ASSIGEMENT

TO KULDEEP SIR

BY PURUSHOTTAM

Amplifier types
Gain, or multiplication factor that relates the magnitude of the output signal to the input signal. The gain may be specified as the ratio of output voltage to input voltage (voltage gain), output power to input Amplifiers are described according to their input and output properties. They have some kind of power (power gain), or some combination of current, voltage, and power. In many cases, with input and output in the same unit, gain is unitless (though often expressed in decibels). For others this is not necessarily so. For example, a transconductance amplifier has a gain with units of conductance (output current per input voltage). The power gain of an amplifier depends on the source and load impedances used as well as its voltage gain; while an RF amplifier may have its impedances optimized for power transfer, audio and instrumentation amplifiers are normally employed with amplifier input and output impedances optimized for least loading and highest quality. So an amplifier that is said to have a gain of 20 dB might have a voltage gain of ten times and an available power gain of much more than 20 dB (100 times power ratio), yet be delivering a much lower power gain if, for example, the input is a 600 ohm microphone and the output is a 47 kilohm power amplifier's input socket. In most cases an amplifier should be linear; that is, the gain should be constant for any combination of input and output signal. If the gain is not constant, e.g., by clipping the output signal at the limits of its capabilities, the output signal is distorted. There are however cases where variable gain is useful. There are many types of electronic amplifiers, commonly used in radio and television transmitters and receivers, high-fidelity ("hi-fi") stereo equipment, microcomputers and other electronic digital equipment, and guitar and other instrument amplifiers. Critical components include active devices, such as vacuum tubes or transistors. A brief introduction to the many types of electronic amplifier follows.
Power amplifier

The term power amplifier is a relative term with respect to the amount of power delivered to the load and/or sourced by the supply circuit. In general a power amplifier is designated as the last amplifier in a transmission chain (the output stage) and is the amplifier stage that typically requires most attention to power efficiency. Efficiency considerations lead to various classes of power amplifier based on the biasing of the output transistors or tubes: see power amplifier classes.
Power amplifiers by application

Audio power amplifiers RF power amplifier, such as for transmitter final stages (see also: Linear amplifier). Servo motor controllers, where linearity is not important. Piezoelectric audio amplifier includes a DC-to-DC converter to generate the high voltage output required to drive piezoelectric speakers.[2]

Power amplifier circuits

Power amplifier circuits include the following types:

Vacuum tube/valve, hybrid or transistor power amplifiers Push-pull output or single-ended output stages

Vacuum-tube (valve) amplifiers

An ECC83 tube glowing inside a preamp

According to Symons, while semiconductor amplifiers have largely displaced valve amplifiers for low power applications, valve amplifiers are much more cost effective in high power applications such as "radar, countermeasures equipment, or communications equipment" (p. 56). Many microwave amplifiers are specially designed valves, such as the klystron, gyrotron, traveling wave tube, and crossed-field amplifier, and these microwave valves provide much greater single-device power output at microwave frequencies than solid-state devices . Valves/tube amplifiers also have niche uses in other areas, such as

electric guitar amplification in Russian military aircraft, for their EMP tolerance niche audio for their sound qualities (recording, and audiophile equipment)

Transistor amplifiers See also: Transistor, Bipolar junction transistor, Field-effect transistor, JFET, and MOSFET

The essential role of this active element is to magnify an input signal to yield a significantly larger output signal. The amount of magnification (the "forward gain") is determined by the external circuit design as well as the active device.

Many common active devices in transistor amplifiers are bipolar junction transistors (BJTs) and metal oxide semiconductor field-effect transistors (MOSFETs). Applications are numerous, some common examples are audio amplifiers in a home stereo or PA system, RF high power generation for semiconductor equipment, to RF and Microwave applications such as radio transmitters. Transistor-based amplifier can be realized using various configurations: for example with a bipolar junction transistor we can realize common base, common collector or common emitter amplifier; using a MOSFET we can realize common gate, common source or common drain amplifier. Each configuration has different characteristic (gain, impedance...).
Operational amplifiers (op-amps)

An LM741 general purpose op-amp Main articles: Operational amplifier and Instrumentation amplifier

An operational amplifier is an amplifier circuit with very high open loop gain and differential inputs that employs external feedback to control its transfer function, or gain. Though the term today commonly applies to integrated circuits, the original operational amplifier design used valves.
Fully differential amplifier

A fully differential amplifier is a solid state integrated circuit amplifier that uses external feedback to control its transfer function or gain. It is similar to the operational amplifier, but also has differential output pins. These are usually constructed using BJTs or FETs.

