Sound and Hearing Functional Aspects

Dr. Mohieddin Moradi
mohieddinmoradi@gmail.com
1
Dream
Idea
Plan
Implementation

Sound and Hearing
Some Functional Aspects of Hearing
Sound Measurement
Loudness Units
Microphones
Speakers
MIDI Standard
Audio Tools and Connectors
Audio Production and Postproduction
Digital Audio Principals
Domestic and Broadcast Audio Delivery Formats
Outline
2

“Audio” is a Latin word meaning “ I Hear “
Similarly “ Video “ means “ I see” . 4

Human Aural Perception
– The ear is a pressure sensitive organ that divides the
audio spectrum into a set of overlapping frequency
bands whose bandwidth increases with frequency.
– Human can hear from 20Hz to 20kHz.
• Sometimes called bandwidth.
• Frequency perception is not flat.
• High frequency perception degrades with age.
• Most adults cannot hear above 15kHz.
– The human ear can hear a wide dynamic range.
• From very quiet sounds to very loud sounds.
– Brain misses certain sounds.
• Certain frequencies are hidden by other frequencies.
• Loud sounds swamp quiet sounds. 5

Species-Specific Frequency Range
6

7
How the Human Ear Works?
Ear Drum
Hammer
Anvil
Stirrup

– Sound is collected by the outer ear.
– Sound waves are concentrated by the Ear Canal.
Outer ear
Ear Canal
8

– Concentrated sound waves vibrate the Ear Drum.
– This moves the Hammer, Anvil and Stirrup.
– Stirrup vibrates the Oval Window in the Cochlea.
Ear Drum Hammer
Anvil
Stirrup
Hammer
Ear Drum
Anvil
Stirrup
9

– Cochlea is filled with liquid and lined with sensory hairs.
– Length of the sensory hairs is different and picks up different
sounds.
Oval window
Cochlea
Three rows of Outer Sensory Hair Cells
One Row of Inner Sensory Hair Cells10

– Different sounds fire different sensory hairs receptors.
– Signals sent from receptors to the brain via the auditory nerve.
Auditory nerve
Semi-circular Canals
Vestibular nerve
Auditory nerve
Stirrup
Cochlea
11

– Ear is also responsible for detecting movement and position.
– 3 semi-circular canals about 90 degrees from each other filled with fluid.
– Head movement results in fluid movement, picked up by vestibular nerve.
Vestibular nerve
Vestibular nerve
Auditory nerve
Stirrup
Cochlea
12

The “Impedance Problem”
99.9% of sound energy in the air is reflected at the
air: water boundary
(10 log(0.1/100)) = -30 dB loss) (1/1000x)
How does the ear compensate for this loss as sound
energy is transmitted from the air to the fluid that
filled the cochlea?
2 dB gain via ossicular leverage (1.6x)
25 dB gain via surface area condensation
(eardrum  stapes) (316x)
~5 dB gain at mid-frequencies (3x) due to pinna
and auditory canal resonance

14
Nature of the Sound Stimulus

Nature of the Sound Stimulus
“Sound” is the rhythmic compression and
decompression of the air around us caused by a
vibrating object.
15

Properties of Waves
– Wavelength () is measured from crest-to-crest (or trough-to-trough, or upswing to upswing, etc.)
– Speed (c) for traveling waves (sound, light, water)
– Frequency (f) refers to how many cycles pass by per second
• It measured in Hertz, or Hz: cycles per second
• It associated with period: T = 1/f
– These three are closely related:
f = c
 or T
pressure
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Speed of Sound
Acoustic energy results from a traveling wave of
rhythmic “compression” through a physical medium
(e.g., air; water; steel).
It is the “compression” that travels not the medium.
The characteristic speed of this travelling wave
varies as a function of the medium (elasticity;
density).
The speed of acoustic energy through the air (aka
“sound”) is 331 m/sec (or 742 MPH) at 0-deg C
(Faster at higher temperatures).
17

Characteristics of Sound
19
• Loudness
Sensory characteristic of sound
produced by the amplitude (intensity)
of the sound wave.
• Pitch
Sensory characteristic of sound
produced by the frequency of the
sound wave.
• Timbre
Quality of a sound wave that derives
from the wave’s complexity.

