Dynamic range compression (DRC) or simply compression is a signal processing

operation that reduces the volume of loud sounds or amplifies quiet sounds by narrowing
or compressing an audio signal's dynamic range. Audio compression reduces loud sounds
above a certain threshold while leaving quiet sounds unaffected.

Compression is commonly used in sound recording and reproduction, broadcasting,[1]

live sound reinforcement and in some instrument amplifiers.

A dedicated electronic hardware unit or audio software that applies compression is called
a compressor. In the 2000s, compressors became available as software plugins that run
in digital audio workstation software. In recorded and live music, compression
parameters may be adjusted to change the way they effect sounds. Compression and
limiting are identical in process but different in degree and perceived effect.

A limiter is a compressor with a high ratio and, generally, a fast attack time.

Types
Two main methods of dynamic range reduction

Downward compression

Upward compression
Downward compression reduces loud sounds over a certain threshold while quiet
sounds remain unaffected. Upward compression increases the loudness of sounds below
a certain threshold while leaving louder sounds unaffected. Both downward and upward
compression reduce the dynamic range of an audio signal.[2]

An expander performs the opposite function, increasing the dynamic range of the audio
signal.[3] Expanders are generally used to make quiet sounds even quieter by reducing the
level of an audio signal that falls below a set threshold level. A noise gate is a type of
expander.[2]

Design

A feed-forward compressor design (left) and feedback design (right)

The signal entering a compressor is split, one copy sent to a variable-gain amplifier and
the other to a side-chain where a circuit controlled by the signal level applies the required
gain to an amplifier stage. This design, known as a "feed-forward" type, is used today in
most compressors. Earlier designs were based on a "feedback" layout where the signal
feeding the control circuit was taken after the amplifier.

There are a number of technologies used for variable gain amplification, each having
different advantages and disadvantages. Vacuum tubes are used in a configuration called
'variable-mu': the grid-to-cathode voltage changes to alter the gain.[4] Also used is a
voltage controlled amplifier (VCA), which has its gain reduced as the power of the input
signal increases. Optical compressors use a light sensitive resistor (LDR) and a small
lamp (LED or electroluminescent panel[5]) to create changes in signal gain. This
technique is believed by some to sound smoother because the response times of the light
and the resistor soften the attack and release. Other technologies used include field effect
transistors and a diode bridge.[6]

When working with digital audio, digital signal processing techniques are commonly
used to implement compression via digital audio editors, or dedicated workstations. Often
the algorithms used emulate the above analog technologies.
Controls and features

Different compression ratios

Threshold

A compressor reduces the level of an audio signal if its amplitude exceeds a certain
threshold. It is commonly set in decibels dB, where a lower threshold (e.g. -60 dB) means
a larger portion of the signal is treated (compared to a higher threshold of, e.g., −5 dB).

Ratio

The amount of gain reduction is determined by ratio: a ratio of 4:1 means that if input
level is 4 dB over the threshold, the output signal level is 1 dB over the threshold. The
gain (level) has been reduced by 3 dB:

Threshold = −10 dB
Input = −6 dB (4 dB above the threshold)
Output = −9 dB (1 dB above the threshold)
It is important to understand that when the compressor is lowering the signal to the
desired ratio it is lowering the entire signal, so that the compressor is still working even
after the input signal has fallen below the threshold, for an amount of time determined by
the release.

The highest ratio of ∞:1 is often known as 'limiting'. It is commonly achieved using a
ratio of 60:1, and effectively denotes that any signal above the threshold is brought down
to the threshold level (except briefly after a sudden increase in input loudness, known as
an attack).

Attack and release

The attack and release phases in a compressor
A compressor might provide a degree of control over how quickly it acts. The 'attack
phase' is the period when the compressor is decreasing gain to reach the level that is
determined by the ratio. The 'release phase' is the period when the compressor is
increasing gain to the level determined by the ratio, or, to zero dB, once the level has
fallen below the threshold.

