Objects produce sound when vibrating in an elastic medium. Solids,
liquids and gas all conduct sound. When something vibrates in the
atmosphere, it pushes the air around it creating an acoustic compression
wave. The trail of this wave creates a drop in pressure, called
rarefaction.
Sound waves, which travel at about 1,086 feet per second (331.1 meters
per second) in the air, have three basic properties, frequency,
wavelength, and volume (amplitude).
Frequency (Hz)
Frequency
Frequency is the number of distinct positive or negative sound wave
elements that repeat in one second. Frequency is measured in Hertz
(Hz). A 20 Hz frequency contains 20 positive and negative cycles of
individual components each second (20 distinct waves passing by in
one second). A 20 kHz (kilohertz) frequency contains 20,000 of these
cycles every second.
Wavelength
Wavelength is the distance between two points on consecutive waves.
It is measured from the same position on a wave in two consecutive
cycles. Wavelength can be measured by taking the horizontal distance
from a point (at the peak in our example) of one wave cycle to the same
point at the peak in the second wave cycle.
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Low frequency sounds have long wavelengths and high frequency
sounds have short wavelengths. The length of a 20 Hz sound wave is
about 56 feet. Speakers that produce low frequencies must therefore be
large in size with long excursions (the distance a speaker moves in and
out) to produce large and long waves. Speakers producing high
frequency sounds must be small enough to move rapidly and produce
the very small waves of high frequencies (about two thirds of an inch at
20 kHz).
Wavelength and Amplitude
Volume (Amplitude)
Volume is the relative loudness or power of an audio signal resulting
from the amplitude of a sound wave. Amplitude is the vertical distance
from zero to the highest point or peak. Sound waves with higher
amplitudes carry more acoustic power and therefore higher volume.
Volume is measured in units called decibels (dB). A dB is one-tenth of
a Bel, named in part after Alexander Graham Bell (the “B” is
capitalized for Bell) and is used in both audio and video applications.
Decibel is a logarithmic scale measuring the intensity (pressure level)
of sound. Decibels are ratios, not fixed quantities. Decibels are also
referred to as a measurement of "gain" with respect to amplifiers (refer
to the glossary).
For the non-linear human ear to perceive a sound that seems twice as
loud, a ten-decibel (10 dB) increase doubles the sound pressure level,
20dB is twice the sound level of 10dB, and 30dB is twice as loud as
20dB. 40dB is twice the sound level of 30dB and four times the sound
level of 20 decibels.
With some kinds of equipment, such as microphones, analog tape
recorders, or LP playback systems, the dB measurement is "weighted"
as to audibility, because the ear is more sensitive to particular
frequencies. Two common corrections for hearing characteristics are
the A-weighted and the more rigorous C-weighted scales, indicated as
dBA or dBC, respectively.
The term decibel is also used in various other measurements such as
signal-to-noise ratio, gain and dynamic headroom. In these instances,
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decibel refers to the measurement of signal increase or signal strength
instead of sound pressure level, but the logarithmic scale concept
remains the same.
Decibel Scale
VU and dB
VU and dB meters both measure the audio power and they both use
logarithmic scales to report that power. In both measures, the zero is
chosen as the highest power for which distortion is acceptable.
Where VU and dB differ is in how they measure audio power. VU is
short for "volume units" and it is a measure of average audio power. A
VU meter responds relatively slowly and considers the sound volume
over a period of time. Its zero is set to a 1% total harmonic distortion
level in the recorded signal.
Decibel (dB) meters measure instantaneous audio power. A dB meter
responds very rapidly and considers the audio power at each instant. Its
zero is set to a 3% total harmonic distortion level. Because of these
differences in zero definitions, zero on the dB meter is approximately
+8 on the VU meter.
Perception
The human ear can usually hear sounds in the range of 20 Hz to 20
kHz, but are most sensitive to sounds from 2 KHz to 4 KHz, the same
range as the human voice. With age, this range decreases, especially at
the upper limit. Very Low frequencies (below 20Hz) cannot be heard,
but loud low frequency sounds can be felt as vibrations on the skin. The
frequency resolution of the ear is, in the middle range, about 2 Hz.
Changes in pitch larger than 2 Hz are noticeable. Even smaller pitch
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differences can be perceived when two pitches interfere and are heard
as a frequency difference pitch.
