Equalization, or simply EQ, in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. The circuit or equipment used to achieve this is called an equalizer.[1][2]
The concept of equalization was first applied in correcting the frequency response of telephone lines using passive filters; this was prior to the invention of electronic amplification. Initially, equalization was used to compensate for the uneven frequency response of an electric system by applying a filter having the opposite response, thus restoring the fidelity of the transmission. A plot of the system's net frequency response would be a flat line, as its response at any frequency would be equal to its response at any other frequency. Hence the term equalization.
Later the concept was applied in audio engineering to adjust the frequency response in recording, reproduction, and live sound reinforcement systems. Sound engineers correct the frequency response of a sound system so that the frequency balance of the music as heard through speakers better matches the original performance picked up by a microphone. Audio amplifiers have long had filters or controls to modify their frequency response. These are most often in the form of variable bass and treble controls, and switches to apply low-cut or high-cut filters for elimination of low-frequency rumble and high-frequency hiss respectively.
Graphic equalizers and other equipment developed for improving fidelity have since been used by recording engineers to modify frequency responses for aesthetic reasons. Hence in the field of audio electronics the term equalization is now broadly used to describe the application of such filters regardless of intent. This broad definition, therefore, includes all linear filters at the disposal of a listener or engineer.
A British EQ or British style equalizer is one with similar properties to those on mixing consoles made in the UK by companies such as Amek, Neve and Soundcraft[4] from the 1950s through to the 1970s. Later on, as other manufacturers started to market their products, these British companies began touting their equalizers as being a cut above the rest. Today, many non-British companies such as Behringer and Mackie[5] advertise British EQ on their equipment. A British style EQ seeks to replicate the qualities of the expensive British mixing consoles.
Filtering audio frequencies dates back at least to acoustic telegraphy[6] and multiplexing in general. Audio electronic equipment evolved to incorporate filtering elements as consoles in radio stations began to be used for recording as much as broadcast. Early filters included basic bass and treble controls featuring fixed frequency centers, and fixed levels of cut or boost. These filters worked over broad frequency ranges. Variable equalization in audio reproduction was first used by John Volkman working at RCA in the 1920s. That system was used to equalize a motion picture theater sound playback system.[7][8]
The Langevin Model EQ-251A was the first equalizer to use slide controls.[when?] It featured two passive equalization sections, a bass shelving filter, and a pass band filter. Each filter had switchable frequencies and used a 15-position slide switch to adjust cut or boost.[9] The first true graphic equalizer was the type 7080 developed by Art Davis's Cinema Engineering.[when?] It featured 6 bands with a boost or cut range of 8 dB. It used a slide switch to adjust each band in 1 dB steps. Davis's second graphic equalizer was the Altec Lansing Model 9062A EQ. In 1967 Davis developed the first 1/3 octave variable notch filter set, the Altec-Lansing "Acousta-Voice" system.[8]
Daniel N. Flickinger introduced the first parametric equalizer in early 1971. His design leveraged a high-performance op-amp of his own design, the 535 series[11][failed verification] to achieve filtering circuits that were before impossible. Flickinger's patent from early in 1971[12] showed the circuit topology that would come to dominate audio equalization until the present day, as well as the theoretical underpinnings of the elegant circuit. Instead of slide potentiometers working on individual bands of frequency, or rotary switches, Flickinger's circuit allowed arbitrary selection of frequency and cut or boost level in three overlapping bands over the entire audio spectrum. Six knobs on his early EQs would control these sweepable filters. Up to six switches were incorporated to select shelving on the high and low bands, and bypassing for any unused band for the purest signal path.
Similar designs appeared soon thereafter from George Massenburg (in 1972) and Burgess McNeal from ITI corp. In May 1972 Massenburg used the term parametric equalization in a paper presented at the 42nd convention of the Audio Engineering Society.[13] Most channel equalization on mixing consoles made from 1971 to the present day rely upon the designs of Flickinger, Massenburg and McNeal in either semi or fully-parametric topology.[citation needed] In the late 1990s and in the 2000s, parametric equalizers became increasingly available as digital signal processing (DSP) equipment, usually in the form of plug-ins for various digital audio workstations. Standalone outboard gear versions of DSP parametric equalizers were also quickly introduced after the software versions.
