsoft, is WMA, (Windows Media Audio) which can be both lossless and lossy subject to which Windows Audio Codec is used.
Compression (Audio) is the reduction of an audio signal's. Dynamic Range by altering the ratio of input level to output level starting at a defined threshold.
Another term for a sound mixer, usually a large desk-like mixer.
The ratio of the peak value to the RMS value. Musical signals can have peaks many times higher than the RMS value. The larger the transient peaks, the larger the crest factor.
In broadcast, stage and post-production work, to “cue up” a sound source (a record, a sound effect on a CD, a song on a tape) means to get it ready for playback by making sure you are in the right position on the “cue,” making sure the level and EQ are all set properly. This requires a special monitoring circuit that only the mixing engineer hears. It does not go out on the air or to the main mixing buses. This “cueing” circuit is the same as pre-fader (PFL) solo on a Mackie mixer, and often the terms are interchangeable.
Cubase is a widely used music production software program developed by Steinberg. If you use Cubase and would like to use QS Sound Lab Online Mastering for your mixed tracks then export them and/or record them as stereo WAV's or AIFF's and try us out.
D to A Converter or Digital to Analogue Converter is a stand alone piece of hardware or a computer interface that converts a digital audio signal for use in the analogue world i.e. our ears. The quality of the D to A converter will affect how accurately the analogue signal compares to its digital counterpart.
DAW or a Digital Audio Workstation is a computer based system used, in conjunction with proprietary hardware, to record and manipulate audio and MIDI data. DAW's popular at QS Sound Lab are Pro Tools, SADiE, Pyramix and Sonic SoundBlade.
DDP - Disc Description Protocol is a type of image file used in the production of audio CD's. It includes files such as the audio content of the disc as well track timings and any other additional information. CD manufacturing houses accept DDP images and can be stored on a CD/DVD ROM or easily transferred over a network.
The opposite of peaking, of course, used in audio to describe the shape of a frequency response curve. A dip in an EQ curve looks like a valley, or a dip. Dipping with an equalizer reduces a range of frequencies. (See guacamole.)
Digital Performer is a commonly used DAW made by MOTU (Mark of the Unicorn).
This is an interesting technique to reduce the audibility of low level noise in a digital recording. Low level random noise is added to the analog signal before the sampling stage, reducing an effect called quantization error.
DMM or Direct Metal Mastering pertains to a specific method of cutting to copper rather than Lacquer discs for the production of vinyl masters. Copper masters generally produce a clearer sound while lacquer masters are preferred for dance music due to their warmer bass response. QS Sound Lab Online Mastering has two DMM cutting lathes and one lacquer lathe.
Dolby Digital (DD or AC3) is a type of data rate reduction algorithm, not dissimilar to DTS, and is usually used for stereo or surround sound audio on standard DVD-V's.
A delay effect, where the original signal is mixed with a medium (20 to 50 ms) delayed copy of itself. When used carefully, this effect can simulate double-tracking (recording a voice or instrument twice).
DSD (Direct Stream Digital) is a type of digital audio stream developed by Sony and Philips and is used on SACD's (Super Audio Compact Discs). Where PCM audio on standard CD's have 16 bit samples at a Sampling Frequency of 44.1kHz, DSD audio uses 1 bit samples at a rate of 2.8224MHz i.e. 1 bit x 64 x 44.1kHz).
DTS is a type of data rate reduction algorithm, not dissimilar to Dolby Digital, and is usually used for surround sound audio on standard DVD-V's. DTS arguably sounds better than its Dolby Digital counterpart although both DTS and Dolby now provide a lossless encoder for use on the Blu-Ray.
Usually means without reverberation, or without some other applied effect like delay or chorusing. Dry is not wet, i.e., totally unaffected.
Digital Signal Processing can accomplish the same functions found in analog signal processors, but performs them mathematically in the digital domain, with more precision and accuracy than its analog counterpart. Since DSP is a software-based process, parameters and processing functions are easily changed and updated by revising the software, rather than redesigning the hardware. DSP can be found in an outboard
effects device, such as a reverb or delay unit, or it can be integrated into a DAW or digital mixing console.
A mode of operation for a stereo ampliﬁer that routes a single input to both channels, but still allows independent level control over each ampliﬁer output.
DVD-A is a digital format for delivering high-fidelity audio content on a DVD. DVD-Audio is not intended to be a video delivery format and is not the same as video DVDs containing concert films or music videos. Read More...
Dolby Digital or DTS. Moreover DVD-A's use a lossless type of data rate reduction known as MLP (Meridian Lossless Packing) to pack more information on the disc. It is also possible to have a DVD hybrid disc containing both DVD-V and DVD-A content.
