24/96
Audio industry jargon. The phrase 24/96 is our current shorthand way of expressing digital audio data with a bit depth of 24-bits and a sample rate of 96 kHz. 24/96 has become a widely used industry term in part because the emerging DVD standard supports it, which is likely to make it the minimum standard for digital audio sometime in the future. For this and other reasons many equipment manufacturers are making digital equipment that is capable of 24/96 recording. Currently many other sample rates and bit-depths are in use, but 24/96 is being positioned by some companies as the "new" standard for audio in the new millennium.
A/D Converter (ADC)
An A/D (Analog to Digital) converter is an electronic device whose function is to convert analog voltages into a digital representation of electrical ones and zeros which can be stored, manipulated, and later retrieved or converted back to analog. In the domain of audio recording these converters are found built in to virtually all digital audio products such as DAT machines and digital signal processors. There are also a variety of high quality stand alone converter boxes which will accept line or mic level analog signals and output digital equivalents which can then be input directly into a digital device.
Acoustic Treatment
Acoustically treating a room is necessary in audio production due to the fact that very few "spaces" have the physical qualities that make for accurate monitoring or desired recording. There are many things that can be done to a space before and during construction to optimize its acoustic behavior. These include the shape of the space, its isolation, and the surface materials. Once a room is already constructed, Acoustic Treatment mostly tends to consist of treating the surfaces. There are two primary elements to consider: absorption and diffusion. Acoustic foam is well suited to alleviate slap and flutter echo, the two most common problems in rooms not specifically designed for music recording and performance. In fact, foam can turn even the most cavernous warehouse or gymnasium into a suitable acoustic environment. Diffusion keeps sound waves from grouping, so there are no hot spots or nulls in a room. In conjunction with absorption, diffusion can effectively turn virtually any space into one that is appropriate and useful for the purpose of recording or monitoring sound with a high degree of accuracy. ADAT Optical
A specific form of optical audio data transfer developed by Alesis for their ADAT machines. ADAT optical uses the same interconnects as the ubiquitous TosLink two channel format, but includes eight channels of digital audio data. ADAT optical was a key cog in the development of the original ADAT back in the early 1990s. Today it is one of the standard digital I/O connections found on many pieces of digital audio gear such as mixers and recorders.
ADR
Abbreviation for Automatic Dialog Replacement or Additional Dialogue Recording depending upon who you talk to. Either way it's a process most associated with film and video sound work. Basically it is just dubbing or overdubbing, done in addition to, or as a substitution for, captured location sound. Automatic refers to the process known as "looping," where the desired scene is looped so the actor can repeat the line many times quickly until he or she gets the performance exactly right. Otherwise ADR is somewhat of a mysterious term, but it has a certain appeal as it obscures the fact that dubbing was involved when it appears in the credits of a film. (See also Dubbing Session)
Aliasing
In digital sampling and recording, aliasing is digital distortion that occurs when the frequency being sampled is higher than one-half the sample rate (called the Nyquist Frequency). Essentially, when a frequency exceeds the Nyquist Frequency, it is "folded over" and becomes an audible component of the signal. Most digital recorders have filters, etc., to prevent aliasing from occurring. In samplers, aliasing also becomes apparent when a sample has been "stretched" too far in pitch...
Auto Punch
Short for Automatic Punch In. In order to make punching in and punching out easier and exact, many software and hardware DAWs as well as other recording systems allow users to set in and out recording points, which will cause the recorder to automatically go into and out of record mode at the appropriate times. Autolocate/Autolocator
In audio production parlance an autolocator is a device that is connected to a multitrack recorder to provide control over various functions. It is similar to a remote control in that it brings control of the recorder to a location convenient to the user, but goes beyond mere remote control in that it allows a number of functions to be automated. Specifics vary from machine to machine, but they almost all allow a variety of "locations" to be stored and recall for easy locating. For example, one could easily store the location of the first chorus of a song, making it easy to recall and have the machine 'locate,' or 'autolocate' to that spot later. In most production environments many locate points are stored as a project matures. This makes it easy for the engineer to later return to any point of concern without having to find it by guessing. Most autolocators make it easy to store locations on the fly so the engineer can mark spots for later attention while working on something else. Autolocators also generally provide easy ways to repeatedly play back a section of audio and provide a variety of automatic punch in and out features.
Automation
In audio production automation refers to having things programmed to happen in real time during a mixdown. In the 1970's, when big multitrack tape machines were becoming common, and overdubbing parts became a standard way of working, the process of getting a good mix became exponentially more difficult. No longer was the whole recording of a live performance where the musicians pretty much balanced their own levels. Many components were put in later and eventually it became trendy to do mixes at other studios optimized for that purpose, thereby causing the mix to have to be created from scratch. Anyone who has ever had the occasion to be one of the three or four people huddled over the mixer making adjustments during a manual mixdown can appreciate the benefits of being able to automate most of the process. Early automation systems were basic level controls. They were synchronized to the tape machine by some form of Time Code (not necessarily SMPTE) and would remember any moves the engineer made and then play the data back causing the level change to occur at the proper time (assuming the automation stayed in sync with the tape - not a given). They worked by either having motorized faders, where the motors could be controlled by the automation, or by using VCA's (Voltage Controlled Amp), which was a much less expensive and cantankerous option. VCA's, however, didn't sound as pure as the passive fader with a motor attached so most successful systems were "moving fader" based. Later the quality of the VCA based systems rose (while the cost declined) and they became popular among smaller studios, but moving fader systems are still considered the best choice for analog. Not only because they sound better, but because the tactile feedback of physically moving faders is something many engineers prefer. During the 1980's many other aspects of mixing began to be automated. Things like aux sends, panning, and eventually even EQ and compression could be put under computer control. Nowadays there are many analog mixing boards that are totally under digital control and virtually every parameter can be automated. Further, with the advent of the DAW, complete recall and automation of every aspect of a mix has become a standard.