Although the range of equalization functions is governed by the theory of linear filters, the adjustment of those functions and the flexibility with which they can be adjusted varies according to the topology of the circuitry and controls presented to the user.
Shelving controls are usually simple first-order filter functions that alter the relative gains between frequencies much higher and much lower than the cutoff frequencies. A low shelf, such as the bass control on most hi-fi equipment, is adjusted to affect the gain of lower frequencies while having no effect well above its cutoff frequency. A high shelf, such as a treble control, adjusts the gain of higher frequencies only. These are broad adjustments designed more to increase the listener's satisfaction than to provide actual equalization in the strict sense of the term.
A parametric equalizer has one or more sections each of which implements a second-order filter function. This involves three adjustments: selection of the center frequency (in Hz), adjustment of the Q which determines the sharpness of the bandwidth, and the level or gain control which determines how much those frequencies are boosted or cut relative to frequencies much above or below the center frequency selected. In a semi-parametric equalizer the bandwidth is preset by the designer. In a quasi-parametric equalizer, the user is given limited switchable options for bandwidth.[14]
A graphic equalizer also implements second-order filter functions in a more user-friendly manner but with somewhat less flexibility. This equipment is based on a bank of filters covering the audio spectrum in up to 31 frequency bands. Each second-order filter has a fixed center frequency and Q factor, but an adjustable level. The user can raise or lower each slider in order to visually approximate a graph of the intended frequency response.
Equalization in the context of audio reproduction is not used strictly to compensate for the deficiency of equipment and transmission channels. A high-pass filter modifies a signal by eliminating only lower frequencies. An example of this is a low-cut or rumble filter, which is used to remove infrasonic energy from a program that may consume undue amplifier power and cause excessive diaphragm excursions in (or even damage to) loudspeakers. A low-pass filter only modifies the audio signal by removing high frequencies. An example of this is a high-cut or hiss filter, which is used to remove annoying white noise at the expense of the crispness of the program material.
A first-order low-pass or high-pass filter has a standard response curve that reduces the unwanted frequencies well above or below the cutoff frequency with a slope of 6 dB per octave. A second-order filter will reduce those frequencies with a slope of 12 dB per octave and moreover may be designed with a higher Q or finite zeros in order to effect an even steeper response around the cutoff frequency. For instance, a second-order low-pass notch filter section only reduces (rather than eliminates) very high frequencies, but has a steep response falling to zero at a specific frequency (the so-called notch frequency). Such a filter might be ideal, for instance, in completely removing the 19 kHz FM stereo subcarrier pilot signal while helping to cut even higher frequency subcarrier components remaining from the stereo demultiplexer.
In addition to adjusting the relative amplitude of frequency bands, an audio equalizer usually alters the relative phases of those frequencies. While the human ear is not as sensitive to the phase of audio frequencies,[15] music professionals may favor certain equalizers because of how they affect the timbre of the musical content by way of audible phase artifacts.[16]
A high-pass filter is a filter, an electronic circuit or device, that passes higher frequencies well but attenuates lower-frequency components. A low-pass filter passes low-frequency components of signals while attenuating higher frequencies. In audio applications these high-pass and low-pass filters are frequently termed low cut and high cut, respectively, to emphasize their effect on the original signal. For instance, sometimes audio equipment will include a switch labeled high cut or described as a hiss filter (hiss being high-frequency noise). In the phonograph era, many stereos would include a switch to introduce a high-pass (low cut) filter, often called a rumble filter, to eliminate infrasonic frequencies. High and low-pass filters are used in audio crossovers to direct energy to the speaker drivers capable of reproducing it. For instance, a low-pass filter is used in the signal chain before a subwoofer to ensure that only deep bass frequencies reach the subwoofer.
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