The length of each period is determined by the rate of change and the required change in
gain. For more intuitive operation, a compressor's attack and release controls are labeled
as a unit of time (often milliseconds). This is the amount of time it takes for the gain to
change a set amount of dB, decided by the manufacturer—often 10 dB. For example, if
the compressor's time constants are referenced to 10 dB, and the attack time is set to
1 ms, it takes 1 ms for the gain to decrease by 10 dB, and 2 ms to decrease by 20 dB.[7]

In many compressors the attack and release times are adjustable by the user. Some
compressors, however, have the attack and release times determined by the circuit design
and these cannot be adjusted by the user. Sometimes the attack and release times are
'automatic' or 'program dependent', meaning that the times change depending on the input
signal. Because the loudness pattern of the source material is modified by the compressor
it may change the character of the signal in subtle to quite noticeable ways depending on
the settings used.

Soft and hard knees

Hard Knee and Soft Knee compression

Another control a compressor might offer is hard/soft knee. This controls whether the
bend in the response curve between below threshold and above threshold is abrupt (hard)
or gradual (soft). A soft knee slowly increases the compression ratio as the level increases
and eventually reaches the compression ratio set by the user. A soft knee reduces the
audible change from uncompressed to compressed, especially for higher ratios where a
hard knee changeover would be more noticeable.[8]

Peak vs RMS sensing

A peak sensing compressor responds to the instantaneous level of the input signal. While
providing tighter peak control, peak sensing might yield very quick changes in gain
reduction, more evident compression or sometimes even distortion. Some compressors
apply an averaging function (commonly RMS) on the input signal before comparing its
level to the threshold. This produces a more relaxed compression that more closely
relates to human perception of loudness.

Stereo linking

A compressor in stereo linking mode applies the same amount of gain reduction to both
the left and right channels. This is done to prevent image shifting that can occur if each
channel is compressed individually. It becomes noticeable when a loud element that is
panned to either edge of the stereo field raises the level of the program to the
compressor's threshold, causing its image to shift toward the center of the stereo field.

Stereo linking can be achieved in two ways: Either the compressor sums to mono the left
and right channel at the input, then only the left channel controls are functional; or, the
compressor still calculates the required amount of gain reduction independently for each
channel and then applies the highest amount of gain reduction to both (in such case it
could still make sense to dial different settings on left and right channels as one might
wish to have less compression for left-side events[9]).

Makeup gain

Because the compressor is reducing the gain (or level) of the signal, the ability to add a
fixed amount of make-up gain at the output is usually provided so that an optimum level
can be used.

Look-ahead

The look-ahead function is designed to overcome the problem of being forced to

compromise between slow attack rates that produce smooth-sounding gain changes, and
fast attack rates capable of catching transients. Look-ahead is a misnomer in that the
future is not actually observed. Instead, the input signal is split, and one side is delayed.
The non-delayed signal is used to drive the compression of the delayed signal, which then
appears at the output. This way a smooth-sounding slower attack rate can be used to catch
transients. The cost of this solution is that the signal is delayed.
Reverb

Reverberation, in psychoacoustics and acoustics, is the persistence of sound after a

sound is produced.[1] A reverberation, or reverb, is created when a sound or signal is
reflected causing a large number of reflections to build up and then decay as the sound is
absorbed by the surfaces of objects in the space – which could include furniture, people,
and air.[2] This is most noticeable when the sound source stops but the reflections
continue, decreasing in amplitude, until they reach zero amplitude.

Reverberation is frequency dependent: the length of the decay, or reverberation time,

receives special consideration in the architectural design of spaces which need to have
specific reverberation times to achieve optimum performance for their intended
activity.[3] In comparison to a distinct echo that is a minimum of 50 to 100 ms after the
initial sound, reverberation is the occurrence of reflections that arrive in less than
approximately 50 ms. As time passes, the amplitude of the reflections is reduced until it
is reduced to zero. Reverberation is not limited to indoor spaces as it exists in forests and
other outdoor environments where reflection exists.

Reverberation occurs naturally when a person sings, talks or plays an instrument

acoustically in a hall or performance space with sound-reflective surfaces.[4] The sound
of reverberation is often electronically added to the vocals of singers in live sound
systems and sound recordings by using effects units or digital delay effects.

Contents
• 1 Reverberation time
◦ 1.1 Sabine equation
◦ 1.2 Absorption Coefficient
◦ 1.3 Measurement of reverberation time
• 2 Creating reverberation effects
◦ 2.1 Chamber reverberators
◦ 2.2 Plate reverberators
◦ 2.3 Spring reverberators
◦ 2.4 Electronic reverb
◦ 2.5 Digital reverberators
▪ 2.5.1 Convolution reverb
• 3 See also
• 4 References
• 5 External links

Reverberation time
Sound level in a reverberant cavity excited by a pulse, as a function of time (very
simplified diagram).
The time it takes for a signal to drop by 60 dB is the reverberation time.