Bit Depth
A sort of odd phrase that has come into use to describe how many bits a digital recording or digital device uses. In digital audio the sample rate (or sample frequency) will determine the upper limit of audio bandwidth that can be digitized. The number of bits in each sample determines the theoretical maximum dynamic range of the audio data regardless of sample rate. Each additional bit adds 6 dB to the dynamic range of the audio. Bit depth is just a phrase occasionally used to specify how many bits are being or have been collected in the data. A 16-bit recorder, for example, would produce digital files with a bit depth of 16. The "depth" concept arose out of illustrations describing the effect of more bits on an audio recording. If dynamic range is looked at as a vertical line, the upper limit of which is always fixed at 0dBFS, more bits produce a line that goes deeper into the noise floor and lower ranges of audio to be captured. More bits help capture quieter data more accurately.
Bouncing
The process of combining several tracks together and re-recording them onto another track is called bouncing. This is normally done to free up tracks for more recording. For example, you might have three background vocals recorded on tracks 1 to 3. By combining these tracks with a mixer, and routing them to track 5 (for example) tracks 1 to 3 can be erased and used to record other materials. Keep in mind that bouncing does require that you pre-decide on levels, EQ, etc. for the tracks being bounced - once they are combined, it is impossible to adjust their relative levels (overall level and EQ of the bounced track can, of course, still be adjusted).
Broadcast Wave File (BWF)
A special version of the standard .wav audio file format developed by the EBU in 1996 and commonly used by PC's. BWF's are designed to be a subset of standard .wav files. More restrictions exist about what types of audio data can be used (mainly linear PCM and MPEG encoded), plus there is an extra 'chunk' of data, known as the broadcast extension chunk (bext) that contains information on the title, origination, date, time, etc. of the audio content. A BWF file containing MPEG audio data also includes a further extra chunk they call 'mext.' One of the most important aspects of BWF's for us in the audio industry is the feature of time stamping. Time stamping allows files to be moved from one recording platform to another and easily aligned to their proper point in time.
Byte
In digital data bits are arranged into what is known as "words." A byte is a binary word that is 8 bits long. A kilobyte (Kb) contains 1024 bytes (not 1000). This is because 1024 is a power of 2, which is the basis for binary digital data.
Capture
Generically the act of recording audio or video can be thought of as capturing. Sampling is a form of capturing. But the word capture is often used in video circles to specifically refer to the actual process of getting video off of a tape-based system and into an editing system. The capture process usually starts with a process known as logging, which is where someone looks at all the raw footage and logs the sections that seem most promising for final production. These clips are then captured into the editing system, which can occur one-by-one while being logged, or in a separate operation known as a batch capture where all of the selected or logged clips from a given reel of tape are captured automatically by the system. CD-RW
Stands for CD rewritable, the latest incarnation of CD writing technology. CDRW is pretty much like regular WORM (see WFTD archive WORM) CD writers except the media can be overwritten with new data.CD-RW technology supports a variety of recording modes (disc at once, track at once, multisession, linked multisession, and packet writing). It also supports disc formats such as Red Book (CD audio), Yellow Book (CD-ROM, CD-XA), Green Book (CD-i), Blue Book (enhanced CD, CD Extra), White Book (Video CD), Orange Book Parts II and III (CD-RW), CD-ROM XA (Photo CD), mixed mode, hybrid, and bootable discs.
Comp/Comping
In musical terms comp is short for accompaniment. Comping is the act of playing an accompanying musical part. In recording these definitions apply, but it has also come to mean the process whereby a part is played or sung several times on different tracks (or virtual tracks) of a recorder and then the best pieces of those performances are later "comped" or combined into one ideal performance. It may be something as simple as keeping the second verse from the first take, using the chorus from the second take, and the rest of the performance coming from the fifth take. Or, it could be as involved as taking the first syllable from the fifth word of the vocal and splicing it onto the beginning of the same word each time it comes up in the song. This has been going on since the early days of multitrack recording, but now that DAW's make it so easy to cut and past parts, the process is pretty much considered a standard recording technique, and it's not uncommon to have dozens or even hundreds of small edits in a single vocal performance.
Compressor
A compressor is a device that reduces the dynamic range of an audio signal. First a threshold is established. When the audio signal is louder than this threshold, its gain is reduced. The amount of gain reduction applied depends on the compression ratio setting. For example, with a 2:1 ratio, for every 2 decibels the input signal increases, the output is allowed to increase only 1 decibel. A variety of other parameters in the compressor will also affect its performance processing specific signals; attack time, release time and others are very important.There are a variety of uses and applications for compressors, the most obvious one being to control the dynamic range of a live performance so that it will fit into the fairly narrow dynamic range of recorders, etc. Other applications include making a signal's average level higher, increasing the apparent sustain on a guitar, evening out a vocal or bass guitar performance, fattening up sounds, and on and on. The list of possibilities is extensive!
COSM
Abbreviation for Composite Object Sound Modeling. COSM is a powerful modeling technology that Roland premiered in 1995 with the VG-8 V Guitar System, and continues in the newer VG-88 system. It enables guitarists to emulate a range of classic and modern guitars, amps, cabinets, and microphones, plus it can produce "futuristic" synth-like tones. Today COSM can be found in keyboards, digital recorders, mixers, etc. It can model rotary effects, different speaker colorations, and can even approximate expensive microphones using just an ordinary dynamic mic. Its name comes from "composite object" because its core function revolves around breaking audio producing or reproducing devices down to their component parts and creating a set of instructions to emulate how these various parts interact with each other to produce a new composite that can be dynamically controlled. Of course, that's what all modeling is, but Roland coined this name to call attention to it.