Video amplifiers

These deal with video signals and have varying bandwidths depending on whether the video signal is for SDTV, EDTV, HDTV 720p or 1080i/p etc.. The specification of the bandwidth itself depends on what kind of filter is usedand at which point (-1 dB or -3 dB for example) the bandwidth is measured. Certain requirements for step response and overshoot are necessary for an acceptable TV image.
Oscilloscope vertical amplifiers

These deal with video signals that drive an oscilloscope display tube, and can have bandwidths of about 500 MHz. The specifications on step response, rise time, overshoot, and aberrations can make designing these amplifiers difficult. One of the pioneers in high bandwidth vertical amplifiers was the Tektronix company.
Distributed amplifiers

These use transmission lines to temporally split the signal and amplify each portion separately to achieve higher bandwidth than possible from a single amplifier. The outputs of each stage are combined in the output transmission line. This type of amplifier was commonly used on oscilloscopes as the final vertical amplifier. The transmission lines were often housed inside the display tube glass envelope.
Switched mode amplifiers

These nonlinear amplifiers have much higher efficiencies than linear amps, and are used where the power saving justifies the extra complexity.
Negative resistance devices

Negative resistances can be used as amplifiers, such as the tunnel diode amplifier.
Microwave amplifiers Travelling wave tube amplifiers

Traveling wave tube amplifiers (TWTAs) are used for high power amplification at low microwave frequencies. They typically can amplify across a broad spectrum of frequencies; however, they are usually not as tunable as klystrons.
Klystrons

Klystrons are specialized linear-beam vacuum-devices, designed to provide high power, widely tunable amplification of millimeter and sub-millimetre waves. Klystrons are designed for large

scale operations and despite having a narrower bandwidth than TWTAs, they have the advantage of coherently amplifying a reference signal so its output may be precisely controlled in amplitude, frequency and phase.

Musical instrument amplifier

An audio power amplifier is usually used to amplify signals such as music or speech. Several factors are especially important in the selection of musical instrument amplifiers (such as guitar amplifiers) and other audio amplifiers (although the whole of the sound system components such as microphones to loudspeakers affect these parameters):

Frequency response not just the frequency range but the requirement that the signal level varies so little across the audible frequency range that the human ear notices no variation. A typical specification for audio amplifiers may be 20 Hz to 20 kHz +/- 0.5dB. Power output the power level obtainable with little distortion, to obtain a sufficiently loud sound pressure level from the loudspeakers. Low distortion all amplifiers and transducers distort to some extent. They cannot be perfectly linear, but aim to pass signals without affecting the harmonic content of the sound more than the human ear can tolerate. That tolerance of distortion, and indeed the possibility that some "warmth" or second harmonic distortion (Tube sound) improves the "musicality" of the sound, are subjects of great debate

Neutralization
Neutralization generally only affects operation near or at the desired operating frequencies. Neutralization is normally optimized near the upper frequency end of operation, perhaps between 15 and 30 MHz in a 1.8-30 MHz transmitter or amplifier. Neutralization is sometimes needed because tubes have unwanted internal capacitances. The capacitance between the output element and the input element inside the tube will cause the output circuit to couple back to the input. If large enough, this regenerative feedback could cause a loss of efficiency. It might cause the output maximum to occur off the plate current dip, reducing efficiency. It might increase IM distortion or in rare severe cases may cause the amplifier to oscillate someplace the operating frequency. (This problem is common with grounded grid amplifiers using 572B's like the Dentron Clipperton L, or quads of 811A's, like the Collins 30L1. Yaesu has this problem is some FL2100's.) While a need to neutralize does occur in some HF grounded grid amplifiers, it is more common in very high gain grid-driven amplifiers.