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Loudness, Pitch and Quality/Timber

Loudness
− It is the characteristic of the sound by which a loud sound can be
distinguished from a soft sound.
− Loudness is a psychoacoustic property of sound, in that it represents our
human perception of how “loud” or “soft” sounds of various intensities are.
− Loudness depends on the intensity of sound near the ear.
− It is the measure of ear's response to the vibration.
− Units of measurement include the phon and sone
− The loudness of a 1000 Hz pure tone at 40 dB=40 phons
− The 40 phon curve represents the sound intensity that is required at each
frequency to sound equally loud to a 1000 Hz tone at 40 dB.
21

Loudness Units (Phon and Sone)
− The phon is a unit of loudness level for pure tones.
− The loudness of a 1000 Hz pure tone at 40 dB=40 phons
− The 40 phon curve represents the sound intensity that is required at
each frequency to sound equally loud to a 1000 Hz tone at 40 dB.
− The phon scale tells us that one sound is louder than another.
− The sone scale can tell us how much louder one sound is than
another. (i.e. 2x as loud, 4x as loud). (1 sone=40 phons)
− A change of 10 dB results in a doubling or halving of the perceived
loudness. (A 1000 Hz tone at 40 dB is perceived as being half as
loud as a 1000 Hz tone at 50 dB)
22

Pitch
− Our perception of frequency.
− Pitch is a psychoacoustic property of sound, in that it represents our human perception of how “high”
or “low” sounds of various frequencies are.
− If the frequency of a vibrating object is low then the pitch of sound is said to be low and vice versa.
− Pitch of a sound depends upon the frequency of the sound recorded by the human.
− Unit of measure is the mel
− The pitch of a 1000 Hz tone at 40 dB SPL=1000 mels
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Linear Plot
Pitch (mel Scale)
− Our perception of frequency, i.e., pitch, is logarithmic.
− The mel scale is not linear (Doubling the perceived pitch does not equate to doubling the frequency)
− Perceived pitch increases “linearly” with stimulus frequency below 4KHz; but grows at a much slower rate at 4KHz
and above.
− 2000 mels is equivalent to a pitch that sounds twice as loud as a 1000 mel tone
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Reference unit of perceived Pitch:
1000 Hz = 1000 Mels
Semi-Log Plot

25

Quality/Timbre
− The characteristic of sound which enables us to distinguish between the sounds if same loudness and pitch.
Quality of a sound depends on wave form of the sound.
− Pure tones have a pure sine wave shape and a single frequency. However the notes from musical instruments
and voices are more complex and normally contain many frequencies.
− If a trumpet and a clarinet play the same note we can still tell the difference between the two instruments.
Likewise, different voices sound different even when singing the same note. Why?
− We know that if they are playing or singing the same pitch the fundamental frequency is the same for both.
− These differences in the quality of the pitch are called timbre and depend on the actual shape of the wave
which in turn depends on the other frequencies present and their phases.
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− The subjective perception of pitch,
loudness and timbre are each related to
more than one quantity that can be
measured in the laboratory.
− The diagram shows some of the
connections between objective
(laboratory) measurements and
subjective perception.
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− Our perception of loudness is mainly determined
by the intensity of the sound (energy per second
per square meter) but also is influenced by
frequency and waveform of the sound.
− Our perception of pitch is mainly determined by
the fundamental frequency but also influenced
by intensity and waveform.
− Timbre is determined by waveform (which is
determined by the other frequencies present and
their phases) with influences from intensity and the
fundamental frequency.
− The duration of a sound also affects how we
perceive its pitch, loudness and timbre.
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Longitudinal vs. Transverse Waves
– The sound is a longitudinal wave, meaning that the motion of particles is along the
direction of propagation
– The water waves and light have Transverse waves, meaning that the motion of particles is
perpendicular to the direction of propagation
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Why is Sound Longitudinal?
• Waves in air can’t really be transverse, because the atoms/molecules are not bound to
each other.
• Air molecules can really only bump into one another
• Imagine people in a crowded train station with hands in pockets
• Pushing into crowd would send a wave of compression into the crowd in the direction of push
(longitudinal)
• Jerking people back and forth (sideways, over several meters) would not propagate into the
crowd
• But if everyone held hands (bonds), this transverse motion would propagate into crowd

Sound Pressure
– The acoustic or sound pressure vibrations are superimposed on the surrounding static air
pressure which has a value of 100,000 Pascal.
• Sound Pressure is measured in units called Pascal.
• 1 Pascal (Pa) = 1 Newton of force/m2
• 1 Atmosphere = 100,000 Pa
• Human absolute hearing threshold = 0.00002 Pa = 20 µPa
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Sound Pressure
Pressure variations in the surrounding air– The emission of the pressure variations
can be compared to the ripples in a
pond caused by a stone thrown in the
water.
– The ripples spread out from the point
where the stone entered.
– However the water itself does not
move away from the center.
– The water stays where it is, moving up
and down to produce the circular
ripples on the surface.
– The stone is the sound source, the
pond is the air, and the ripples are the
resulting sound wave (Sound is like
this).
34