De-Esser
A special type of compressor that is tuned to be sensitive to sibilant sounds, or sounds with high frequencies such as the sound produced by the letter "s", hence the name de-esser. The need for de-essing arises out of a combination of the presence peak many microphones have in their frequency response to accentuate vocal recording combined with close proximity vocal work and possible added high frequency boost from equalizers and tone controls. While these things often make a vocal track have more "air" and high-end clarity, they can also add enough accentuation to certain consonants (especially the "s") that they become too pronounced. The problem can range from being slightly annoying to being bad enough to cause distortion in the signal path. Many years ago broadcast engineers figured out they could tune compressors to be more sensitive to these frequencies, which in effect produces an automatic volume control that can turn down the audio anytime one of the sibilant sounds occur. In fact, any compressor with a sidechain input can be turned into a de-esser by inserting an EQ and boosting the offending frequencies. Even more flexibility comes from using a multi-band compressor. The de-essing action no longer has to lower the overall signal level. It can just lower the level in the specific range of frequencies specified. Some modern de-essers, however, have very sophisticated circuitry and controls that are optimized for achieving results beyond what would be easy with a simple compressor with an EQ in the sidechain.
Delay
In audio production, a delay is an electronic device designed to store a signal for a specified period of time and then release it, thereby delaying the signal relative to other parts of an audio program. Delays are often used to create echo effects, where a particular signal may repeat several times, with each repeat being lower in level than the prior one. This application goes back many years to a time when delays were only able to be accomplished with tape recorders (see WFTD Tape Delay). Later, analog circuitry was developed that could store signals long enough to be useful as a delay. This often involved a complex process of dumping the signal from one circuit with a finite amount of storage into another, and so on. Some of these early designs were thus known as "bucket brigade" circuits because they pretty much worked like an old time fireman's bucket brigade where water would be passed in buckets from person to person in a long line between the source and the fire. The big downside of these units was that it cost a lot to build one with enough bucket brigades to amount to any length of time, and the sound quality sometimes degraded as the signal was passed along, which was exacerbated when regeneration was applied to achieve multiple repeats. Ultimately delays became most flexible and useful when digital technology became practical. In a digital delay the signal is simply stored in memory chips until it is needed. The longer a signal needs to be stored, and the higher the sample rate and bit depth, the more memory is required, so early digital delays tended to suffer from some of the same problems as their analog counterparts. It wasn't long, however, before digital delays could capture a nearly perfect recording of a signal and store it for minutes if necessary. Delay technology is at the core of most time based effects such as flangers, chorus units etc. They have also been widely used in broadcast applications over the years to provide a few seconds of delay to "live" broadcasts. These few seconds can be used by an attentive engineer to "bleep" out things like curse words, etc. Dither
Literally, dither is noise added intentionally to a digital recording. Low level signals are difficult for digital gear to record; the sampling machine simply has difficulty deciding whether the necessary bits should be turned on or off, creating "quantization noise." By adding a small amount of very controlled noise to the original signal, the bits can be made to positively switch on or off, improving low level sound resolution. The noise used is often "shaped" to be in-offensive to human ears. Good dithering algorithms, whether hardware or software based, can make an incredible difference in the sound quality of a digital recording!
DVD
Latest info says "DVD" no longer stands for anything! It used to mean "digital versatile disc" - and before that it meant "digital video disc." A new type of 12-centimeter (4.72") compact disc (same size as audio CDs and CD-ROMs) that holds 10 times the information. Capable of holding full-length movies and a video game based on the movie, or a movie and its soundtrack, or two versions of the same movie - all in sophisticated discrete digital audio surround sound. The DVD standard specifies a laminated single-sided, single-layer disc holding 4.7 gigabytes, and 133 minutes of MPEG-2 compressed video and audio. It is backwards compatible, and expandable to two-layers holding 8.5 gigabytes. Ultimately two discs could be bounded together yielding two-sides, each with two-layers, for a total of 17 gigabytes. There are three versions: DVD-Video (movies), DVD-Audio (music-only) and DVD-ROM (games and computer use). The DVD-Audio standard is still being defined. Meanwhile a fourth member has joined the family: DVD-RAM defines specs for a rewritable system, opening the door for recording.
Dynamics
In written music, the gradation of volume in music. Long before we could actually measure the volume of sound, music masters defined the overall volume of a passage of music with written indications on sheet music. The following are the most commonly used dynamics:
ff (Fortissimo) - Very loud, or very strong.
f (Forte) - Loud, or strong.
mf (Mezzo forte) - Medium loud, medium strong or "half loud."
p (Piano) - Soft or quiet.
pp (Pianissimo) - Very soft or very quiet.
In music and sound recording, dynamics also refers to the dynamic range of the material. Audio passages with a large dynamic range are sometimes said to have a lot of dynamics. This is subjective and depends, to some extent, on the span of time between the loudest and softest passages.
Dynamics also refers to the general category of audio equipment designed to modify the dynamic range of audio in some capacity. This would include things like compressors, limiters, expanders, gates, etc. These are sometimes known as dynamics processors. Finalize
This is a word that gets used in many different contexts lately. One company has even used it to name a product. Up until recently, however, its relevance to us was mostly in the domain of recording CDs on a CD recorder. Finalization is the process where an Orange Book disc (one that is in process) is made into a Red Book CD, which is suitable for playback on any CD player. The idea being that finalization is the last step in the process, after which there is no turning back. Finalized WORM discs cannot be changed or updated in any way without rendering them useless. This is because the key ingredient in finalizing a disc is writing the TOC. Sometimes this process is also called "fix-up."