DVD (Digital Versatile Disc or Digital Video Disc) is the type of DVD usually used for movies with audio, and graphics. All the surround sound audio must be data rate reduced (see Dolby Digital or DTS) while stereo sound can be at the full data rate with no reduction. Read More…
The class of microphones that generate electrical signals by the movement of a coil in a magnetic ﬁeld. Dynamic microphones are rugged, relatively inexpensive, capable of very good performance and do not require external power.
A type of processor that only affects the overall amplitude level of the signal (sometimes as a function of its frequency content), such as a compressor, expander, limiter, or gate.
Dynamic range of a piece of music is the difference between the loud parts and the quiet parts. Many modern recordings are, at the artist's or producer's request, mastered as loud as possible (so they stand out on your iPod) however this is at the expense of a track's dynamic range.
An EAN-13 barcode (originally European Article Number, but now renamed International Article Number even though the abbreviation EAN has been retained) is a 13 digit (12 data and 1 check) barcoding standard which is a superset of the original 12-digit Universal Product Code (UPC) system developed in the United States.1 The EAN-13 barcode is defined by the standards organization GS1.
The EAN-13 barcodes are used worldwide for marking products often sold at retail point of sale. The numbers encoded in EAN-13 bar codes are product identification numbers, which are also called Japanese Article Number (JAN) in Japan. All the numbers encoded in UPC and EAN barcodes are known as Global Trade Item Numbers (GTIN), and they can be encoded in other GS1 barcodes.
The reﬂection of sound from a surface such as a wall or a ﬂoor. Reverberation and echo are terms that are often used interchangeably, but in audio parlance a distinction is usually made: echo is considered to be a distinct, recognizable repetition (or series of repetitions) of a word, note, phrase or sound, whereas reverberation is a diffuse, continuously smooth decay of sound. Echo and reverberation can be added in sound mixing by sending the original signal to an electronic (or electronic/acoustic) system that mimics natural echoes, and then some. The added echo is returned to the mix through additional mixer inputs.
An external signal processor used to add reverb, delay, spatial or psychoacoustic effects to an audio signal. An effects processor may be usedas an insert processor (serial) on a particular input or subgroup, or it may be used via the aux send/return system (parallel). See also echo, reverb.
Equivalent Input Noise. A speciﬁcation that helps measure the “quietness” of a gain stage by deriving the equivalent input noise voltage necessary to obtain a given preamp’s output noise. Numerically, it’s the output noise at a given gain setting minus the gain. EIN is usually measured at maximum gain and typically ranges from -125 to -130 dBm.
Electro-Magnetic Interference. This refers to current induced into the signal path as a result of an external magnetic ﬁeld. In audio systems, this is usually manifested as a 60 Hz or 120 Hz hum or buzz (50 Hz or 100 Hz in 50 Hz systems). The source of this noise can be from a ground loop or from the signal wire coming too close to a strong magnetic ﬁeld such as a transformer or high-current linecord.
Short for equalization.
A graph of the response of an equalizer, with frequency on the x (horizontal) axis and amplitude (level) on the y (vertical) axis.
types and effects are often named after the shape of the graphed response curve, such as peak, dip, bell, shelf, or notch.equalizationEqualization (EQ) refers to purposefully changing the frequency response of a circuit, sometimes to correct for previous unequal response (hence the term, equalization), and more often to boost or cut the level at certain frequencies for sound enhancement, to remove extraneous sounds, or to create completely new and different sounds. treble controls on your stereo are EQ; so are the units called parametrics and graphics and notch ﬁlters.
A lot of how we refer to equalization has to do with what a graph of the frequency response looks like. A ﬂat response (no EQ) is a straight line; a peak looks like a hill, a dip is a valley, a notch is a really skinny valley, and a shelf looks like a plateau (or a shelf). The slope is the grade of the hill on the graph.Aside from the level controls, EQs are probably the second most powerful controls on any mixer (no, the power switch doesn’t count!).
Another name for an audio level control. Today, the term refers to a straight-line slide control rather than a rotary control.family of curvesA composite graph showing on one chart several examples of possible EQ curves for a given equalizer or equalizer section.
A simple equalizer designed to remove certain ranges of frequencies. A low-cut ﬁlter (also called a high-pass ﬁlter) attenuates frequencies below its cutoff frequency. There are also highcut (low-pass) ﬁlters, bandpass ﬁlters, which cut both high and low frequencies but leave a band of frequencies in the middle untouched, and notch ﬁlters, which remove a narrow band but leave the high and low frequencies alone.