The “decibel” (dB)
– The decibel is a logarithmic unit used to describe a ratio (i.e., log (x/y))
– In engineering analyses, it is used to normalize “power” measurements to a known reference
and then compresses the resulting ratio using a log10 operation.
– This format is convenient for engineering analyses involving wide dynamic ranges (when
very small and the very large magnitudes must be considered simultaneously).
𝒅𝑩 = 𝟏𝟎 log(
𝑶𝒃𝒔𝒆𝒓𝒗𝒆𝒅 𝑷𝒐𝒘𝒆𝒓
𝑹𝒆𝒇𝒆𝒓𝒆𝒏𝒄𝒆 𝑷𝒐𝒘𝒆𝒓
)
35

dB-SPL or dBSPL
dB-SPL : Observed sound pressure compared to reference sound pressure
– The transducers (microphones) on sound level meters measure sound pressure (i.e., N/m2 or
Pascal).
– Pressure needs to be converted to power prior to calculation of the decibel equivalent….i.e.,
Acoustic Power = (Acoustic Pressure)2
– Finally, we need to agree upon a Reference value.
– By convention, we use 20 µPa (i.e., the hearing threshold)
Thus:
(𝒅𝑩 − 𝑺𝑷𝑳) 𝑳 𝒑 = 𝟏𝟎 log
𝑶𝒃𝒔𝒆𝒓𝒗𝒆𝒅 𝑨𝒄𝒐𝒖𝒔𝒕𝒊𝒄 𝑷𝒐𝒘𝒆𝒓
(𝟐𝟎 µ𝑷𝒂) 𝟐
(𝒅𝑩 − 𝑺𝑷𝑳) 𝑳 𝒑 = 𝟐𝟎 log
𝑶𝒃𝒔𝒆𝒓𝒗𝒆𝒅 𝑨𝒄𝒐𝒖𝒔𝒕𝒊𝒄 𝑷𝒓𝒆𝒔𝒔𝒖𝒓𝒆
𝟐𝟎 µ𝑷𝒂
36

Range of Sound Pressure Levels in dB
37Perception of dBs
𝒅𝑩 − 𝑺𝑷𝑳
𝐿 𝑝 = 20 log
𝑃[𝑃𝑎]
𝑃0
𝐿 𝑝 = 20 log
𝑂𝑏𝑠𝑒𝑟𝑣𝑒𝑑 𝑆𝑜𝑢𝑛𝑑 𝑃𝑟𝑒𝑠𝑠𝑢𝑟𝑒 [𝑃𝑎]
20µ𝑃𝑎

Simple Rules for Conversion
[dB]
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+6 dB-SPL = doubling sound pressure +20 dB-SPL = 10x pressure
+3 dB-SPL = doubling acoustic power +10 dB-SPL = 10x acoustic power

Conversion to dB using Charts
39

130 Loud hand clapping at 1 m distance
110 Siren at 10 m distance
95 Hand (circular) power saw at 1 m
80 Very loud expressway traffic at 25 m
60 Lawn mower at 10 m
50 Refrigerator at 1 m
40 Talking; Talk radio level at 2 m
35 Very quiet room fan at low speed at 1 m
25 Normal breathing at 1 m
0 Absolute threshold
dB-SPL
Range of Sound Pressure Levels
40

Intensity
– = wave amplitude = volume = loudness
– This is a measurement of the height of the sound wave
– Measured in decibels (dB)
– The human ear responds to a great range of sound intensities from 0dB
(threshold of hearing) to 135dB (the threshold of pain)
– VU meter in sound production is a volume unit measure
Characteristics of Sound
41

Sound Intensity
42
In the case of free space propagation in which no interference effects were present we do need to vector.

Area of sphere = 4πr2
43
Inverse-Square Law
Sound Intensity

Sound Intensity Level (𝑳 𝑰)
𝐿𝐼 = 10 log
𝑂𝑏𝑠𝑒𝑟𝑣𝑒𝑑 𝑆𝑜𝑢𝑛𝑑 𝐼𝑛𝑡𝑒𝑛𝑠𝑖𝑡𝑦 [𝑊/𝑚2
]
10−12[𝑊/𝑚2]
dB-SIL : Observed sound intensity compared to reference sound intensity
44
𝐿𝐼 = 10 log
𝐼 [𝑊/𝑚2
]
𝐼0 [𝑊/𝑚2]