Frequency
Literally the number of times something occurs per unit of time. In the audio world the frequency of sound vibrations are directly related to what we hear as pitch, though the relationship is NOT linear. It is also inversely related to wavelength. We use the word frequency, and the values associated with it, as an objective way to speak about sound characteristics. Saying a unit has a frequency response of 20 Hz to 20kHz is much more accepted than specifying the response in terms of pitch.
House Sync
See Video Sync
House Sync
Refers to a distributed synchronization signal available to all recording equipment in a studio. In modern production environments it's important to have all digital and/or video equipment in a studio synchronizing its clock or frame rate to a common, stable source. This makes synchronization among different recorders (such as video and audio machines) much easier and more consistent, plus it enables material to more easily be transferred from one to another. Of course, in digital systems good word clock must be transferred between every digital device (mixer, recorder, DAW, CD recorder, etc.) to keep the signal in tact. The same results can be achieved, however, if all devices are synchronized to a common clock. Historically house sync was just another word for video sync or black burst, which have been common in video houses for years. However, with digital equipment becoming so widespread most studios are now distributing word clock signals instead of, or at least in addition to, black burst. In studios where video and digital equipment is integrated, both the word clock and video frame rate signal (black burst) must be resolved to one another. Of course, if there are any analog tape machines then standard LTC (or sometimes VITC for video machines), usually in the form of SMPTE, must be used, and that too should be resolved to the video and/or digital clock rate. Many modern house sync generators and distribution systems now deliver digital word clock and black burst. Interleaved Stereo
A stereo sound file or digital recording where the data making up the left and right channels are mixed together as one contiguous block of data. Interleaved stereo files are common in the DAW world, but are also utilized in R-DAT recorders, and other digital tape machines.
Jitter
In a digital recorder or sampler, errors in the timing of sample acquisition due to rapid amplitude changes is called jitter (also known as Sample Offset Uncertainty). Jitter introduces some distortion and phase discrepancy into an audio signal. Higher frequencies are more susceptible to jitter than lower. In most modern gear, use of clocked buffers for digital data streams has minimized the effects of jitter.
Limiter
A limiter is a dynamics processor very similar to a compressor (see inSync WFTD 10/13). In fact, many compressors are capable of acting as limiters when set up properly. The primary difference is the ratio used in reducing gain. In a limiter, this ratio is set up to be as close to infinity:1 as possible (no matter how much the input signal changes, the output level should remain pretty much constant). The idea is that a limiter establishes a maximum gain setting, and prevents signals from getting any louder than that setting.Like compressors, limiters are used for a variety of applications. A few: Maximizing signal levels while preventing distortion when using digital recorders, preventing overload in a signal chain, setting a maximum volume level to protect users of in-ear monitors, protecting speakers and amplifiers from clipping, and so on. Any time you want to establish a maximum gain setting and prevent signals from passing it, a limiter is your tool of choice!
LocatePoint
As found on most multitrack recording units, DAW systems, or MIDI sequencing software, a locate point is a "pointer in time" marked or saved at a specific time in your recording or sequence so that you can move to that point in the session quickly. This makes navigating to specific parts of your song a great deal easier. For example, you could set locate points for the beginning of each chorus so that you could jump to the different parts of the song very quickly. In some systems much more information can be stored as part of locate points, including screen attributes such as zoom settings, etc.
Loop
Common sense probably gives one an idea what a loop is. In audio it is a sound that continually repeats itself over and over again. It is called a loop because back in the "old days" tapes were used. One could cut a section of tape out with a sound or passage and connect the end of it back to the beginning and form a physical loop. The audio would repeatedly play over and over in a tape machine. This innovation was used by companies who manufactured tape based looping delay or echo units such as the Echoplex and Roland Space Echo. The 8-track tapes of the 1970's were an endless loop system. They never had to be rewound. In digital keyboards "looping" became a standard way to allow the sustained part of a sound to decay without having to have an actual (memory hungry) recording of the entire decay of the sound. Most sustained musical sounds fall into a relatively static state after a second or two. At that point it is possible to loop the static portion of the sound and have it play over and over while a VCA (or pick the name of this function in your synth) cause it to fade away simulating the actual decay of the instrument. "Looping" is the process of finding good loop points in sounds so they can be made to take on desired characteristics while looped. Often it is desired to loop sample so it accurately recreates some instrument, which can be quite difficult. In the 1980's and 90's looping became an art form and quite often loops would make or break sounds used in digital instruments. In recent years audio loops of entire musical passages have become very popular. A looped rhythm section, for example, can be a great foundation for another tune or arrangement.