A term for an effect similar in sound to phasing. Before we had electronic delay units, ﬂanging was accomplished by playing two tape machines in synchronization, then delaying one slightly by rubbing a ﬁnger on the reel ﬂange.
FL Studio (formerly known as FruityLoops2) is a digital audio workstation developed by the Belgian company Image-Line. FL Studio features a graphical user interface based on a pattern-based music sequencer.
The number of times an event repeats itself in a given period of time. Generally the time period for audio frequencies is one second, and frequency is measured in cycles per second, abbreviated Hz, honoring the physicist Dr. Heinrich Hertz (who did not invent the rental car). One Hz is one cycle per second. One kHz (kilohertz) is 1000 cycles per second.The audio frequency range is generally considered to be 20 Hz to 20, 000 Hz. This covers the fundamental pitch and most overtones of musical instruments.
FTP or File Transfer Protocol is a method of transferring data from one computer to another across the internet or through a network. QS Sound Lab Online Mastering uses a secure form of FTP when receiving and returning tracks.
The measure of how much a circuit ampliﬁes a signal. Gain may be stated as a ratio of input to output voltage, current or power, such as a voltage gain of 4, or a power gain of 1.5, or it can be expressed in decibels, such as a line ampliﬁer with a gain of 10 dB.gain stage. An ampliﬁcation point in a signal path, either within a system or a single device. Overall system gain is distributed between the various gain stages.
A dynamics processor that automatically turns off an input signal when it drops below a certain level. This can reduce the overall noise level of your mix by turning off inputs when they are not in use. Threshold, attack time, hold, and release time are some of the adjustable gate parameters.
GarageBand is a software application developed by Apple for Mac OS X. If you use GarageBand and would like to use QS Sound Lab Online Mastering for your mixed tracks then record/export them as stereo WAV's or AIFF's and try us out.
A graphic equalizer uses slide pots for its boost/cut controls, with its operating frequencies evenly spaced through the audio spectrum. In a perfect world, a line drawn through the centers of the control shafts would form a graph of the frequency response curve. Or, the positions of the slide pots give a graphic representation of boost or cut levels across the frequency spectrum. Get it?
Also called earth. Ground is deﬁned as the point of zero voltage in a circuit or system, the reference point from which all other voltages are measured. In electrical power systems, ground connections are used for safety purposes, to keep equipment chassis and controls at zero voltage and to provide a safe path for errant currents. This is called a safety ground. Maintaining a good safety ground is essential to prevent electrical shock. Follow manufacturer’s suggestions and good electrical practices to ensure a safely grounded system. Never remove or disable the grounding pin on the power cord.In sensitive electronic equipment, tiny currents and voltages riding on the ground (so it’s not truly zero volts) can cause noise in the circuits and hamper operation. Often a ground separate from the power ground is used as the reference point for the electronics, isolating the sensitive electronics from the dirty power ground. This is called a technical ground. Quality audio equipment is designed to maintain a good technical ground and also operate safely with a good safety ground.
A ground loop occurs when the technical ground within an audio system is connected to the safety ground at more than one place. This forms a loop around which unwanted current can, and does ﬂow, causing noise in the audio 9system. Never disable the safety ground in an attempt to solve hum problems.
A psychoacoustic effect in which the time of arrival of a sound to the left and right ears affects our perception of direction. If a signal is presented to both ears at the same time and at the same volume, it appears to be directly in front of us. But if the signal to one ear, still at the same volume, is delayed slightly, the sound appears to be coming from the earlier (nondelayed) side.
The difference between nominal operating level and peak clipping in an audio system. A mixer with a nominal operating level of +4 dBu and a maximum output level of +22 dBu has 18 dB of headroom. Plenty of room for surprise peaks.
The unit of frequency, equal to 1 cycle per second. Abbreviated Hz. kHz 1000 Hz, and is usually pronounced “kay”(with “Hertz” implied) by sound professions who ask for “a little more two and a half K” when they want you to boost 2.5 kHz.
Short for Hertz.
ICPN (International Code Product Number) is a unique 13 digit number record companies use to identify a physical audio product. For example the ICPN for the CD version of an artist's album will differ from the ICPN for the vinyl version.
The A.C. resistance, capacitance, and inductance in an electrical circuit, measured in ohms. In audio circuits (and other ac circuits) the impedance in ohms can often be much different from the circuit resistance as measured by a dc ohmmeter. Maintaining proper circuit impedance relationships is important to avoid distortion and minimize added noise. Mackie input and output impedances are set to work well with the vast majority of audio equipment.
A holdover from the days when the only way that real consoles were built was in modular fashion, one channel per module. See channel strip.