Neutralization Adjustment Methods

Neutralization is generally accomplished by adding an external capacitance that is excited exactly 180 degrees out-of-phase with the feedthrough capacitance. One typical adjust procedure is to disable the PA stage by removing screen or filament voltage. A sensitive RF detector is connected to the transmitter output. Normal drive is applied, and the neutralizing capacitor is adjusted until feedthrough power is minimum. The tuning controls are continually peaked for maximum power on the sensitive detector throughout the process. A second less accurate method is to watch the plate current dip in a properly tuned normally operating transmitter. The neutralization capacitor is adjusted until maximum power output and minimum plate current occur simultaneously as the plate capacitor is tuned. The best method varies with the PA design, but in general the most accurate method is by applying drive to a cold PA stage (generally either screen or filament power is removed) and feedthrough power is measured with a sensitive detector.

What Happens If We Don't Neutralize a New Tube?

Many times nothing noticeable occurs if we don't neutralize a PA. The results really depend on how much different the internal capacitance is in the new tube(s) when compared to the capacitance of the tube(s) being replaced. If the PA requires neutralization and we don't neutralize or re-neutralize it, we could find IM distortion higher. We would probably find maximum output power occurs well-off the plate current dip. The un-neutralized stage, in severe cases, might oscillate somewhere near the operating frequency under certain conditions of tuning and loading. Neutralization is generally only accurate over a limited range of frequencies, but fortunately it is almost always at the higher frequency end of the operating range where the PA needs neutralized. The manufacturer probably knows what the optimum adjustment point is. Unrelated Problems are sometimes Blamed on Neutralization Since neutralization is the canceling of feedthrough capacitance, and since that doesn't change over the life of a tube (or even much from a hot tube to a cold tube), neutralization won't "drift out" with certain tube types. Any tube will either neutralize right from the start, or it won't. If it appears to drift out of adjustment something other than the neutralization is at fault. Tubes in HF PA's cannot drift in and out of neutralization because the capacitance is set by the tube's physical construction... not by emission, age, or any other time-variable parameter. People sometimes blame neutralization for problems when they really have gassy or defective emission. Gassy tubes can go into current runaway or even flash over inside. Doing this for 35 years for a living, I've never yet seen a tube in a HF or lower VHF amplifier "drift" or age out of neutralization. The capacitance is for the most part related only to the physical characteristics of the tube, like internal lead length, size of the elements, and spacing of the elements. That's why it is perfectly acceptable to neutralize a cold tube (no filament voltage). The change in feedthrough is very small when the tube is operating compared to when it is cold.

MIXTURE
An professional audio, a mixing console, or audio mixer, also called a sound board, mixing desk, audio production console, or mixer is an electronic device for combining (also called "mixing"), routing, and changing the level, timbre and/or dynamics of audio signals. A mixer can mix analog or digital signals, depending on the type of mixer. The modified signals (voltages or digital samples) are summed to produce the combined output signals. Mixing consoles are used in many applications, including recording studios, public address systems, sound reinforcement systems, broadcasting, television, and film post-production. An example of a simple application would be to enable the signals that originated from two separate microphones (each being used by vocalists singing a duet, perhaps) to be heard through one set of speakers simultaneously. When used for live performances, the signal produced by the mixer will usually be sent directly to an amplifier, unless that particular mixer is "powered" or it is being connected to powered speakers. Among the highest quality bootleg recordings of live performances are the so-called soundboard recordings that are sourced from this mixer output to the speakers. A typical analog mixing board has three sections: Channel inputs Master controls Audio level metering The channel input strips are usually a bank of identical monaural or stereo input channels. The master control section has sub-group faders, master faders, master auxiliary mixing bus level controls and auxiliary return level controls. In addition it may have solo monitoring controls, a stage talk-back microphone control, muting controls and an output matrix mixer. On smaller mixers the inputs are on the left of the mixing board and the master controls are on the right. In larger mixers, the master controls are in the center with inputs on both sides. The audio level meters may be above the input and master sections or they may be integrated into the input and master sections themselves Channel input strip The input strip is usually separated into these section Input jacks Microphone preamplifiers

equalization Dynamics processing (e.g. dynamic range compression, gating)

Routing including direct outs, aux-sends, panning control and subgroup assignments Input Faders