Mastering
Though used (too) generically in our industry, this word can mean many subtly different things. The most common (and correct) understanding is that mastering is the process where recorded material is taken from a "master tape" and prepared for duplication in the format of the final release media. Historically this originated with transferring material from tapes to an acetate master disk, which was the first process in making phonograph records. The entire process was as much an art as a science, and as production quality and technology advanced, many practices were developed that helped make better and better sounding records. This advancement included many potential processes of the audio signal such as equalization, compression, limiting, normalization, widening the stereo image, editing fades, and just putting the songs in the correct order. They started out as simple tweaks that had to be done to make audio play correctly on vinyl records, but producers learned that a good mastering engineer could be the difference in how an album ended up sounding as a whole. A well mastered record was better and more consistent in terms of levels and tone quality, which became more and more important as radio airplay and home hi-fi systems became more prevalent in our society.Nowadays, with the convenience, quality, and affordability of digital audio equipment, many of these steps are done in the recording studio (home or otherwise) where the material was recorded. A significant percentage of the equipment sold at Sweetwater Sound is for this purpose and is being purchased by beginners and pros alike. Still, however, the last few steps of the mastering process, which is very different for CD's than it was for LP's (see WFTD archive LP), are done after the material leaves the studio. Sometimes an actual mastering house is used, and other times the "mastered" material is sent directly to a duplication facility where they can also do the final few steps. The lines between how much of it are done in the studio, versus a mastering house, versus the duplication house are very blurry at this point.More in depth info on mastering can be found at the following Web sites:
MIDI Machine Control (MMC)
A part of the MIDI spec that allows MIDI devices to control hardware devices, MIDI Machine Control is commonly used to send transport control messages to hardware recorders. Play, Stop, and Locate are examples of MMC messages. Note that MMC does not include synchronization information, although MIDI sync info could also be sent to/from the device that MMC is addressing. MMC allows you to centralize control of your studio from a MIDI source (often a sequencer). A common scenario: Pressing play on a MIDI seqencer sends an MMC play message to a connected multitrack recorder, which begins playing. As the deck plays, it generates MIDI Time Code (MTC) which the sequencer then synchronizes to (chases). When "stop" is pressed on the sequencer, the deck also stops, and ceases to send out MTC. When MTC stops, the sequencer stops chasing. Locating to a point within the sequence will cause the deck to fast forward or rewind to the corresponding location on tape.
Mixdown
The process of mixing a multi-track recording down to a lower number of tracks. Traditionally this has always been two tracks, but nowadays with the advent of DVD and other multi-channel technologies it is common for mixes to be mixed to as many as seven tracks (or even more in some circumstances). This is traditionally the last, and arguably most critical phase of music production that is done in a recording studio. After this step the master tape (data) is sent to a mastering facility that specializes in the process of preparing them for public consumption. Sometimes even the mixdown is done at a separate facility with a special mixing engineer.
Motorized Fader
Simply a fader attached to some type of motor for the purposes of automation control. Motorized faders are commonly found in mixing boards with automation. These systems work by recording and saving as data the positions and movements of the faders as an engineer mixes. During playback the data is converted to a form that can be used to control movements of a precision motor (servo motor), which will then return the fader to the corresponding positions, providing an automated mix. In some motorized fader systems the audio levels are controlled by VCA's (audio doesn't pass through the fader at all); the fader is only used as a means to control the VCA levels and as a means to display the virtual fader positions as levels are controlled by the automation system. Multitrack
Means literally multiple tracks, or more than one track. The term is used to refer to recording devices used to record multiple tracks. It is also used to denote the process of making a recording of multiple tracks as well as the resultant recording (multitrack recording).
Non-Linear Editing
Any editing done on a system that has the ability to randomly access data can probably be characterized as non-linear editing. The term has historically been used to differentiate between editing with tape (whether splicing an audio tape or an A/B roll video editor) and the more modern conventions based on some type of computer system. But use of a computer does not in and of itself necessarily define editing as non-linear, nor does use of a tape machine have to mean that a system is not non-linear. For example, there have been systems that allow the user to enter time code values for edits, which are then carried out automatically by controlling tape machines (usually two playback machines and one record machine). Whether or not such a system is "non-linear" could be debated. In most cases, however, the line is pretty clear. A system where the user can define a region and move it forward or backward in relation to a sequence of other regions is clearly non-linear. Nyquist Frequency
In audio it is the highest frequency that may be accurately sampled based upon the sampling rate. It is based on the Nyquist Theory, which applies to many different fields where data is captured. In general terms the Nyquist Theory is the minimum number of resolution elements required to properly describe or sample a signal. In order to reconstruct (interpolate) a signal from a sequence of samples, sufficient samples must be recorded to capture the peaks and trough of the original waveform. In digital audio the Nyquist Frequency is half of the sampling rate. For example, when a digital recording uses a sampling rate of 44.1kHz, the Nyquist frequency is 22.050kHz. If a signal being sampled contains frequency components that are above the Nyquist limit, aliasing will be introduced in the digital representation of the signal unless those frequencies are filtered out prior to digital encoding. For those who want the gory details there is an excellent paper called Consequences of Nyquist Theorem for Acoustic Signals Stored in Digital Format available at the Digital Recordings website.
Orange Book
CD-WO or CD Write Once. This is the spec detailing physical and optical characteristics for recordable or writable CDs, whether audio or CD-ROM.
Overdub
The process of laying new audio material in, over, or with existing material. Generally this applies to adding parts to a multitrack recording. You could have the basic tracks of a band down and then add vocals or other instruments. These are known as overdubs. The word originates from the word dub. Years ago, when all we had were single track tape machines, the only way to add material to an existing recording was to dub that tape to a second machine and mix in the new material along the way. This process was known as overdubbing.
Play List
Often written as playlist (one word), the term has come into use with hard disk recording systems. It refers to a list, or some sort of view of audio items that can be defined to play in a particular order (usually the order shown in the list). One track of Pro Tools (or almost any digital audio program) can be thought of as a play list. The shown audio regions will play back in order, even though they may have nothing to do with one another. Though the term is used more loosely like that these days, it originates with early disk recording systems that simply showed the sound files in text form in the order they were to be played. One of the earliest of these, Digidesign's Sound Designer, was very popular with rap and R & B artists for making dance edits of their tunes. Though we have progressed quite a bit since then, some relatively high-end systems still use a play list style of editing to this day. And there are people who still swear by Sound Designer for the ease of use in doing this type of editing.