Noun – a place where a signal path can be broken and a processing device placed in line with the signal. It’s usually a TRS jack with one conductor being an output (send) and the other being an input (return). The jack is wired with a normalled connection so that with nothing plugged in, the send and return are connected together, as if it wasn’t even there.
ISRC Code or the International Standard Recording Code is a 12 digit code that uniquely identifies a sound recording used anywhere in the world. They are used to facilitate the logging of radio plays and therefore royalty collections. In the UK in order to obtain ISRC's you need to become a member of the PPL (Phonographic Performance Limited) who can be found at www.ppluk.com.
A knee is a sharp bend in a curve (an EQ frequency response or compressor gain curve) not unlike the sharp bend in your leg.
Lacquer indicates the material used during the production of vinyl mastering when cutting to lacquer rather than copper (DMM) discs. Copper masters generally produce a clearer sound while lacquer masters are preferred for dance music due to their warmer bass response. QS Sound Lab Online Mastering has two DMM cutting lathes and one lacquer lathe.
Limiting is a type of compression, with a high, usually 10:1 or greater, compression ratio i.e. a 10dB change at the input will cause only a 1dB change at the output. Limiting with a compression ratio of infinity: 1 is known as Peak limiting (also called brick wall limiting).
Another word for signal voltage, power, strength or volume. Audio signals are sometimes classiﬁed according to their level. Commonly used levels are: microphone level (-40 dBu or lower), instrument level (-20 to -10 dBu), and line level (-10 to +30 dBu).
A signal whose level falls between -10 dBu and +30 dBu.
LOGIC Pro, Logic Express and Logic Studio are popular DAW's currently owned by Apple Inc. If you use Logic and would like to use QS Sound Lab Online Mastering then export your mixed tracks as stereo WAV's or AIFF's and try us out.
Lossless Compression - see Compressed files
Lossy Compression - see Compressed files
Short for main or house speakers in a sound reinforcement system.
A control affecting the ﬁnal output of a bus on which one or more signals are mixed. A mixer may have several master controls, which may be slide faders or rotary controls.
Mastering is the final creative stage of the audio recording process and generally includes EQ'ing, compressing, limiting, and assembling individual tracks to make a final product. When there is more than one track the final running order and the gaps between the tracks may need to be determined by the artist. Even in these days of shuffling iPods there are still some people who listen to an album from start to finish!
MD5 Checksum - see Checksum
See mic preamp.
The typical level of a signal from a microphone. A mic level signal (usually but not always coming from a microphone) is generally lower than -30 dBu. With a very quiet source (a pin dropping?) the signal can be -70 dBu or lower. Some microphones, notably vintage or vintagestyle condenser mics, deliver a higher signal level than this for the same sound pressure level. A “hot” mic output level isn’t necessarily a measure of the microphone’s quality, it’s just an option that the designer chose.
Short for microphone preampliﬁer. An ampliﬁer whose job is to bring the very low microphone level signal up to line level, or in the case of a mic preamp built into a mixer, the mixer’s internal operating level (approximately 0 dBu). Mic preamps often have their own volume control, called a trim control, to properly set the gain for a particular source. Setting the mic preamp gain correctly with the trim control is an essential step in establishing good signal-tonoise ratio and sufﬁcient headroom for your mix.
MIDI (Musical Instrument Digital Interface) has since 1983 been an industry standard protocol that enables electronic musical instruments, computers, etc. to communicate, control and synchronise with each other. MIDI does not transmit an audio signal moreover it transmits time related event messages such as note value and velocity which can be recorded, manipulated if necessary then played back.
An electronic device used to combine various audio signals into a common output. Different from a blender, which combines various fruits into a common libation.
MLP (Meridian Lossless Packing) - see DVD-A
MP3 - see Compressed Files
Long for mono. Literally, pertaining to or having the use of only one ear. In the audio ﬁeld, monaural describes a signal or system which carries audio information on a single channel with the intent of reproducing it from a single source. One microphone is a mono source; many microphones mixed to one channel is a mono mix; a stereo (or, to be picky, a two-channel) mix of many microphones panned left and right is a stereo mix of mono sources. Monaural listening, and therefore mono compatibility of a stereo mix, is more important than you may realize. Most people hear television audio in mono. Most clock radios are mono.
In sound reinforcement, monitor speakers (or monitor headphones or in-the-ear monitors) are those speakers used by the performers to hear themselves. In the video and broadcast world, monitor speakers are often called foldback speakers. In recording, the monitor speakers are those used by the engineer and production staff to listen to the recording as it progresses. In zoology, the monitor lizard is the lizard that observes the production staff as the recording progresses. Keep the lizard out of the mixer.