On many consoles, these sections are color-coded for quick identification by the operator. Each signal that is input into the mixer has its own channel. Depending on the specific mixer, each channel is stereo or monaural. On most mixers, each channel has an XLR input, and many have RCA or quarter-inch TRS phone connector line inputs. Basic input controls Below each input, there are usually several rotary controls (knobs, pots). The first is typically a trim or gain control. The inputs buffer the signal from the external device and this controls the amount of amplification or attenuation needed to bring the signal to a nominal level for processing. This stage is where most noise of interference is picked up, due to the high gains involved (around +50 dB, for a microphone). Balanced inputs and connectors, such as XLR or phone connectors, reduce interference problems. There may be insert points after the buffer/gain stage, which send to and return from external processors which should only affect the signal of that particular channel. Effects that operate on multiple channels are connected to the auxiliary sends (below). Auxiliary send routing The auxiliary send routes a split of the incoming signal to an auxiliary bus which can then be used with external devices. Auxiliary sends can either be pre-fader or post-fader, in that the level of a pre-fade send is set by the auxiliary send control, whereas post-fade sends depend on the position of the channel fader as well. Auxiliary sends can be used to send the signal to an external processor such as a reverb, which can then be routed back through another channel or designated auxiliary returns on the mixer. These will normally be post-fader. Pre-fade auxiliary sends can be used to provide a monitor mix to musicians onstage; this mix is thus independent of the main mix. Channel equalization Further channel controls affect the equalization of the signal by separately attenuating or boosting a range of frequencies, e.g., bass, midrange, and treble. Many mixing consoles have a parametric equalizer on each channel. Some mixers have a general equalization control (either graphic or parametric) at the output.

Cue System The cue system allows the operator to listen to one or more selected signals without affecting the console's audio outputs. The signal from the cue system is fed to the console's headphone amp and may also be available as a line-level output that is intended to drive a monitor speaker system. The terms PFL (Pre Fade Listen) and AFL (After Fade Listen) are used to characterize the point in the signal flow from which the cue signal is derived. Input channels are usually configured as PFL so the operator can audition the channel without sending it to any mix. Consoles with a cue feature will have a dedicated button that is typically labeled "Cue" (although AFL, PFL, Solo and Listen are also used) on each channel. Solo In Place (SIP) is a related feature found on more advanced consoles. It typically is controlled by the Cue button but unlike Cue, SIP is destructive of the output mix. Its purpose is to mute everything except the channel or channels that are being soloed. SIP is useful for setup and trouble-shooting in that it allows the operator to quickly mute everything but the signal being worked on. For obvious reasons, SIP is a function that would be dangerous to a mix engineer's career if engaged during performance. For this reason most consoles require very deliberate actions by the operator to engage SIP mode. Master output controls Subgroup and main output fader controls are often found together on the right hand side of the mixer or, on larger consoles, in a center section flanked by banks of input channels. Matrix routing is often contained in this master section, as are headphone and local loudspeaker monitoring controls. Talkback controls allow conversation with the artist through their monitors, headphones or in-ear monitor. A test tone generator might be located in the master output section. Finally, there are usually one or more VU or peak meters to indicate the levels for each channel, for the master outputs and to indicate whether the console levels are clipping the signal. Most mixers have at least one additional output, besides the main mix. These are either individual bus outputs, or auxiliary outputs, used, for instance, to output a different mix to on-stage monitors. As audio is heard in a logarithmic fashion (both amplitude and frequency), mixing console controls and displays are almost always in decibels, a logarithmic measurement system. Since it is a relative measurement, and not a unit itself, the meters must be referenced to a nominal level. The "professional" nominal level is considered to be +4 dBu.[citation needed] The "consumer grade" level is 10 dBV. Digital versus analog