RT60 is the time required for reflections of a direct sound to decay 60 dB. Reverberation
time is frequently stated as a single value, if measured as a wide band signal (20 Hz to
20 kHz), however, being frequency dependent, it can be more precisely described in
terms of frequency bands (one octave, 1/3 octave, 1/6 octave, etc.). Being frequency
dependent, the reverb time measured in narrow bands will differ depending on the
frequency band being measured. For precision, it is important to know what ranges of
frequencies are being described by a reverberation time measurement.

In the late 19th century, Wallace Clement Sabine started experiments at Harvard
University to investigate the impact of absorption on the reverberation time. Using a
portable wind chest and organ pipes as a sound source, a stopwatch and his ears, he
measured the time from interruption of the source to inaudibility (a difference of roughly
60 dB). He found that the reverberation time is proportional to room dimensions and
inversely proportional to the amount of absorption present.

The optimum reverberation time for a space in which music is played depends on the
type of music that is to be played in the space. Rooms used for speech typically need a
shorter reverberation time so that speech can be understood more clearly. If the reflected
sound from one syllable is still heard when the next syllable is spoken, it may be difficult
to understand what was said.[5] "Cat", "Cab", and "Cap" may all sound very similar. If on
the other hand the reverberation time is too short, tonal balance and loudness may suffer.
Reverberation effects are often used in studios to add depth to sounds. Reverberation
changes the perceived spectral structure of a sound, but does not alter the pitch.

Basic factors that affect a room's reverberation time include the size and shape of the
enclosure as well as the materials used in the construction of the room. Every object
placed within the enclosure can also affect this reverberation time, including people and
their belongings.

Sabine equation

Sabine's reverberation equation was developed in the late 1890s in an empirical fashion.
He established a relationship between the RT60 of a room, its volume, and its total
absorption (in sabins).

The total absorption in sabins (and hence reverberation time) generally changes
depending on frequency (which is defined by the acoustic properties of the space). The
equation does not take into account room shape or losses from the sound travelling
through the air (important in larger spaces). Most rooms absorb less sound energy in the
lower frequency ranges resulting in longer reverb times at lower frequencies.

Sabine concluded that the reverberation time depends upon the reflectivity of sound from
various surfaces available inside the hall. If the reflection is coherent, the reverberation
time of the hall will be longer; the sound will take more time to die out.

The reverberation time RT60 and the volume V of the room have great influence on the
critical distance dc (conditional equation) where critical distance is measured in meters,
volume is measured in m³, and reverberation time is measred in seconds.

Absorption Coefficient

The absorption coefficient of a material is a number between 0 and 1 which indicates the
proportion of sound which is absorbed by the surface compared to the proportion which
is reflected back into the room. A large, fully open window would offer no reflection as
any sound reaching it would pass straight out and no sound would be reflected. This
would have an absorption coefficient of 1. Conversely, a thick, smooth painted concrete
ceiling would be the acoustic equivalent of a mirror, and would have an absorption
coefficient very close to 0.

Sound absorption coefficients of common materials used in buildings are presented in

this table.

Measurement of reverberation time

Automatically determining T20 value - 5dB trigger - 20dB measurement - 10dB

headroom to noise floor.
Historically reverberation time could only be measured using a level recorder (a plotting
device which graphs the noise level against time on a ribbon of moving paper). A loud
noise is produced, and as the sound dies away the trace on the level recorder will show a
distinct slope. Analysis of this slope reveals the measured reverberation time. Some
modern digital sound level meters can carry out this analysis automatically.[6]

Several methods exist for measuring reverb time. An impulse can be measured by
creating a sufficiently loud noise (which must have a defined cut off point). Impulse
noise sources such as a blank pistol shot or balloon burst may be used to measure the
impulse response of a room.
Alternatively, a random noise signal such as pink noise or white noise may be generated
through a loudspeaker, and then turned off. This is known as the interrupted method, and
the measured result is known as the interrupted response.