Short for monaural.
Short for multiple. In audio work, a mult is a parallel connection (in a patch bay or with specially built cables or wiring) used to feed an output to more than one input. A “Y” cable is a type of mult connection. Also used a verb, as in “Why did you mult the ﬂanger into every input in the board?”
Native Instruments is a technology company that develops software and hardware for music production and DJing. The company has originally been identified mostly with software instruments, but has also expanded to various other music equipment segments in recent years.
Current products of Native Instruments include software synthesizers, samplers and effect processors, sound libraries and emulations of acoustic instruments, groove production systems and audio interfaces, as well as various products for computer-based DJing that include DJ software, Digital Vinyl Systems, hardware controllers and specific DJ audio interfaces.
Whatever you don’t want to hear. Could be hum, buzz or hiss; could be crosstalk or digital hash or your neighbor’s stereo; could be white noise or pink noise or brown noise; or it could be your mother-in-law reliving the day she had her gallstone removed.
The residual level of noise in any system. In a well designed mixer, the noise ﬂoor will be a quiet hiss, which is the thermal noise generated by electrons bouncing around in resistors and semiconductor junctions. The lower the noise ﬂoor and the higher the headroom, the more usable dynamic range a system has.
A wiring method which electrically ties together two jacks or two poles of one jack so that in normal operation, there is signal ﬂow between them. Inserting a plug breaks this connection, allowing the signal path to be modiﬁed. Normal wiring is common in patchbays and insert jacks.
This theorem states that, when an analog signal is converted to a digital signal, it must be sampled at a frequency that is at least twice the highest audio frequency present in the analog signal. If the audio frequency should exceed one-half the sampling frequency, aliasing can result. Thus, if an analog-to-digital converter is sampling at 44.1 kHz, the audio signal should not exceed 22.05 kHz.12
ORANGE Book is the Philips/Sony set of specifications for CD-R's.
Acronym for Public Address. Today, people who work with PA systems like to say they’re working in “sound reinforcement”. See SR.
Short for panoramic potentiometer. A pan pot is used to position (or even dynamically move) a monaural sound source in a stereo mixing ﬁeld by adjusting the source’s volume between the left and right channels. Our brains sense stereo position by hearing this difference in loudness when the sound strikes each ear, taking into account time delay, spectrum, ambient reverberation and other cues.
A mode of operation for a stereo ampliﬁer that routes a single input to both channels, but combines the outputs of both channels into a single output by strapping the positivie output terminals together, thereby providing twice the current of an individual output.
A collection of usually a large number of jacks allowing convenient access to various points in a system’s interconnect wiring. A patchbay can make re-routing signals very convenient without having to ﬁsh around with cables in the back of racks or consoles. See spaghetti.
A “fully” parametric EQ is an extremely powerful equalizer that allows smooth, continuous, and independent control of each of the three primary EQ parameters: frequency, gain, and bandwidth. “Semi” parametric EQs allow control of fewer parameters, usually frequency and gain (i.e., they have a ﬁxed bandwidth, but variable center frequency and gain).
PCM (Pulse Code Modulation) is a digital representation of an analogue signal which has been sampled at regular intervals and is used for uncompressed digital audio in computers and on standard Red book CD's.
The opposite of dipping, of course. A peak is an EQ curve that looks like a hill, or a peak. Peaking with an equalizer ampliﬁes a band of frequencies.PFLAn acronym for Pre Fade Listen. Broadcasters would call it cueing. Sound folks call it being able to solo a channel with the fader down.
Peak Limiting or Brick Wall Limiting is used to make sound artificially loud. It can be likened to driving your VW beetle into the back wall of your garage where the rear end (quieter sounds) get closer to the front end (louder sounds), but the curve of the bonnet gets progressively squashed and arguably less attractive the more you crush it.
A system of providing electrical power for condenser microphones (and some electronic pickup devices) from the microphone input jack. The system is called phantom because the power is carried on standard microphone audio wiring in a way that is “invisible” to ordinary dynamic microphones. Mackie mixers use standard +48 volt DC power, switchable on or off. Most quality condenser microphones are designed to use +48 VDC phantom power. Check the manufacturer’s recommendations. Generally, phantom power is safe to use with non-condenser microphones as well, especially dynamic microphones. However, unbalanced microphones, some electronic equipment (such as some wireless microphone receivers) and some ribbon microphones can short out the phantom power and be severely damaged. Check the manufacturer’s recommendations and be careful!