Digidesign's Venue Profile mixer on location at a corporate event. This digital mixer allows plugins from third-party vendors See also: Comparison of analog and digital recording Digital mixing console sales have increased dramatically since their introduction in the 1990s. Yamaha sold more than 1000 PM5D mixers by July, 2005, and other manufacturers are seeing increasing sales of their digital products. Digital mixers are more versatile than analog ones and offer many new features, such as reconfigure signal routing at the touch of a button. In addition, digital consoles often include processing capabilities such as compression, gating, reverb, automatic feedback suppression and delay. Some products are expandable via thirdparty software features (called plugins) that add further reverb, compression, delay and toneshaping tools. Several digital mixers include spectrograph and real time analyzer functions. A few incorporate loudspeaker management tools such as crossover filtering and limiting. Digital signal processing can perform automatic mixing for some simple applications, such as courtrooms, conferences and panel discussions. Consoles with motorized faders can read and write console automation. Propagation delay Digital mixers have an unavoidable amount of latency or propagation delay, ranging from 1.5 ms to as much as 10 ms, depending on the model of digital mixer and what functions are engaged. This small amount of delay is not a problem for loudspeakers aimed at the audience or even monitor wedges aimed at the artist, but can be disorienting and unpleasant for IEMs (In-ear monitors) where the artist hears their voice acoustically in their head and electronically amplified in their ears but delayed by a couple of milliseconds. Every analog to digital conversion and digital to analog conversion within a digital mixer entails propagation delay. Audio inserts to favorite external analog processors make for almost double the usual delay. Further delay can be traced to format conversions such as from ADAT to AES3 and from normal digital signal processing steps. Within a digital mixer there can be differing amounts of latency, depending on the routing and on how much DSP is in use. Assigning a signal to two parallel paths with significantly different processing on each path can result in extreme comb filtering when recombined. Some digital mixers incorporate internal methods of latency correction so that such problems are avoided. Ease of use

16-channel mixing console with compact short-throw faders Analog consoles remain popular due to their continuing to have one knob, fader or button per function, a reassuring feature for the user. This takes up more physical space but allows more rapid response to changing performance conditions. Most digital mixers take advantage of the technology to reduce the physical space requirements of their product, entailing compromises in user interface such as a single shared channel adjustment area that is selectable for only one channel at a time. Additionally, most digital mixers have virtual pages or layers which change the fader banks into separate controls for additional inputs or for adjusting equalization or aux send levels. This layering can be confusing for operators. Analog consoles make for simpler understanding of hardware routing. Many digital mixers allow internal reassignment of inputs so that convenient groupings of inputs appear near each other at the fader bank, a feature that can be disorienting for persons having to make a hardware patch change. On the other hand, many digital mixers allow for extremely easy building of a mix from saved data. USB flash drives and other storage methods are employed to bring past performance data to a new venue in highly portable manner. At the new venue, the traveling mix technician simply plugs the collected data into the venue's digital mixer and quickly makes small adjustments to the local input and output patch layout, allowing for full show readiness in very short order. Some digital mixers allow offline editing of the mix, a feature that lets the traveling technician use a laptop to make anticipated changes to the show while en route, further shortening the time it takes for the sound system to be ready for the artist. Sound quality Both digital and analog mixers rely on analog microphone preamplifiers, a high-gain circuit that increases the low signal level from a microphone to a level that is better matched to the console's internal operating level. In this respect, both formats are on par with each other. In a digital mixer, the microphone preamplifier is followed by an ADC which quantizes the audio stream. Ideally, this process is carefully engineered to deal gracefully with overloading and

clipping while delivering an accurate digital stream over the linear dynamic range. Further processing and mixing of digital streams within a mixer need to avoid clipping and truncation if maximum audio quality is desired. Analog mixers, too, must deal gracefully with overloading and clipping at the microphone preamplifier and as well as avoiding overloading of mix buses. Background hiss in an analog mixer is always present, though good gain stage management minimizes its audibility. Idle subgroups left "up" in a mix will add their background hiss to the main outputs; many digital mixers avoid this problem by low-level gating. Digital circuitry is more resistant to outside interference from radio transmitters such as walkie-talkies and cell phones. Many electronic design elements combine to affect perceived sound quality, making the global "analog mixer vs. digital mixer" question difficult to answer. Controlled ABX double-blind listening tests have not been published at this date; no conclusive answer can be reached. Experienced live sound professionals agree that microphones and loudspeakers (with their innate higher distortion levels) are a much greater source of coloration of sound than the choice of mixer. The mix style of the person mixing is also more important than the make and model of audio console. Analog and digital mixers both have been associated with extremely high-quality concert performances and studio recordings. Remote control Analog mixing in live sound has had the option since the 1990s of using wired remote controls for certain digital processes such as monitor wedge equalization and parameter changes in outboard reverb devices. That concept has expanded until wired and wireless remote controls are being seen in relation to entire digital mixing platforms. It's possible to set up a sound system and mix via wireless (or wired) laptop, touchscreen or tablet, especially if the performance requires no unpredictable fast responses to multiple changing conditions on stage. Computer networks can connect digital system elements for expanded monitoring and control, allowing the system technician to make adjustments to distant devices during the performance. The use of remote control technology can be utilized to reduce "seat-kills", allowing more paying customers into the performance space. Software mixers For recorded sound, the mixing process can be performed on screen, using computer software and associated input, output and recording hardware. The traditional large control surface of the mixing console is not utilized, saving space at the engineer's mix position. In a software studio, there is either no physical mixer fader bank at all or there is a compact group of motorized faders designed to fit into a small space and connected to the computer via USB or Firewire. Many project studios use such a space-efficient solution, as the mixing room at other times can serve as business office, media archival, etc. Software mixing is heavily integrated as