− The sound power level, sound intensity level, and sound pressure level be approximately
equivalent in the case of free space propagation in which no interference effects were
present.
− The main use of sound power is for the noise rating of machines etc. and sound intensity is
mainly used for location and rating of noise sources.
45
𝒅𝑩 − 𝑺𝑰𝑳 𝐿𝐼 = 10 log
𝑂𝑏𝑠𝑒𝑟𝑣𝑒𝑑 𝑆𝑜𝑢𝑛𝑑 𝐼𝑛𝑡𝑒𝑛𝑠𝑖𝑡𝑦 [𝑊/𝑚2]
10−12[𝑊/𝑚2]
𝒅𝑩 − 𝑷𝑾𝑳 𝐿 𝑊 = 10 log
𝑂𝑏𝑠𝑒𝑣𝑒𝑑 𝑇𝑜𝑡𝑎𝑙 𝐴𝑐𝑜𝑢𝑠𝑡𝑖𝑐 𝑃𝑜𝑤𝑒𝑟 [𝑊]
10−12[𝑊]
𝒅𝑩 − 𝑺𝑷𝑳 𝐿 𝑝 = 20 log
𝑂𝑏𝑠𝑒𝑟𝑣𝑒𝑑 𝑆𝑜𝑢𝑛𝑑 𝑃𝑟𝑒𝑠𝑠𝑢𝑟𝑒 [𝑃𝑎]
20µ𝑃𝑎
Acoustic Intensity Level (𝑳 𝑰), Acoustic Power Level (𝑳 𝑾 ), and
Acoustic Pressure Level (𝑳 𝑷 )

Sound Level Examples
• Sound requires energy (pushing atoms/molecules through a distance), and therefore a power
• Sound is characterized in decibels (dB), according to:
• Sound (Pressure/Intensity) Level = 10log(I/I0) = 20log(P/P0) dB
• I0 = 1012 W/m2 is the sound intensity threshold (0 dB)
• P0 = 2105 N/m2 is the sound pressure threshold (0 dB)
• Atmospheric pressure is about 105 N/m2
Examples:
• 60 dB (conversation) means log(I/I0) = 6, so I = 106 W/m2 and log(P/P0) = 3, so P = 2102 N/m2 = 0.0000002 atmosphere!!
• 120 dB (pain threshold) means log (I/I0) = 12, so I = 1 W/m2 and log(P/P0) = 6, so P = 20 N/m2 = 0.0002 atmosphere
• 10 dB (barely detectable) means log(I/I0) = 1, so I = 1011 W/m2 and log(P/P0) = 0.5, so P  6105 N/m2
46

Some Typical Sound Amplitude Values
47

All Shapes of Waveforms
• Different Instruments have different waveforms
• a: glockenspiel
• b: soft piano
• c: loud piano
• d: trumpet
• Our ears are sensitive to the detailed shape of
waveforms!
• More waveforms:
• e: French horn
• f: clarinet
• g: violin
48

Most Sound Stimuli are Complex
49

Complex Sound = Sum of Sines
(Fourier Theorem Revisited)
J.B.J. Fourier
(1768-1830)
Fourier Sound Applet
50

1
2
3
0
5
6
8
7
4
1
2
3
0
5
6
8
7
4
90 degrees out of phase
180 degrees out of phase
1
2
3
0
5
6
8
7
4
In phase
Sound Wave Interference and Beats
– When two sound waves are present, the superposition leads to interference
• by this, we mean constructive and destructive addition
• Two similar frequencies produce beats (“beating” of sound amplitude)
signal A
signal B
A + B beat
(interference)
in phase: add out of phase: cancel
51

Sensitivity of the Human Ear and Equal-Loudness Relations
– We can hear sounds with frequencies ranging
from 20 Hz to 20,000 Hz
• An impressive range of three decades
(logarithmically)
• About 10 octaves (factors of two)
• Compare this to vision, with less than one octave!
• Most sensitive to 2,000 Hz to 5,000 Hz
52

Sensitivity of the Human Ear and Equal-Loudness Relations
– If we play two pure tones (sinusoidal sound waves)
with the same amplitude but different frequencies
• One may sound louder than another
• The ear does not hear low or high frequencies
as well as mid-range ones (speech)
• This can be shown with equal-loudness curves
which plot perceived loudness on the axes of
true loudness and frequency.
53

Some Functional
Aspects of Hearing

Minimum Audibility Curve
Average detection threshold for
18-yr-olds for 1KHz tone at sea
level is
20 microPa (μPa)
Minimum occurs at approx. 3 KHz
Binaural intensity thresholds are 3
dB lower than mono

Clinical Audiogram (dBHL)
dB-HL (Hearing Level) uses a different
reference level for each test
frequency.
That reference level represents the
average threshold (18 yr-olds)
demonstrated at that frequency.
Hence, a value of 0 dB-HL means
“average” hearing level at the
frequency under test.

Loudness
Stevens’ SONE SCALE of Loudness Perception
Perceptual Anchor:
1 sone = loudness of 1 KHz
at 40 dB (40 phons)
Find the dB level that is twice as loud (2 sones)
or half as loud (0.5 sones), etc. and construct a
scale.
[i.e., Magnitude Estimation]
The psychological magnitude of sound (i.e.,
“Loudness”) grows at a slower rate than the
physical magnitude of the sound stimulus.