Plug-in
Software that is designed to be integrated within another software environment. Plug-ins are a common method programmers use to provide additional tools for users of a given product. This is advantageous for everyone because it means that the user doesn't have to switch to an entirely different application to perform one specific task that's its specialty. For an early example, PhotoShop - software designed to manipulate digital images in a computer - has a plug-in environment where users can purchase any number of add-on technologies to enhance the capability of the program. These may be things like special lighting effects rendering, painting and motion effects, or anything anyone can think of to add (fans of PhotoShop know there are now hundreds of available plug-ins). Digidesign's Sound Designer II audio recording/editing software was one of the first music oriented programs to adopt the plug-in architecture. Digidesign or other third-party developers wrote software plug-ins for additional functions such as compression, equalization, and eventually many other things that enhanced the capabilities already included in the program. Nowadays many sophisticated applications (for music and otherwise) have the ability to utilize plug-in technology for enhancements.
Preamp
Short for preamplifier. A type of amplifier specifically designed to amplify very weak signals before they are fed to subsequent gain stages or devices. Preamps are commonly used to bring things like the output of microphones up to a level where more equipment can work with the signal. Similarly, magnetic pickups (as used in guitars and basses), and phonograph cartridges are generally run through a preamp to prepare the signal to be used by other equipment downstream. Preamps are called upon to deliver extremely high amounts of gain while introducing very low amounts of noise and distortion. As such they are a critical component in the audio chain, and in recent years have come under much scrutiny by recording engineers causing many dozens of stand-alone mic preamps to be developed that allege to have superior sonic characteristics.
Punch In
A process where the onset of recording on one or more tracks in the midst of an already existing recording can be precisely (a relative term) controlled by the user. For instance, a mistaken word or phrase by an announcer or singer can be corrected by listening to the playback and punching in at the exact moment with the performer correcting the part. In the earlier days of recording punch-ins required considerable skill by the engineer. Many tape machines of the day had subtle differences and inconsistencies in punch in behavior and speed, and don't forget, engineers were often punching in on nearly complete tracks containing otherwise good performances. Stress ran very high during many a punch in those days. Nowadays machines can be set up to automatically punch in and out at specified times, and those times can be rehearsed in advance. And better yet, we have so many tracks and virtual tracks at our disposal these days that punching in is rarely even necessary. We just record a whole new take on a different track and "comp" them together later.
R-Buss
Presumably short for Roland Buss. R-Buss is a connectivity scheme (or bus scheme) developed at Roland for interfacing their digital mixer and recorder products with one another. The idea is for one cable to carry all necessary signals between units so an R-Buss cable can carry eight channels of digital audio with word clock as well as transport and location information. Two 8-track Roland recorders can thus be connected to each other with one R-Buss connection, and they can each be connected to a compatible Roland digital mixer with one R-Buss connection each. Roland also manufactures interfaces for ADAT and other popular MDM and DAW formats so they can be integrated with a Roland mixer using R-Buss.
Record Enable
A function on audio recorders that "arms" a track for recording. It is of particular importance on multitrack recorders because you need some way to arm recording on specific tracks without recording over other tracks that already have material you want to keep, such as when doing overdubs.
Reverb
The remainder of sound that exists in a room after the source of the sound has stopped is called reverberation, sometimes mistakenly called echo (which is an entirely different sounding phenomenon). We've all heard it when doing something like clapping our hands (or bouncing a basketball) in a large enclosed space (like a gym). All rooms have some reverberation, even though we may not always notice it as such. The characteristics of the reverberation are a big part of the subjective quality of the sound of any room in which we are located.Our brains learn to derive a great deal of information about our surroundings from the sound of a room and it's reverberation. Consequently it is necessary to have the proper type and amount of reverberation on recordings in order for them to be aesthetically pleasing or to sound natural to us. This can be accomplished with careful microphone placement, but it is often necessary to employ artificially created reverb.To create reverb, a device known as a reverb unit is employed. Reverb units have historically come in many shapes and sizes, and have used many different techniques to create the reverberation. These days most of the reverb units employed throughout the world are digital, where the sound of the reverb is generated by a computer algorithm and mixed with the original signal. We will be discussing other types of reverb units in the future.
Sample Rate
In a digital recorder or sampler, the sample rate is how many times per second the source material is being "sampled" or recorded. Sample rate affects the frequency response of the final recording or sample; the highest frequency that can accurately be sampled is 1/2 the sample rate. In general, the higher the sample rate, the better the sound quality. But, the best sample rate to use will depend on your application, your gear, and the amount of storage available (the higher the rate, the more storage required). CDs use a 44.1 kHz sample rate, while DAT recorders often default to 48 kHz. Multimedia applications may use rates of 22.05 kHz or even 11.025 kHz for maximum efficiency.
Sample Rate Conversion/Converter
A process or device that converts digital audio from one sample rate to another. Normally when audio plays back at a different sample rate from which it was recorded, the pitch of the audio will change. A sample rate converter must change the rate at which the samples occur without changing the pitch or introducing any other degradation. This is very difficult to do well because it involves adding additional or (usually) removing data from the bit stream. Historically, the resulting audio from such a process was less than desirable, and was a major factor in the decision of many engineers to capture (record) their audio at the sample rate of the delivery medium (usually CD at 44.1 kHz). Nowadays many converters are much better. Sample rate conversion is now part of the every day life of many engineers and is considered a viable part of the production process (assuming a high quality process is used).