A two port measurement system can also be used to measure noise introduced into a
space and compare it to what is subsequently measured in the space. Consider sound
reproduced by a loudspeaker into a room. A recording of the sound in the room can be
made and compared to what was sent to the loudspeaker. The two signals can be
compared mathematically. This two port measurement system utilizes a Fourier
transform to mathematically derive the impulse response of the room. From the impulse
response, the reverberation time can be calculated. Using a two port system allows
reverberation time to be measured with signals other than loud impulses. Music or
recordings of other sound can be used. This allows measurements to be taken in a room
after the audience is present.

Reverberation time is usually stated as a decay time and is measured in seconds. There
may or may not be any statement of the frequency band used in the measurement. Decay
time is the time it takes the signal to diminish 60 dB below the original sound. It is often
difficult to inject enough sound into the room to measure a decay of 60 dB, particularly at
lower frequencies. If the decay is linear, it is sufficient to measure a drop of 20 dB and
multiply the time by 3, or a drop of 30 dB and multiply the time by 2. These are the so-
called T20 and T30 measurement methods.

The concept of Reverberation Time implicitly supposes that the decay rate of the sound is
exponential, so that the sound level diminishes regularly, at a rate of so many dB per
second. It is not often the case in real rooms, depending on the disposition of reflective,
dispersive and absorbing surfaces. Moreover, successive measurement of the sound level
often yields very different results, as differences in phase in the exciting sound build up
in notably different sound waves. In 1965, Manfred R. Schroeder published "A new
method of Measuring Reverberation Time" in the Journal of the Acoustical Society of
America. He proposed to measure, not the power of the sound, but the energy, by
integrating it. This made it possible to show the variation in the rate of decay, and to free
acousticians from the necessity of averaging many measurements.

Creating reverberation effects

A performer or a producer of live or recorded music often induces reverberation in a
work. Several systems have been developed to produce or to simulate reverberation.

Chamber reverberators

The first reverb effects created for recordings used a real physical space as a natural echo
chamber. A loudspeaker would play the sound, and then a microphone would pick it up
again, including the effects of reverb. Although this is still a common technique, it
requires a dedicated soundproofed room, and varying the reverb time is difficult.
Plate reverberators

A plate reverb system uses an electromechanical transducer, similar to the driver in a

loudspeaker, to create vibration in a large plate of sheet metal. A pickup captures the
vibrations as they bounce across the plate, and the result is output as an audio signal. In
the late 1950s, Elektro-Mess-Technik (EMT) introduced the EMT 140;[7] a 600-pound
(270 kg) model popular in recording studios, contributing to many hit records such as
Beatles and Pink Floyd albums recorded at Abbey Road Studios in the 1960s, and others
recorded by Bill Porter in Nashville's RCA Studio B.[citation needed] Early units had one
pickup for mono output, later models featured two pickups for stereo use. The reverb
time can be adjusted by a damping pad, made from framed acoustic tiles. The closer the
damping pad, the shorter the reverb time. However, the pad never touches the plate.
Some units also featured a remote control.

Spring reverberators

"Spring reverb" redirects here. For the album by The Big Wu, see Spring Reverb
(album).

Folded line reverberation device.

The folded coil spring is visible underside of the reverberation device.

A spring reverb system uses a transducer at one end of a spring and a pickup at the other,
similar to those used in plate reverbs, to create and capture vibrations within a metal
spring. Laurens Hammond was granted a patent on a spring-based mechanical
reverberation system in 1939.[8] Guitar amplifiers frequently incorporate spring reverbs
due to their compact construction and low cost. Spring reverberators were once widely
used in semi-professional recording due to their modest cost and small size.

Many musicians have made use of spring reverb units by rocking them back and forth,
creating a thundering, crashing sound caused by the springs colliding with each other.
The Hammond Organ included a built-in spring reverberator, making this a popular effect
when used in a rock band.

Electronic reverb
A reverb "stompbox" pedal for electric guitar.
Electronic effects units capable of creating a reverb effect, including "stompbox" pedals
and rackmount units are used with electric guitar and on the microphoned vocals in live
sound and sound recording. Some instruments, such as clonewheel organs and digital
pianos have onboard electronic or digital reverb effects. Electronic reverb effects are also
built into many 2000s-era guitar amplifiers, keyboard amplifiers and powered mixers
used for live sound in small venues.