The time relationship between two signals, expressed in degrees around a circle. 0 and 360 degrees represents an in-phase relationship – both signals change in the same way at the same time. Anything else is out of phase. 180 degrees out of phase is a special case which, for a continuous waveform, means that at any given time the two signals have the same ampli-13tude but are opposite in polarity. The two legs of a differential output are 180 degrees out of phase. The phase reverse switch found on some mixers or mic preamps actually reverses the signal polarity.When out-of-phase signals are mixed, there will be some cancellation at certain frequencies, the frequencies and the degree of cancellation being a function of the amount of phase shift and the relative amplitude of the signals. Attention to mic placement and careful listening will allow you to use this effect creatively.
A dynamic effect in which the phase relationship between the fundamental and overtone components of a sound is continually changing. This is done by passing the signal through an automatically sweeping ﬁlter. The effect is often simulated by mixing original signal with a delayed (1 to 10 ms) version of itself. The time of the delay is slowly varied, and the combination of the two signals results in a dramatic moving comb-ﬁlter effect. A comb ﬁlter can be found in your back pocket.
Ever see those old telephone switchboards with hundreds of jacks and patch cords and plugs? Or the plug on the end of a headphone cable? Those are phone jacks and plugs, now used widely with musical instruments and audio equipment. A phone jack is the female connector, and we use them in 1/4” two-conductor (TS) and three-conductor (TRS) versions.
The male counterpart to the phone jack, right above.
PMCD or Pre-master CD was originally used to describe a CDR which contained additional information designed to be used to speed up the manufacturing process. Now used to describe a CDR that has been checked and auditioned and has been passed as suitable to be used for manufacturing audio CDs
PMF or Pyramix Media Format is an audio file format exclusive to the Pyramix DAW.
Post Production in a professional audio facility is a catch all term for all the operations that can be carried out on audio once it has been mixed. At QS Sound Lab this includes Mastering (stereo and surround), QS Sound Lab Online Mastering, Re-Mastering (stereo and surround), Copying and bulk CD Duplication.
A term used to describe an aux send (or other output) that is connected so that it is affected by the setting of the associated channel fader. Sends connected this way are typically (but not always) used for effects. A post-fader output from a mixer channel usually is also post-EQ. If pain persists, see your mixer’s block diagram. Also see pre-fader.
In electronics, a variable resistor that varies the potential, or voltage. In audio, any rotary or slide control.
A term used to describe an aux send (or other output) that is connected so that it is not affected by the setting of the associated channel fader. Sends connected this way are typically (but not always) used for monitors (foldback). See post-fader.
The property of many directional microphones to accentuate their bass response when the source-to-mic distance is small, typically three inches or less. Singers generally like this effect even more than singing in the shower.
PreSonus Studio One is a line of digital audio workstation (DAW) software made by PreSonus Software, Ltd for Mac OS X and Microsoft Windows. It is used for music creation, recording, and mastering.
Pro Tools is a very common DAW used throughout professional recording studios and bedroom studios alike. If you use Pro Tools and would like to use QS Sound Lab Online Mastering then export your mixed tracks as stereo WAV's or AIFF's and try us out.
A way of stating the bandwidth of a ﬁlter or equalizer section. An EQ with a Q of .75 is broad and smooth, while a Q of 10 gives a narrow, pointed response curve. To calculate the value of Q, you must know the center frequency of the EQ section and the frequencies at which the upper and lower skirts fall 3 dB below the level of the center frequency. Q equals the center frequency divided by the difference between the upper and lower 3 dBdown frequencies. A peaking EQ centered at 10 14kHz whose -3 dB points are 7.5 kHz and 12.5 kHz has a Q of 2.
The digital representation of an analog signal involves sampling the amplitude of the signal at a fast rate. Quantization is the measurement of the amplitude at the time of each sample, expressed is a digital word. Where an analog signal will be continuous as if it were going up a smooth path, quantization will have discrete steps (similar to stair steps).
Reason is a very popular software music production application made by Propellerheads. If you use Reason and would like to use QS Sound Lab Online Mastering for your mixed tracks then record them as stereo WAV's or AIFF's and try us out.
Random Access Memory is a type of computer memory that can be read from and written to.
Long for RCA jack or phono jack. An RCA phono jack is an inexpensive connector (female) introduced by RCA and originally used to connect phonographs to radio receivers. The phono jack was (and still is) widely used on consumer stereo equipment and video equipment but was quietly fading into obscurity in the professional and semi-professional sound world. Then phono jacks began cropping up in early project-studio multitrack recorders, which (unfortunately) gave them a new lease on life. Since so many stereo recorders are ﬁtted with them, we decided we’d have to put a couple on our mixers for your convenience. But make no mistake: the only thing that the phono jack (or plug) has going for it is low cost.