SCSI
Pronounced "scuzzy," this acronym stands for Small Computer Systems Interface. SCSI is a hardware interface incorporated into computers, disk-based digital recorders, samplers, and other microprocessor-based equipment. It allows for the easy connection of a variety of peripherals such as hard drives, removable media drives, CD-ROM drives, scanners, and more. One SCSI controller can support up to 7 peripherals, each having their own unique "id" or address. The first and last items in a SCSI chain must be terminated for proper operation. The "theoretical" maximum length of a SCSI chain is 19 feet, but in practice, the chain should be as short as possible!
Sequencer
In music production, a sequencer is a hardware or software device designed to record performance data and play it back in sequence, or in a specific order of events. Early sequencers were analog and programmed by setting a series of voltages (representing pitch) with potentiometers that triggered VCO's. The playback involved having a clock step through or trigger each of these "events" in sequence. Modern sequencers have evolved well beyond that original concept and now have very sophisticated editing and performance features. Nowadays a large percentage of music composition and arrangement is done with sequencers and MIDI instruments. We also now have sequencers with digital audio recording and editing capabilities built in.
Shuttle
A term used mostly in the video tape world to mean the fast forwarding or rewinding (though much slower than an actual fast forward or rewind) of tape while being able to see the picture (usually with no sound). The shuttle control of a video tape machine generally works in concert with a jog or scrub control. The shuttle control, which is usually implemented as a dial or wheel type interface, lets the engineer rapidly locate a section of the tape while viewing the picture. Then the jog or scrub control (also a wheel type interface) lets him slowly find the exact location desired while seeing video and hearing audio. Though this use of the term has its roots in video tape editing it has also been used in the audio world from time to time due to the wheel or dial type user interface. When audio recorders implement a similar wheel or dial for locating audio material they often refer to it as a shuttle control. Similarly non-linear video systems have software emulation of the same functions in order to help video engineers feel more at home on computer based systems.
Signal Path
Simply the route a particular signal takes through a chain of equipment and/or electronic components on the way to its destination. When we think of signal paths in audio we are usually thinking about connecting different pieces of equipment together and routing some signal(s) through them. An example of this would be something like a microphone to mixer to speaker or recorder setup. The signal path has the signal from the microphone pass from the microphone through those (and potentially other) devices on the way to being recorded or amplified (or both). But there is also a signal path inside each piece of equipment. A mixer may be configured to route signals in different ways internally bypassing or utilizing different gain stages along the way to achieve different results. Effects processors often have highly configurable internal signal paths depending upon what they are doing.
SMPTE Time Code
SMPTE (see WFTD archive SMPTE) Time Code (see WFTD archive Time Code) was an adaptation of the time coding that was in use by NASA to track space flight. In its current form (as used by audio and video engineers) it is referenced to video frames and is based on a 24-hour cycle. Within the 24-hour period each frame is identified by the exact time with a unique SMPTE address number in hours, minutes, seconds, and frames (also sub frames can be addressed). The signal is comprised of 80-bit digital words that can be recorded on one track of a video or audio tape recorder using a modulated (see WFTD archive Modulation) audio carrier. SMPTE Time Code recorded on analog tape sounds about the same as the sound your fax or modem makes when transferring data. There are other types of time code, but SMPTE is now such a standard in the industry that the words "SMPTE" and "Time Code" are used almost interchangeably. There are many different variations of SMPTE in use today that we'll eventually cover here.
Soft Limiter
Soft limiting is basically applying a soft knee type of process to a limiter. Soft limiting rounds the peaks of audio to allow a hotter signal to be printed to tape or disk without clipping. It comes in especially handy for digital recording. Digital converters are more accurate at higher levels (up to clipping) so applying a soft limiter to the signal allows the overall signal to be captured several dB hotter while not sounding overly compressed (only the peaks are altered). In many ways a properly configured soft limiter will emulate the natural characteristics of analog tape saturation and can thus produce a very pleasing and familiar sound in digital recording.
Solo
A function commonly found on mixing consoles, soloing a channel is the opposite of pushing a mute switch; solo mutes all channels EXCEPT the one being soloed. In general, solo only affects signals in the control room monitors, or headphones on a live console. It does not mute signal being sent out other outputs. This allows the engineer to listen to individual signals while not interfering with other mixer functions (feeding recorders or PA amplifiers, etc.).
Synchronization
In keeping with the release of the Digital Time Piece, our word for today is "synchronization." In audio terms, synchronizing, or synching, is the process of making two devices operate together as one. One device will be the "master", and tell the second "slave" device when to start, when to stop, and how fast to play. Originally, synching devices primarily meant locking two multitrack tape recorders together to allow for more tracks, or locking audio and video decks together when adding sound to picture. Today, synchronization also encompasses locking recorders to computers, various digital devices' clocks to each other, MIDI to SMPTE, and a variety of other possibilities. Synchronizing wildly different technologies together can be a complex process; having a central master sync device like the DTP around can definitely make life much easier!
Tape Delay
A type of delay or echo processor that uses analog recording tape to achieve the effect. Back in the “old days,” producers and engineers created delay and echo effects using tape machines. Basically a signal would be routed to a separate tape recorder (from the one being used for the performance) that was set to monitor off of the repro head. The slight delay that occurred between when the signal was presented and when it finally came off the repro head provided a delayed signal back to the main recording. The delay time could be adjusted by changing the speed of the tape machine used for the delay. Feedback or multiple echoes could be generated by routing the delayed signal back into the machine, and other, more exotic effects could be created by changing the speed of the machine while signals were passing through it. When applied subtly chorus would result, but other more dramatic effects were easy as well.