Digital reverberators

Digital reverberators use various signal processing algorithms in order to create the
reverb effect. Since reverberation is essentially caused by a very large number of echoes,
simple reverberation algorithms use several feedback delay circuits to create a large,
decaying series of echoes. More advanced digital reverb generators can simulate the time
and frequency domain response of a specific room (using room dimensions, absorption
and other properties). In a music hall, the direct sound always arrives at the listener's ear
first because it follows the shortest path. Shortly after the direct sound, the reverberant
sound arrives. The time between the two is called the "pre-delay."

Reverberation, or informally, "reverb," is a standard audio effect used universally in

digital audio workstations (DAWs) and VST plug-ins.

Convolution reverb

Main article: Convolution reverb

Convolution reverb is a process used for digitally simulating reverberation. It uses the
mathematical convolution operation, a pre-recorded audio sample of the impulse
response of the space being modeled, and the sound to be echoed, to produce the effect.
The impulse-response recording is first stored in a digital signal-processing system. This
is then convolved with the incoming audio signal to be processed.

DELAY

Delay (audio effect)

From Wikipedia, the free encyclopedia
"Block diagram of the signal-flow for a typical simple delay-line," tied to an electric
guitar, for example. Dotted line indicates variable component.[1]
Delay is an audio effect which records an input signal to an audio storage medium, and
then plays it back after a period of time.[2] The delayed signal may either be played back
multiple times, or played back into the recording again, to create the sound of a repeating,
decaying echo. Delay effects range from a subtle echo effect to a pronounced blending of
previous sounds with new sounds. Delay effects can be created using tape loops, an
approach developed in the 1940s and 1950s; analog effects units, which were introduced
in the 1970s; digital effects pedals, introduced in 1984; and audio software plugins,
developed in the 2000s.

Contents
• 1 Early delay systems
• 2 Analog delay
◦ 2.1 Tape based
• 3 Solid state
• 4 Digital delay
• 5 Looping
• 6 Computer software
• 7 Artistic uses
• 8 Applications and variants
• 9 See also
• 10 External links
• 11 Samples
• 12 References

Early delay systems

The first delay effects were achieved using tape loops improvised on reel-to-reel
magnetic recording systems. By shortening or lengthening the loop of tape and adjusting
the read and write heads, the nature of the delayed echo could be controlled. This
technique was most common among early composers of Musique concrète (Pierre
Schaeffer), and composers such as Karlheinz Stockhausen, who had sometimes devised
elaborate systems involving long tapes and multiple recorders and playback systems,
collectively processing the input of a live performer or ensemble.[3] Audio engineers
working in popular music quickly adapted similar techniques, to augment their use of
plate reverb and other studio technologies designed to simulate natural echo.

In the late 1940s and early 1950s, several sound engineers began making devices for use
in recording studios and later more compact machines for live purposes. Les Paul was an
early pioneer; a landmark device was the EchoSonic made by American Ray Butts, a
portable guitar amplifier with a built-in tape echo which became used widely in country
music (Chet Atkins) and especially in rock and roll (Scotty Moore).[4]

Tape echoes became commercially available in the 1950s.[5] An echo machine is the
early name for a sound processing device used with electronic instruments to repeat the
sound and produce a simulated echo. The device was popular with guitarists and was
used by Brian May, Jimmy Page and Syd Barrett among others.

One example is the Echoplex which used a tape loop. The length of delay was adjusted
by changing the distance between the tape record and playback heads. Another example
is the Roland Space Echo with a record and multiple playback tape heads and a variable
tape speed. The time between echo repeats was adjusted by varying the tape speed. The
length or intensity of the echo effect was adjusted by changing the amount of echo signal
was fed back into the pre-echo signal. Different effects could be created by combining
the different playback heads. Some models also included a spring reverb.

Analog delay

Echoplex EP-2
Before the invention of audio delay technology, music employing a delayed echo had to
be recorded in a naturally reverberant space, often an inconvenience for musicians and
engineers. The popularity of an easy-to-implement real-time echo effect led to the
production of systems offering an all-in-one effects unit that could be adjusted to produce
echoes of any interval or amplitude. The presence of multiple "taps" (playback heads)
made it possible to have delays at varying rhythmic intervals; this allowed musicians an
additional means of expression over natural periodic echoes.