part of a digital audio workstation.Applications Public address systems use a mixing console to set microphones to an appropriate level, and can add in recorded sounds into the mix. A major requirement is to minimise audio feedback. Most bands use a mixing console to combine musical instruments and vocals. Radio broadcasts use a mixing desk to select audio from different sources, such as CD players, telephones, remote feeds, or prerecorded advertisements. These consoles, often referred to as "air-boards" are apt to have many fewer controls than mixers designed for live or studio production mixing, dropping pan/balance, EQ, and multi-bus monitoring/aux feed knobs in favor of cue and output bus selectors, since, in a radio studio, nearly all sources are either prerecorded or preadjusted. Dub producers/engineers such as Lee "Scratch" Perry were perhaps the first musicians to use a mixing board as a musical instrument. Noise music musicians may create feedback loops within mixers, creating an instrument known as a no-input mixer. The tones generated from a no-input mixer are created by connecting an output of the mixer into an input channel and manipulating the pitch with the mixer's dials

Automatic gain control

Schematic of an AGC used in the analog telephone network; the feedback from output level to gain is effected via a Vactrol resistive opto-isolator.

Automatic gain control (AGC) is an adaptive system found in many electronic devices. The average output signal level is fed back to adjust the gain to an appropriate level for a range of input signal levels. For example, without AGC the sound emitted from an AM radio receiver would vary to an extreme extent from a weak to a strong signal; the AGC effectively reduces the volume if the signal is strong and raises it when it is weaker.

Example use cases

AM radio receivers

In 1925, Harold Alden Wheeler invented automatic volume control (AVC) and obtained a patent. Karl Kpfmller published an analysis of AGC systems in 1928.[1] By the early 1930s essentially all broadcast receivers included automatic volume control.[2] AGC is a departure from linearity in AM radio receivers.[3] Without AGC, an AM radio would have a linear relationship between the signal amplitude and the sound waveform the sound amplitude, which correlates with loudness, is proportional to the radio signal amplitude, because the information content of the signal is carried by the changes of amplitude of the carrier wave. If the circuit were not fairly linear, the modulated signal could not be recovered with reasonable fidelity. However, the strength of the signal received will vary widely, depending on the power and distance of the transmitter, and signal path attenuation. The AGC circuit keeps the receiver's output level from fluctuating too much by detecting the overall strength of the signal and automatically adjusting the gain of the receiver to maintain an approximately constant average output level. For a very weak signal, the AGC has no effect, allowing the receiver to operate at its maximum gain; as the signal increases, the AGC reduces

the gain. It is usually disadvantageous to reduce the gain of the RF front end of the receiver on weaker signals as low gain can worsen signal-to-noise ratio and blocking;[4] therefore, many designs reduce gain only for stronger signals. Since the AM detector diode produces a DC voltage proportional to signal strength, this voltage can be fed back to earlier stages of the receiver to reduce gain. A filter network is required so that the audio components of the signal don't appreciably influence gain; this prevents "modulation rise" which increases the effective modulation depth of the signal, distorting the sound. Communications receivers may have more complex AVC systems, including extra amplification stages, separate AGC detector diodes, different time constants for broadcast and shortwave bands, and application of different levels of AGC voltage to different stages of the receiver to prevent distortion and cross-modulation.[5] Design of the AVC system has a great effect on the usability of the receiver, tuning characteristics, audio fidelity, and behavior on overload and strong signals. FM receivers, even though they incorporate limiter stages and detectors that are relatively insensitive to amplitude variations, still benefit from AGC to prevent overload on strong signals.
Radar

