Reverberation, in terms of psychoacoustics, is the
interpretation of the persistence of sound after a sound
is produced. 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 and people, and air. 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. In comparison
to a distinct echo that is a minimum of 50 to 100 ms after
the initial sound, reverberation is reflections that arrive in less than
approximately 50ms. 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 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 60dB 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 20kHz), 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.[4]
"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). This is
given by the equation:.
where
is the speed of sound in the room (for 20 degrees Celsius), is
the volume of the room in m³, total surface area of room in m²,
is the average absorption coefficient of room surfaces, and
the product is the 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 measured in seconds.
Absorption
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
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.
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.
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 1964, 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; 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
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. 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.
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
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. The process of convolution
multiplies each sample of the audio to be processed (reverberated) with the
samples in the impulse response file.
SUBSCRIBERS - ( LINKS) :FOLLOW / REF / 2 /
findleverage.blogspot.com
Krkz77@yahoo.com
+234-81-83195664
For affiliation:
No comments:
Post a Comment