The male counterpart to an RCA phono jack.
Also called recirculation. A delay effect created by feeding the output of a delay back into itself to cause a delay of the delay of the delay. You can do it right on the front panel of many effects units, or you can route the delay return back into itself on your mixer. Can be a great deal of fun at parties.Regeneration is also a fancy name for feedback. Feedback makes oscillators work and reduces distortion in ampliﬁers. Feedback in sound reinforcement systems, a form of oscillation itself, makes you popular with dogs and unpopular with musicians and audience alike.
RED Book is the Philips/Sony set of specifications for replicated (manufactured) audio CD's.
A return is a mixer line input dedicated to the task of returning processed or added sound from reverb, echo and other effects devices. Depending on the internal routing of your mixer and your own inclination, you could use returns as additional line inputs, or you could route your reverb outputs to ordinary line inputs rather than the returns.
The sound remaining in a room after the source of sound is stopped. It’s what you hear in a large tiled room immediately after you’ve clapped your hands. Reverberation and echo are terms that are often used interchangeably, but in audio parlance a distinction is usually made: reverberation is considered to be a diffuse, continuously smooth decay of sound, whereas echo is one or more distinct, recognizable repetitions of a word, note, phrase or sound which decreases in amplitude with every repeat. Reverberation and echo can be added in sound mixing by sending the original sound to an electronic (or electronic/acoustic) system that mimics natural reverberation, or worse. The added reverb is returned to the blend through additional mixer inputs. Highly reverberant rooms are called live; rooms with very little reverberation are called dead. A sound source without added reverb is dry; one with reverb or echo added is wet.
Radio Frequency Interference. High frequency radiation that often results from sparking circuits. This can be manifested in a number of ways in audio systems, but is usually evident as a high-frequency buzz or hash sound.
Read only memory is a type of computer memory that cannot be written to, but only read from.
An acronym for root mean square, a conventional way to measure the effective average value of an audio signal or other AC voltage. Most AC voltmeters are calibrated to read RMS volts, though on many meters that calibration is accurate only if the waveform is sinusoidal.
A measure of the relative liveness of a room. A low Sa means a very live room, and a high Sa means a dead room. S = the total surface area of the room, and a = the average absorption coefﬁcient of all the surfaces.
SACD (Super Audio Compact Disc) is an optical audio disc format developed by Sony/Philips aimed at providing higher fidelity digital audio than CD. SACD's use the DSD data stream for this high quality audio while Hybrid SACD's contain two layers, one being conventional 16 bit 44.1kHz audio, the other being SACD. When played in a suitable player, the laser can focus on the layer required.
Sampling Frequency or Sample Rate is the number of times per second a continuous (analogue) audio signal is sampled to create a discrete (digital) signal. A sample of digital audio has a binary value with a predetermined Bit Depth. Standard audio CD's have sampling frequencies of 44.1kHz (i.e. 44,100 samples per channel per second with a bit depth of 16 bits) while other formats such as DVD-A can have 96kHz sampling frequencies (at a bit depth of 24 bits). As a general rule the lower the sampling frequency (and bit depth) the lower the quality of (PCM) digital audio.
A greater number of samples per second results in a wider frequency range. The highest frequency that can be recorded is approximately half the sampling frequency. Other sample rates used in professional recording environments are 48kHz (common when using video), 88.2kHz, 96kHZ, and 192khz (that’s 192,000 samples per second!). Therefore files with higher sampling frequencies have higher file sizes and require greater storage capacity.
Scarlet Book - the technical term for the SACD.
SDII or SD2 (Sound Designer 2) is, or was, DigiDesign's (Pro Tools) proprietary audio file format which has now been superceded by WAV's and AIFF's.
A term used to describe the output of a secondary mix of the input signals, typically used for foldback monitors, headphone monitors or effects devices. Mackie mixers call it an Aux Send.
A term used to describe the shape of an equalizer’s frequency response. A shelving equalizer’s response begins to rise (or fall) at some frequency and continues to rise (or fall) until it reaches the shelf frequency, at which point the response curve ﬂattens out and remains ﬂat to the limits of audibility. If you were to graph the response, it would look like a shelf. Or more like a shelf than a hiking boot. See also peaking and dipping.
This is a speciﬁcation that describes how much noise an audio component has compared to the signal. It is usually expressed in dB below a given output level.
A single-delay echo without any repeats. Also see echo.
Italian for alone. In audio mixers, a solo circuit allows the engineer to listen to individual channels, buses or other circuits singly or in combination with other soloed signals. sound reinforcement. A system of amplifying acoustic and electronic sounds from a performance or speech so that a large audience can hear clearly. Or, in popular music, so that a large audience can be excited, stunned, or even partially deafened by the tremendous ampliﬁcation. Means essentially the same thing as PA (Public Address).