A related application of AGC is in radar systems, as a method of overcoming unwanted clutter echoes. This method relies on the fact that clutter returns far outnumber echoes from targets of interest. The receiver's gain is automatically adjusted to maintain a constant level of overall visible clutter. While this does not help detect targets masked by stronger surrounding clutter, it does help to distinguish strong target sources. In the past, radar AGC was electronically controlled and affected the gain of the entire radar receiver. As radars evolved, AGC became computer-software controlled, and affected the gain with greater granularity, in specific detection cells.
Audio/video

An audio tape generates a certain amount of noise. If the level of the signal on the tape is low, the noise is more prominent, i.e., the signal-to-noise ratio is lower than it could be. To produce the least noisy recording, the recording level should be set as high as possible without being so high as to clip or seriously distort the signal. In professional high-fidelity recording the level is set manually using a peak-reading meter. If high fidelity is not a requirement, a suitable recording level can be set by an AGC circuit which reduces the gain as the average signal level increases. This allows a usable recording to be made even for speech some distance from the microphone of an audio recorder. Similar considerations apply with VCRs. A potential disadvantage of AGC is that when recording something like music with quiet and loud passages such as classical music, the AGC will tend to make the quiet passages louder and

the loud passages quieter, compressing the dynamic range; the result can be a reduced musical quality if the signal is not re-expanded when playing, as in a companding system. Most reel-to-reel tape recorders and cassette decks have AGC circuits. Those used for highfidelity allow it to be overridden manually. Most VCR circuits use the amplitude of the vertical blanking pulse to operate the AGC. Video copy control schemes such as Macrovision exploit this, inserting spikes in the pulse which will be ignored by most television sets, but cause a VCR's AGC to overcorrect and corrupt the recording.
Vogad

A voice-operated gain-adjusting device. or volume-operated gain-adjusting device. (Vogad) is a type of AGC or compressor for microphone amplification. It is usually used in radio transmitters to prevent overmodulation and to reduce the dynamic range of the signal which allows increasing average transmitted power. In telephony, this device takes a wide variety of input amplitudes and produces generally consistent output amplitude. In its simplest form, a limiter can consist of a pair of back-to-back clamp diodes, which simply shunt excess signal amplitude to ground when the diode conduction threshold is exceeded. This approach will simply clip off the top of large signals, leading to high levels of distortion. While clipping limiters are often used as a form of last-ditch protection against overmodulation, a properly designed vogad circuit actively controls the amount of gain to optimise the modulation depth in real time. As well as preventing overmodulation, it boosts the level of quiet signals so that undermodulation is also avoided. Undermodulation can lead to poor signal penetration in noisy conditions, consequently vogad is particularly important for voice applications such as radiotelephones. A good vogad circuit must have a very fast attack time, so that an initial loud voice signal does not cause a sudden burst of excessive modulation. In practice the attack time will be a few milliseconds, so a clipping limiter is still sometimes needed to catch the signal on these short peaks. A much longer decay time is usually employed, so that the gain does not get boosted too quickly during the normal pauses in natural speech. Too short a decay time leads to the phenomenon of "breathing" where the background noise level gets boosted at each gap in the speech. Vogad circuits are normally adjusted so that at low levels of input the signal is not fully boosted, but instead follow a linear boost curve. This works well with noise cancelling microphones.
Telephone recording

Devices to record both sides of a telephone conversation must record both the relatively large signal from the local user and the much smaller signal from the remote user at comparable loudnesses. Some telephone recording devices incorporate automatic gain control to produce

acceptable-quality recordings.
Biological

As is the case with many concepts found in engineering, automatic gain control is also found in biological systems, especially sensory systems. For example, in the vertebrate visual system, calcium dynamics in the retinal photoreceptors adjust gain to suit light levels. Further on in the visual system, cells in V1 are thought to mutually inhibit, causing normalization of responses to contrast, a form of automatic gain control. Similarly, in the auditory system, the olivocochlear efferent neurons are part of a bio-mechanical gain control loop.

THE END