Loudness Using magnitude estimation techniques, S.S. Stevens has
quantified this nonlinear relationship as:
L = k * P0.6
= k * I0.3
L=loudness; P=sound pressure (µPa)
I=sound intensity (pW/m2)
Stevens’ Power Law; Linear in log-log plot; slope ≈
exponent
log(L)=log(k)+0.3 log(I) straight line
log(L)≈0.3 log(I)
Hence, a log unit increase (10dB) of
intensity yields 0.3 log (100.3 or 2-fold)
increase in loudness.
Note: Binaural presentation perceived as approx. 2x more
loud than monaural equivalent.

Sone Scale Landmarks
Normal conversation 1-4
Automobile @ 10m 4-16
Vacuum cleaner 16
Major roadway @ 10 m 16-32
Long-term hearing damage dosage 32+
Jackhammer @ 1m 64
Brief-exposure hearing damage 256
Pain threshold 676

Equal Loudness Contours
Frequency differentiation is flattened at high amplitudes; Speech and music sounds “tinny” at
high loudness levels; Remember change in cochlear nerve tuning at higher intensity levels.

Temporal Summation (< 200 msec)
Complements Binaural (i.e., Spatial) Summation

Tonal Masking:
Psychophysical Tuning Curves
Fixed test tone (e.g., 1KHz @ +10 dB)
Frequency of masking tone varied
How intense must masking tone be in order to
make the test tone indiscriminable?
Plot of masking intensity thresholds reveals
frequency tuning of underlying auditory
processing channel(s)

Multiple “Frequency Channels”
Revealed by Masking Curves

Localization of Sound
• At low frequencies (< 1000 Hz), detect phase difference
• wave crest hits one ear before the other
• “shadowing” not very effective because of diffraction
• At high frequencies (> 4000 Hz), use relative intensity in both ears
• one ear is in sound shadow
• even with one ear, can tell front vs. back at high freq.

Localization Accuracy vs. Frequency
Signature of a dual-mechanism process.

Localization Accuracy vs. Frequency:
Low Freq – Interaural Time Difference
High Freq – Interaural Intensity Difference
ΔIΔT

Sound Shadowing
(Interaural Intensity Difference –IID)
High-frequency sound waves are “blocked” by the
human head and cast a “shadow” at the far ear
(Strong IID cue)
Low-frequency sound waves wrap easily around
the head and cast little or no sound shadow (Weak
IID Cue)
ΔI

dBW
– The decibel watt or dBW is a unit for the measurement of the strength of a signal expressed
in decibels relative to 1 watt.
𝑑𝐵𝑊 = 10 log
𝑂𝑏𝑠𝑒𝑟𝑣𝑒𝑑 𝑃𝑜𝑤𝑒𝑟
1 𝑊𝑎𝑡𝑡
– It is used because of its capability to express both very large and very small values of
power in a short range of number.
0 dBW = 1W
100W amplifier is 20dBW
1000W amplifier is 30dBW
71

– The dBm abbreviation is used to express the root-mean-square (RMS) power of a sine-wave
signal with respect to 1 mW reference power.
𝑑𝐵𝑚 = 10 log
1𝑚 𝑊𝑎𝑡𝑡
– A power of 1 mW dissipated into a 600-ohm load results in a 0.775 V RMS Voltage.
0 dBm = 1mw, 600 Ohm load (0.775 V)
dBm
---------------------------------------
---------------------------------------
t
2.16
v
v
i
600 Ohm
72

N(dBm)
Notes
1- The power of 1 mW is defied as 0dBm.
2- when dissipated into the load values , different voltages results.
The above formula assume that the voltage is measured across identical impedances (600
ohm)
73
𝑁(𝑑𝐵𝑚) = 10 log
1𝑚 𝑊𝑎𝑡𝑡
𝑁(𝑑𝐵𝑚) = 20 log
𝑂𝑏𝑠𝑒𝑟𝑣𝑒𝑑 𝑉𝑜𝑙𝑡𝑎𝑔𝑒
0.775 𝑉

dBu
– Since in audio equipment usually we handle voltages rather than powers, an alternate method of
expressing audio signal levels is dBu.
𝑑𝐵𝑢 = 20 log
0.775 𝑉
– The reference signal level is 0.775 V RMS
– The dBu concept assumes a near-zero source impedance and a near-infinite load impedance.
– Under this idealized open load conditions, the source does not dissipate any power into the load.
– 0 dBu = 0.775 V
Rs=0
Load
RL= ∞
74