That mess of wires and cables in the back of your rack and/or console. You really can tame this
An acronym for Sound Reinforcement, which refers to the process (or a system for) amplifying acoustic and electronic sounds from a performance or speech so that a large audience can hear clearly. Or, in popular music, so that a large audience can be excited, stunned or even partially deafened by the tremendous ampliﬁcation. The term “SR” is to
Just as a radian is an angular unit of measure in 2-dimensional space, so a steradian is an angular unit of measure in 3-dimensional space (solid angle).
Believe it or not, stereo comes from a Greek word that means solid. We use stereo or stereophony to describe the illusion of a continuous, spacious sound ﬁeld that is seemingly spread around the listener by two or more related audio signals. In practice, stereo often is taken to simply mean two channels.
Multi-channel audio playback systems in four, ﬁve, or six channel formats. Surround sound is typically found in movie theaters and home theater systems.
Stuffit is a family of software utilities for archiving and compressing files on Apple Mac computers. QS Sound Lab Online Mastering currently doesn't accept Stuffit files as the current OSX platform comes with the alternative Zip/Unzip utility.
An equalizer that allows you to “sweep” or continuously vary the frequency of one or more sections.
The ringing in the ears that often results from prolonged exposure to very loud sound levels. A sound in the ears, such as buzzing, ringing, or whistling, caused by volume knob abuse!
In audio mixers, the gain adjustment for the ﬁrst ampliﬁcation stage of the mixer. The trim control allows the mixer to accommodate the wide range of input signal levels that come from real-world sources. It is important to set the trim control correctly; its setting determines the overall noise performance in that channel of the mixer. See mic preamp.
Acronym for Tip-Ring-Sleeve, the three parts of a two-conductor (plus shield) phone plug. Since the plug or jack can carry two signals and a common ground, TRS connectors are often referred to as stereo or balanced plugs or jacks. Another common TRS application is for insert jacks, used for inserting an external processor into the signal path.
Acronym for Tip-Sleeve, the two parts of a single conductor (plus shield) phone plug. TS connectors are sometimes called mono or unbalanced plugs or jacks. A 1/4” TS phone plug or jack is also called a standard phone plug or jack.
An electrical circuit in which the two legs of the circuit do not have the identical impedance to ground. Often one leg is also at ground poten-17tial. Unbalanced circuit connections require only two conductors (signal “hot” and ground). Unbalanced audio circuitry is less expensive to build, but under certain circumstances is more susceptible to noise pickup.
A circuit or system that has its voltage gain adjusted to be one, or unity. A signal will leave a unity gain circuit at the same level at which it entered – no ampliﬁcation, but no loss either. In Mackie mixers, unity gain is achieved by setting all variable controls to the marked and usually detented “U” setting. Mackie mixers are optimized for best headroom and noise ﬁgures with all gain stages beyond the preamp set at unity gain.
The Universal Product Code (UPC) is a barcode symbology (i.e., a specific type of barcode) that is widely used in the United States, Canada, the United Kingdom, Australia, New Zealand and in other countries for tracking trade items in stores. Its most common form, the UPC-A, consists of 12 numerical digits, which are uniquely assigned to each trade item. Along with the related EAN barcode, the UPC is the barcode mainly used for scanning of trade items at the point of sale, per GS1 specifications.1 UPC data structures are a component of GTINs (Global Trade Item Numbers). All of these data structures follow the global GS1 specification which bases on international standards. Some retailers (clothing, furniture) do not use the GS1 System (other bar code symbologies, other article number systems). Other retailers use the EAN/UPC bar code symbology but without using a GTIN (for products brands sold at such retailers only).
The sound level in an audio system. Perhaps the only thing that some bands have too much of.
Acronym for Volts Root Mean Square. See RMS.
A signal with added reverberation or other effect like echo, delay or chorusing.
WAV or WAVE (Waveform Audio File Format) is an uncompressed digital audio file format, developed by Microsoft and IBM, and along with AIFF's are currently the most commonly used uncompressed digital audio file formats. A BWF (Broadcast Wave Format) is a WAV file that also carries time position information commonly used in conjunction with video content.
WinZip - see Zip/Unzip
WMA (Windows Media Audio) - see Compressed files.
A cable with one input and two outputs, used to mult a source to two inputs.
The electrical symbol for impedance.
Zip/UnZip or WinZip, developed for Microsoft, is a software utility for archiving and compressing files. Zip is a form of lossless compression so the integrity of the audio files remains intact.