The lower limit of audibility is defined as 0 dB, there is no defined
upper limit. The upper limit is more a question of where the ear will be
physically harmed. This limit depends on the time exposed to the
sound. The ear can be exposed to short periods of sounds of 120 dB
without harm, but long exposure to 80 dB sounds can do permanent
damage. The human voice range is about 68-76 dB; a jet plane creates
about 120 dB of sound.
Sound waves radiate out from the source in straight lines regardless of
frequency or wavelength. But low frequency (long wavelength) sounds
do not fit in confined spaces. They loose their directional character and
that is why you only need one subwoofer for a sound system; you can't
tell where the lowest frequency sounds are coming from when the
sound is confined in a room.
The model of sound so far described is a simplified version, operating
in just one dimension (as opposed to the real world three-dimensions),
and assumes that the vibrating particles of air are held semi-rigidly. In
the real world, air molecules are in constant random motion in all three
dimensions. Air pressure constantly changes as the molecules collide
and rebound from each other and from objects in the environment. This
random motion creates a background level of noise and helps define the
lower limit of hearing.
Timbre
Musical timbre is a property of sound. It is composed of spectral
components containing perceptual cues, and can be described by
Fourier series coefficients. The spectral “envelope” of a sound, the
profile of the Fourier series, is the sound amplitude behavior over time.
The pattern that this sound pressure variation creates is the waveform
of the sound.
Timbre is the temporal evolution of the spectral envelope. This
envelope consists of an "attack" portion at the beginning or onset of the
sound, a sustained portion (stationary state), and a decay portion.
Timbre has psychoachoustic properties. A number of transient
fluctuations occur during the initial part (attack), for example, the
moment a violinist puts the bow to the string. These are called onset transients and are important in identifying the sound and its location in
space.
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Waveform Envelope Examples:
Psychoacoustics
Psychoacoustics is the study of human auditory perception. It includes;
the physical characteristics of sound waves, the physiological structure
of the ear, the electrical signal from the ear to the brain, and the
subjective interpretation of the listener. Understanding psychoacoustics
is essential to creating surround sound.
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Localization
Our stereophonic ears can discern azimuth or horizontal (left-right)
directionality, and zenith or vertical (up-down) directionality. We
perceive directionality using the localization mechanisms of: interaural
time difference, interaural intensity difference, pinna filtering, and
motion parallax.
Interaural Time Difference
The horizontal position of a sound is determined by comparing the
information coming from the left and right ears. The Interaural Time
Difference is the difference in arrival time at each ear. The approximate
six-inch separation of the ears slightly delays the sound, each ear
receiving a slight difference when the sound is not equally distant from
the two ears. Although the time delay differences are very slight, the
brain extracts precise directional information from this information.
Human listeners are able to accurately locate the sources of sound from
almost any direction, even from above when interaural differences are
almost zero. Listeners are also capable of locating sound sources in a
room when the reflections from the walls are louder than the sound
coming directly from the source.
Interaural Time Difference
Interaural Intensity Difference
The head, shoulders and upper torso create a sound barrier at one ear or
the other. This acoustical shadow called the Interaural Intensity
Difference.
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Interaural Intensity Difference
For example, a sound coming from the extreme left has a lowered
intensity in the right ear in addition to a slight time delay. The reduced
intensity is the additional distance plus the effect of the acoustical
shadow. The amount of this effect depends on frequency, and is useful
for high frequencies up to wavelengths twice the distance between the
ears (about 1 kHz). Lower frequencies, with longer wavelengths, bend
around obstructions.
Pinna Filtering
The pinna structure is the outer part of the ear. Its forward pointing
position and complex curves affect the way sound is heard. A sound is
coming from behind or above bounces off the pinna in a different way
than from in front or below. When the indirect (reflected) sounds from
the pinna combine with the direct sounds, the wavelengths of the sound
are altered.
Pinna Filtering
The brain, interpreting the altered sounds, produces directional
information. To provide additional cues, small head movements
(motion parallax) allow the brain to judge relative differences.
The Precedence Effect
The precedence effect is a listening strategy unconsciously used to cope
with distorted localization cues in a confined space. Localization
judgments are based on the first arriving sound waves at the beginning
of a sound. This strategy is known as the precedence effect, because the
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earliest arriving sound wave is given precedence over the subsequent
reflections and reverberations.