dBV
– Another method of expressing Audio signal Level.
𝑑𝐵𝑉 = 20 log
1𝑉
– This is an open-load voltage concept and reference voltage is 1V .
– The dBV is used by microphone manufacturers.
– 0 dBV = 1 V
75
Rs=0
Load
RL= ∞

dBFS
– Decibels relative to full scale (dBFS or dB FS)
– It is a unit of measurement for amplitude levels in digital systems, such as pulse-code
modulation (PCM), which have a defined maximum peak level.
𝑑𝐵𝐹𝑆 = 20 log
𝑆𝑦𝑚𝑏𝑜𝑙 𝑉𝑎𝑙𝑢𝑒
Maximum Symbol Value
• dBFS: Level compared to full scale.
• Always a negative number.
• 0dBFS is the loudest signal.
76
Full Scale

dBFS
77
EBU Digital −18 dBFS = +4 dBu = 0 VU
SMPTE Digital −20 dBFS = +4 dBu = 0 VU

78
PPM (Peak Program Meter ) vs VU Meter (Volume Unit Meter)
− A peak meter tracks the level of the
signal instantaneously; therefore its
reading is identical to the signal
level.
− An average reading takes time to rise
and fall, but the resultant readout is
reflective of our perception of
loudness.
− We can say that an average readout
seems lazy compared to peak
readout.

− Unfortunately, audio signals are complex waveforms (not pure sine) and their rms value is not 0.707
times peak but can range from as small as 0.04 times peak to as high as 0.99 times peak.
− To solve this problem, a special instrument is designed for measuring speech and music and calibrated
in units called VU.
− The dBm and the VU are almost identical; the only difference is their usage.
− The instrument used to measure VU is called the volume indicator (VI) instrument. (Some users ignore
this and incorrectly call it a VU meter.)
− Both dBm meters and VI instruments are specially calibrated voltmeters.
− Consequently, the VU and dBm scales on these meters give correct readings only when the
measurement is being made across the impedance for which they are calibrated (usually 150 or 600
Ω).
79
The VU and the Volume Indicator (VI) Instrument

– VU is a moving coil overage – meter type audio signal Level measuring instrument. (Introduced in 1939)
– It is fitted with two scales:
A VU scale marked 0 ( reference deflection ) extending from -20 to +3
A percentage scale, with 100% corresponding to 0VU reading
– Usually 0VU reference level corresponds to +4dBu under steady state sinusoidal voltage conditions.
• Much the same as dB meter (Used for general level setting.)
• Response peak to average
• Ballistics rated at @300mS settle time (rise & fall).
• Similar response to the human ear.
80

moving coil overage
81
SMPTE Digital −20 dBFS = +4 dBu = 0 VU (or 0 dBVU)

• Two different types of scales are available for VI meters
• Scale (a) is a VU scale (recording studio use), and scale (b) is a
modulation scale (broadcast use).
• On complex waveforms (speech and music), the readings observed
and the peak levels present are about10 dB apart.
• This means that with a mixer amplifier having a sine wave output
capability of +18 dBm, you are in danger of distortion with any signal
indicating more than +8 VU on the VI instrument ( +18 dBm – [ +10 dB]
peaking factor or meter lag equals +8 VU).
82

PPM (Peak Program Meter )
– Whereas the VU meter reflects the perceptual mechanism of the human hearing system, and thereby
indicates the loudness of a signal, the PPM is designed to indicate the value of peaks of an audio
waveform.
– Peak Program Meter, is a peak-reading instrument capable of displaying audio signal transients.
– Some current design have 10 ms attack time (rise time) and a 2.8 ms (or 1.7) fall back time.
– Neither the scale nor the display are standardized
83(Left) BBC-type peak programme meter (PPM). (Right) European-type PPM.

Reference Levels, Scales, Meters and DR
84

Reference Levels, Scales, Meters and DR
85

Dynamic Range
– Dynamic range is the difference between the quiet
and loud bits.
86

Compressor
– A compressor reduces dynamic range.
• Peaks reduced and overall volume
boosted.
• Same average audio level after
compression.
• Uncompressed audio is ‘dynamic’.
• Compressed audio is ‘loud’.
87
Relative levels before and after compression.
(a) Before compression the fluctuations in level make these
three instruments impossible to balance.
(b) After compression the balancing task is easier.