This technique became so popular that engineers began to devise ways to make the tape continuously loop on a machine so the tape would never “run out.” Manufacturers eventually stepped in with products to make it easier. The Echoplex and Roland’s Space Echo Series were some of the more popular devices. They had features and controls that were optimized for the intended purpose and used proprietary tapes that were already looped and in small cartridges. These machines were tremendously popular because they were easy to use and portable. Besides being great delay/echo units they were also many musician’s first foray into looping effects.
There were some significant downsides though. It was tape. The recording quality wasn’t always great, depending upon the condition of the heads and how worn the tape was. Even on a perfectly tuned machine the quality of the signal would degrade after several regenerations. In other words, on a repeating echo, the quality would change (degrade) with each repeat because this involved playing the first echo out, recording it again, playing it, recording that signal again, etc. The degradation happened fairly quickly (as opposed to the pure looping effects, which were one recording that just played back over and over).
Eventually analog and later digital devices were created that were in many ways better for doing delay effects. They were cheaper, more reliable, required virtually no maintenance and eventually sounded better in terms of the delayed signals being more like the original and not degrading. But by that time so many musicians and producers were accustomed to the tape delay sound that it was missed. Developers of digital delays had to put filters in to roll off high frequencies as echos repeated and so forth. Even with a modicum of parameters designed to help digital devices sound more like their tape-based ancestors, the difference was still significant. In more recent years the technology of modeling has allowed much more accurate emulation of these old techniques. When you see “tape delay” as one of the algorithms in your digital box or software, it’s probably referring to the emulation of these old sounds, and many of their inconsistencies. Still some purists prefer the sound of the old tape units. This is not unlike the arguments that exist for analog recording today: some of the deficiencies of analog recording actually produce a sound many people find desirable. Third Party
Refers to an entity outside of the buyer and seller arrangement. The buyer is considered the first party (similar to "I" being a first person pronoun), the seller the second party, and anyone supplying things from outside that sphere of influence is considered third party. It is a commonly used term in the high tech world we live in today because many working systems involve third party add-on items. For example, if you add RAM to a computer that comes from some other vendor it is considered third party. Sounds you add to your keyboard that come from some other maker are considered third party sounds. A ProCo cable added to a studio setup consisting of many different kinds of equipment is not normally considered third party. An Apogee A/D converter card installed into your Yamaha recorder is considered third party. A Yamaha converter card would not be third party...unless the card is made by someone other than Yamaha (a third party), which is a fine distinction. As often occurs with jargon, the phrase isn't used with a great deal of precision in terms of exactly what it means.
Time Code
Literally, a code containing or relating to timing. Since the early days of audio it has been necessary to synchronize (see WFTD archive synchronization) audio tracks to motion picture. Later, with the advent of video, MIDI, sequencing, and other technologies the necessity for synchronization of these various formats has become a core concern for production studios and engineers. Almost all time codes are used by recording the "code" on the media of two (or more) machines and then, when the machines are operating, the codes are compared by a synchronizing device which can control slave machines to keep them in sync with the master machine. All sorts of different kinds of timing codes have been used over the years to achieve this, but the current standard for most synchronization needs is SMPTE time code which is comprised of an eight-digit number, based on the 24-hour clock, that identifies a specific frame in a tape. Due to this near standardization, and the ubiquitous nature of SMPTE as a time code format, the words "time code" and the acronym "SMPTE" are used almost interchangeably. There are other time code formats, many of which are based on SMPTE, that we will discuss in the future.
Track
In music and audio production, Track is used in context as both a noun and a verb.
Noun: A defined audio recording, or MIDI data performance, typically linear in nature, that exists in an organized manner. In other words, the result of recording (or tracking) a vocalist might be called a Vocal Track. Track, as a noun, can exist in several additional forms such as "stereo track," "MIDI track," "drum tracks," and so forth.
Verb: Often used in place of the verb, Record. In other words, to "track a vocalist" infers that you're going to record a vocalist. Virtual Track
In the domain of random access digital audio virtual tracks refer to tracks that are present in a particular session, but are not able to be heard because other tracks are currently using up the available playback resources of the hardware involved. All random access digital audio hardware has limitations to the maximum number of simultaneous tracks that can be heard on playback. In order to hear any virtual tracks they must either be selected as a playable track (which will likely involve deselecting another track that was playing) or some sort of submix operation will have to be done that will allow all tracks to be heard.In the domain of MIDI production virtual tracks refer to tracks that exist only in MIDI (as opposed to on a tape machine). A technique that became very popular in the 1980's was to never record MIDI instruments to tape. Instead a MIDI sequencer is synchronized to the tape machines and the MIDI tracks are always "flown in" and mixed in real time as if they are live instruments playing along with the tape. The sequencer stays locked to the tape machine throughout all tracking, overdubbing, and through mixdown. This allows artists to have many more effective or virtual tracks available than what their tape machine would otherwise allow.
Word Length
A term used in digital audio to denote the resolution (dynamic accuracy) of the digital sample that has been taken. See also Bit Depth, which means the same thing. Digital words representing audio data are commonly 8-bits to 24-bits long, depending upon their application. Increasing the word length more practically allows for more detailed dynamic range and usually translates into higher quality audio. Where the sampling frequency determines the number of samples that have been taken in one second, and thus the amount of the upper frequency range able to be captured, the word length determines the dynamic accuracy of each of those samples taken per second. Therefore, 16-bit 44.1k audio literally means that a 16-bit dynamically detailed sample has been taken 44,100 times each second. 24/96 means 24-bits taken 96,000 times per second, and so on.
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