Tape based

Many delay processors based on analog tape recording, such as Ray Butts' EchoSonic
(1952), Mike Battle's Echoplex (1959), or the Roland Space Echo (1973), used magnetic
tape as their recording and playback medium. Electric motors guided a tape loop through
a device with a variety of mechanisms allowing modification of the effect's parameters.[6]
In the case of the popular Echoplex EP-2, the play head was fixed, while a combination
record and erase head was mounted on a slide, thus the delay time of the echo was
adjusted by changing the distance between the record and play heads. In the Space Echo,
all of the heads are fixed, but the speed of the tape could be adjusted, changing the delay
time. Thin magnetic tape was not entirely suited for continuous operation, however, so
the tape loop had to be replaced from time to time to maintain the audio fidelity of the
processed sounds.

Several designs were made; the 1959 Ecco-Fonic had a spinning head, the Binson
Echorec used magnetic recording discs. Many were temperamental, such as the Vox
Echomatic. The most successful of the early machines was the 1959 Echoplex by Mike
Battle, "whose sounds are still being experimented with today."[5]

The Binson Echorec, another popular unit, used a rotating magnetic drum or disc (not
entirely unlike those used in modern hard disk drives) as its storage medium. This
provided an advantage over tape, as the durable drums were able to last for many years
with little deterioration in the audio quality.[7]

Often incorporating vacuum tube-based electronics, surviving analog delay units are
sought by modern musicians who wish to employ some of the timbres achievable with
this technology.

Solid state

Early battery guitar delay design, 1976

Solid state delay units using analog bucket brigade delay circuits became available in the
1970s and were briefly a mainstream alternative to tape echo. The earliest known
design,[citation needed] was prototyped at a Boston-based sound reinforcement company in
1976. The core technology used a Reticon SAD1024 IC. This chip and design found its
way into the well known Rockman amplifier some years later. In the 1980s, this design
was used by BOSS for their mass production product.
Though solid state analog delays are less flexible than digital delays and generally have
shorter delay times, several classic models such as the discontinued Boss DM-2 are still
sought after for their "warmer", more natural echo quality and progressively decaying
echos.[citation needed] Additionally, several companies make new analog delays.

Digital delay

Ibanez DE-7 delay pedal

The availability of inexpensive digital signal processing electronics in the late 1970s and
1980s led to the development of the first digital delay effects. Initially, they were only
available in expensive rack-mounted units but eventually as costs came down and the
electronics grew smaller, they became available in the form of foot pedals. The first
digital delay offered in a pedal was the Boss DD-2 in 1984. Rack-mounted delay units
evolved into digital reverb units and on to digital multi-effects units capable of more
sophisticated effects than pure delay, such as reverb and Audio timescale-pitch
modification effects.

Digital delay systems function by sampling the input signal through an analog-to-digital
converter, after which the signal is passed through a series of digital signal processors
that record it into a storage buffer, and then play back the stored audio based on
parameters set by the user. The delayed ("wet") output may be mixed with the
unmodified ("dry") signal after, or before, it is sent to a digital-to-analog converter for
output.

Many modern digital delays present an extensive array of options, including a control
over the time before playback of the delayed signal. Most also allow the user to select the
overall level of the processed signal in relation to the unmodified one, or the level at
which the delayed signal is fed back into the buffer, to be repeated again. Some systems
today allow more exotic controls, such as the ability to add an audio filter, or to play back
the buffer's contents in reverse.

As digital memory became cheaper in the 1980s, units like Lexicon PCM42, Roland
SDE-3000, TC Electronic 2290 offered above 3 seconds delay time, enough to create
background loops, rhythms and phrases. The 2290 was upgradeable to 32 seconds and
Electro-Harmonix offered a 16-second delay and looping machine.

Looping

Gibson Echoplex Digital Pro

See also: Loop (music)
While the mentioned long delay units were a bit clumsy to create loops, the Paradis
LOOP Delay,[8] created in 1992, was the first unit with dedicated looping functions
Record, Overdub, Multiply, Insert, Replace, ... Gibson manufactured a slightly improved
version as Echoplex Digital Pro [9] until 2006. Its software Aurisis LOOP is also the last
loop tool based on a continuous memory structure as used by tape and digital delays.
Most following loopers repeat samples and thus have little in common with a digital
delay, the exceptions being Maneco's early loopers, the Boss DD-20 in digital delay
mode and the Pigtronix Echolution.