88
Compressor
− The nature of the signal between one stage and
another is shown to the left of the chain, along with
a sample graph that is based on the settings shown
under each control name.
–Compressor controls
Threshold controls the level where
compression begins.
Ratio control the amount of compression
above the threshold.
Attack controls the time delay before
compression starts.
Release control the time after compression
finishes.
Voltage-controlled
amplifiers

Compressor
– Sample transfer functions of
compressors, limiters and
expanders, and their effect
on input signals.
– The bottom graphs show the
transfer function of each
processor, all with threshold set
to 0 dB.
– The top row illustrates the
relationship between the input
and output levels.
89

Compressor Controls
90
− The threshold is set to -40 dB.
− Levels below the threshold are not treated, and the input to
output ratio is 1:1.
− Above the threshold, input signals are brought down in level.
− In this specific illustration the input to output ratio above the
threshold is 2:1
Transfer characteristics graphs

Compressor Controls
91
The effect of different ratios on the level envelope of a waveform

Compressor Controls
92
Different ratios on transfer characteristics graphs.
In all these graphs the threshold is set to –12 dB and the input signal is -6 dB (6 dB above the threshold).
With a 1:1 ratio the output signal leaves the compressor at the same level of -6 dB (6 dB above the threshold)
With a 2:1 ratio the signal leaves at -9 dB (3 dB above the threshold)
With a 6:1 ratio the signal leaves at -11 dB (1 dB above the threshold)
With ∞:l ratio the signal leaves at the same level as the threshold.

Compressor Controls
93
− The effect of different attack
and release settings on a
waveform:
(a) The original levels before
compression
(b–e) The resultant levels after
compression.

Compressor Controls
94
− The input to the time stage arrives from the scale stage and denotes the provisional amount of gain
reduction.
− The attack within the time stage slows down the gain reduction growth.
− Note that, even when the input to the time function starts to fall (2.5 ms), the output still rises as the
provisional amount of gain reduction is still higher than the applied amount.
− Only when the provisional amount falls below the applied amount (4 ms) does the latter start to drop.
At this point, the attack phase ends and the release phase starts.

Compressor Controls, Hold
95
− Very few compressors provide a
hold parameter, which is also linked
to the time function.
− This simplified illustration shows the
input levels at the top, the amount
of applied gain reduction in the
middle and the post-compression
level at the bottom.
− We can see that, once the original
level drops, the compressor holds
the gain reduction for a while before
release.

96
Compressor Controls, Hard and Soft Knee
− Soft-knee for more-transparent
compression.
− Hard-knee for more effect.
− With hard-knee, the ratio of
compression is attained instantly as
the signal overshoots.
− With soft-knee the full ratio is
achieved gradually within a
transition region that spreads to both
sides of the threshold.

Headroom
– Headroom is the difference between the average level and
peak level.
• If average at +4dB, peak at +26dB : 22dB of headroom.
– Analogue headroom is ‘kinder’ than digital headroom.
• Distortion appears before peak.
• Analogue headroom rolls off at the peak.
• Harmonic distortion.
– Digital headroom is sudden and ‘cruel’.
• No distortion before peak.
• After each sample gets to all 1’s, there is nowhere to go!
• Any sample higher than this folds over.
• Massive distortion and clicking!
97

98
Equalizer
The four basic bands of the frequency spectrum
− Distinguishing frequencies and mastering their manipulation is perhaps the greatest challenge
mixing has to offer.
− In simple terms, equalizers change the tonality of signals. This simple ability is behind many, many
applications

Reading the meter
– As a general rule there is 8dB difference in levels between VU and PPM. If the VU meter is set to 0dB the PPM is set to -8dB.
– Most meters do not obey the basic VU or PPM ballistics.
– This applies to both analogue and digital meters.
– Accurate analogue ballistics are expensive. (Accurate but small analogue meters are difficult or impossible to build).
– Accurate digital ballistics need to be modelled.
– Meters can be set to peak at 0dB or reference at 0dB.
– Reference is the point at which distortion starts.
– Noticeable distortion begins above reference.
– Audio level kept below reference.
– Audio peaking to about +3dB is often ‘acceptable’.
– Peak is the maximum possible audio signal.
– No signal possible above peak.
– In analogue audio the voltage cannot go any higher.
– In digital audio the samples are all 1’s.
99

LUFS (Loudness Units Full Scale)
– EBU compliant meters have both a relative scale, measured in LU,
and an absolute scale, measured in LUFS.
• 1 LU unit is equal to 1 dB, so raising program material by one LU
is the same as raising it by 1 dB.
• The loudness units measurement system is weighted to the real
perception of audio rather than just the electrical signal.
• The “FS” in LUFS stands for Referenced to Digital Full Scale.
• LUFS is a measurement of loudness units designed in line with
ITU-R BS.1770 recommendation about the loudness and true-
peak level of audio.
– When loudness is displayed on a relative scale, a Loudness Unit (LU)
value of 1 is equal to 1 LUFS from the Loudness Target, so a readout
of -2.1 LU means your audio is 2.1 LUFS below your Loudness Target.
101