Reverberation and Echoes
The ratio of the direct to reverberant energy is a primary cue for range
and space. Very short delays cause a sound image to shift spatially and
color the tone. Longer delays contribute to a spatial impression of
reverberation. Reverberation can also make speech indistinct by
masking the onset transients.
Sound decreases inversely with the square of the distance. But in an
ordinary room, the sound is reflected and scattered against room
boundaries and objects within the room. Reverberation is essentially an
echo that increases by bouncing off of hard surfaces. Reverberations
are dampened when absorbed by soft materials such as rugs, carpet and
sofas. These reflections are most noticeable when the time delay
between the direct sound and the reverberation gets longer than 30 to
50 ms, the echo threshold.
Reverberations
Acoustic designers place importance on early reflections (arriving
within the first 80 ms), which reinforce the direct sound (as long as the
angle of reflection is not too wide). Reflections arriving after 80 ms add
reverberant energy, which gives the sound spaciousness, warmth and
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envelopment. The acoustic design listening spaces usually involves
creating a balance between clarity, definition, and spaciousness.
Listeners often have different preferences regarding this balance.
Temporal Masking
Temporal masking is a defense mechanism of the ear that is activated
to protect its delicate structures from loud sounds. When exposed to a
loud sound, the human ear reacts by contracting slightly, temporarily
reducing the perceived volume of sounds that follow. Loud sounds in
an audio signal tend to overpower other sounds that occur just before
and just after it.
A History of Surround Sound
The simplest method of sound recording is called monaural or mono.
All the sound is recorded on one audio track and played back on one
speaker.
Two-channel recordings played back on speakers on either side of the
listener are referred to as stereophonic or stereo. The simplest twochannel recordings, (binaural recordings) are produced with two
microphones. Playback of these two channels on two speakers recreates
some of the experience of being present at a concert event. But the
listener must be anchored in the "sweet spot" between the speakers to
maintain the illusion of the phantom sound from between the speakers.
Surround recordings add additional audio channels so that sound comes
from multiple directions. In effect, widening the sweet spot and
enhancing the realistic sound quality.
The term "surround sound" refers to specific multi-channel systems
designed by Dolby Laboratories, but is commonly used as a generic
term for theater and home theater multi-channel sound systems.
Early Surround
Walt Disney's "Fantasia" (1941), was one of the first surround sound
motion pictures. Four separate recordings of each orchestra section
were recorded on a separate reel of film and played through speakers
positioned around the theater.
By the late 1950s, movies were encoded with simpler multi-channel
formats. Several different systems emerged, including Cinerama and
Cinemascope. These systems were referred to as stereophonic sound, or
theater stereo. Stereophonic sound used multiple magnetic audio tracks
at the edges of the film. The standard film format could support two
optical audio tracks or up to six magnetic audio tracks. A four-channel
theater system included: left, right, center speakers behind the screen,
and surround speakers along the sides and back of the theater.
Quadraphonic
In the quadraphonic systems of the early 1970s, two rear surround
channels were combined (matrixed or encoded) with the two front
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channels so that the two sides of an LP groove carried four playback
channels. This four-speaker system required a decoder and a separate
rear channel amplifier. Problems with system standardization prevented
technological development.
Dolby
In the mid 1970's, Dolby Laboratories (www.dolby.com) devised a
method to encode additional audio channels. This technology, initially
known as Dolby Stereo when it was launched in 1975, was later
renamed Dolby Surround.
In 1982, Dolby Surround and enhanced Dolby Pro Logic playback
decoder became available to the consumer. Dolby Surround, like the
earlier Quadraphonic systems, used channel matrixing to combine four
audio channels into two signals.
Also described as 4-2-4 matrixing, these signals (compatible with twospeaker stereo playback) can be decoded into multiple channels.
Basic Dolby Surround decoding yields: front left, front right, and one
surround channel (the center channel is a phantom).
The 4-2-4 encoder accepts four separate inputs (left, right, center and
surround) and creates two outputs (left-total and right-total). The front
left and right channels are a regular stereo signal. The center channel is
inserted equally in the left and right channel, with a 3 dB level
reduction to maintain constant acoustic power.
Dolby 4-2-4 Encoding
The surround input is also divided equally between the left-total and
right-total signals but first undergoes three processing steps:
• It is frequency band-limited from 100 Hz to 7 kHz
• It is encoded with a modified Dolby B-type noise reduction
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