Computer software

Steve Harris' Delayorama software

A natural development from digital delay-processing hardware was the appearance of
software-based delay systems. In large part, this coincided with the popularity of both
professional and consumer audio editing software. Software delays, in many cases, offer
much greater flexibility than even the most recent digital hardware delays. Abundant
system memory on modern personal computers offers practically limitless storage for the
audio buffer, and the natural efficiency of audio delay algorithms has made the
implementation trivial for delays offering shifting or random delay times, or the insertion
of other audio effects during the feedback process. Many authors of software plugins
have added functionality to emulate the sounds of the earlier analog units.

Software-based delays are most popular today among musicians in electronic genres or
those who prefer to audition the effect in a digital audio editing and mixing environment.

Artistic uses
In popular and electronic music, electric guitarists use delay to produce densely overlaid
textures of notes with rhythms complementary to the music. Vocalists and
instrumentalists use it to add a dense or ethereal quality to their singing or playing.
Extremely long delays of 10 seconds or more are often used to create loops of a whole
musical phrase.

Robert Fripp used two Revox reel to reel tape recorders to achieve very long delay times
for solo guitar performance. He dubbed this technology "Frippertronics", and used it in a
number of recordings.

John Martyn is widely acclaimed as the pioneer of the echoplex. Perhaps the earliest
indication of his use can be heard on the songs "Would You Believe Me" and "The
Ocean" on the album Stormbringer! released in February 1970. "Glistening
Glyndebourne" on the album Bless The Weather (1971) showcased his developing
technique of playing acoustic guitar through the echoplex. He later went on to experiment
with a fuzz box, a volume/wah wah pedal, and the echoplex on Inside Out (1973) and
One World (1977). Martyn is cited as an inspiration by many musicians including U2's
The Edge.[citation needed]

Applications and variants

Echoplex is a term often applied[citation needed] to the use of multiple echoes which recur
in approximate synchronization with a musical rhythm, so that the notes played combine
and recombine in interesting ways. In fact, it was the name of a particular delay unit, the
Maestro Echoplex.[10]

Doubling echo is produced by adding short-range delay to a recorded sound. Delays of

thirty to fifty milliseconds are the most common; longer delay times become slapback
echo. Mixing the original and delayed sounds creates an effect similar to doubletracking,
or unison performance.

Slapback echo uses a longer delay time (75 to 250 milliseconds), with little or no
feedback. The effect is characteristic of vocals on 1950s rock-n-roll records, particularly
those produced by Sam Phillips and issued by his label Sun Records. In July 1954,
Phillips produced the first of five 78s and 45s that Elvis Presley would release on Sun
Record over the next two years, all of which featured a novel production technique that
Phillips termed "slapback echo".[11] Phillips noted that no live sound was heard without
echo, and hence added the effect to add realism and resultant depth to the played
sound.[citation needed] The effect was produced by refeeding the output signal from the
playback head tape recorder to its record head. The physical space between heads, the
speed of the tape, and the chosen volume being the main controlling factors. Analog and
later digital delay machines also easily produced the effect. Irish guitarist The Edge of
rock band U2 is well known for popularizing the use of slapback delay as a melodic
device in the 1980s. It is also sometimes used on instruments, particularly drums and
percussion.

Flanging, chorus and reverberation (reverb) are all delay-based sound effects. With
flanging and chorus, the delay time is very short and usually modulated. With
reverberation there are multiple delays and feedback so that individual echoes are blurred
together, recreating the sound of an acoustic space.

Straight delay is used in sound reinforcement systems, a straight delay is used to

compensate for the passage of sound through the air. Unlike audio delay effects devices,
straight delay is not mixed back in with the original signal. The delayed signal alone is
sent to loudspeakers so that the speakers reinforce the stage sound at the same time or
slightly later than the acoustic sound from the stage, approximately 1 millisecond of
straight delay per foot of air or 3 milliseconds per meter, depending on the air
temperature's effect on the speed of sound. Because of the Haas effect, this technique
allows audio engineers to use additional speaker systems placed away from the stage and
still give the illusion that all sound originates from the stage. The purpose is to deliver
sufficient sound volume to the back of the venue without resorting to excessive sound
volumes near the front.

Straight delay is also used in audio to video synchronization to align sound with visual
media if the visual source is delayed. Visual media can become delayed by a number of
mechanisms, in which case the associated audio should be delayed to match.