Loudness Normalization
– The loudness algorithm works by pre-filtering, in two stages, a mono channel of audio with
a set of EQ curves designated K - Weighting to differentiate them from A, B and C
weighting.
– The curves attenuate bass frequencies, and account for the acoustical anomalies of the
human head, mimicking human hearing.
K-Weighting Filter Curve
103

– The algorithm handles up to five channels in total—one being mono, two being stereo, five
being surround (the LFE channel is not measured).
– The loudness of each channel is measured and then averaged and summed to produce a
single program loudness value (Integrated Loudness).
IRU-R BS.1770 Channel Processing and Summation
104
surroundmono
RLB (revised low frequency B-weighting)

– The loudness value is averaged over time, so the algorithm overlooks momentary fluctuations
in loudness and quietness via a gating threshold.
Loudness Integration and Gating Charts
105

Measurements on an EBU-R128 Standards Loudness Meters
106
(The statistical distribution of short term
loudness within a program, dynamic
range of the whole track)
(The accurate measurements of
(possible) inter sample peaks)
Momentary and Short Term values
are essential for live broadcast
(target level is set to -23 LU)

– Loudness metering explicitly takes human perception of hearing into account namely the measurements
of Momentary Loudness (M); Short Term Loudness (S); Integrated Loudness (I); Loudness Range (LRA) and
True Peak readings:
– Momentary Loudness measures the loudness of the past 400 Milliseconds.
– Short Term Loudness measures the loudness of the past 3 Seconds.
– Integrated Loudness (or Program Loudness) indicates how loud the program is on average, and is
measured over entire duration of material. (total average volume)
– Loudness Range (LRA) quantifies, in LU, the statistical distribution of short term loudness within a program. It
indicates the dynamic range of the whole track.
– True Peak Level indicates accurate measurements of (possible) inter sample peaks.
107

108
– The Integrated Loudness Value is the value to watch when mixing to the recommended -23 LUFS level.
– On a relative scale, with target level set to -23 LU, Integrated Loudness value will read 0 LU, and on an
absolute scale this number will read -23 LUFS.
– The LRA is also a number for musicians and mixers to keep an eye on. It indicates the dynamic range of the
whole track, so a low LRA (-1 LU to -3 LU) indicates material with a narrow dynamic range (possibly over-
limited or highly compressed), and a high LRA (-6 LU to -9 LU or more) indicates material with a wider
dynamic range.
– For techno and most dance music, the LRA will be low by nature of the genre, but singer-songwriters and
recording musicians would probably want to see larger LRA numbers.
– Momentary and Short Term values are essential for live broadcast and audio production for short form TV
content, and are useful for podcast recordings too.
– Last but not least, the True Peak reading lets us know if any inter sample peaking is occurring.

109

LUFS, LUKS & LU’s
110
A stereo -23 dBFS
1kHz sine tone will
read as -23 LUFS on
the absolute scale
Loudness Target =-23 LUFS
A stereo -23 dBFS 1kHz sine
tone will read as 0LU on
the relative scale.
– LUFS and LUKS (Loudness K-weighted Full Scale) refer to the same thing—Loudness Units Referenced to Digital Full Scale.
– 1 LU unit is equal to 1 dB, raising program material by one LU is the same as raising it by 1 dB.
– A stereo -23 dBFS 1kHz sine tone will read as -23 LUFS on the absolute scale, and as 0 LU on the relative scale, if you have
set your Loudness Target to -23 LUFS.

– LKFS and LUFS are referenced to full scale digital even though they are technically power
measurements (RMS is also a power measurement).
LUFS, LUKS & LU’s
A 0dBFS 1kHz sine tone will measure -3LUFS (as well as -3dB RMS)A -20dBFS 1kHz sine tone will measure -23LUFS (as well as -23dB RMS)
111

– Please note that Logic X’s RMS meter in stereo misreads by 3dB and should measure -20 dB in this
calibration test.
LUFS, LUKS & LU’s
A -20dBFS 1kHz sine tone will measure -23LUFS (as well as -23dB RMS)
112

Target Levels for Mixing to EBU-R128 Standards
– The current target loudness level for broadcast material, as stipulated in EBU-R128, is an
integrated loudness level of -23 LUFS (±0.5 LU) with a maximum permitted true peak level of
-1 dBTP (or -2dBTP if being compressed to codec like mpeg layer 2) and a maximum short
term loudness of -18 LUFS (≤ 0.5 +LU) .
113

Microphones
A microphone is a recording instrument
used to convert sound waves into an
electrical equivalent that can be stored,
transmitted, and played back through an
audio sound system.
116

Microphone Level
– Impedance for professional quality microphones are standardized at 150Ω.
– A typical moving-coil microphone, with source impedance of 150 Ω, generates an open
load voltage of 100µV to 1 mV at 74 dBSPL (or 94 dBSPL)
150Ω
150Ω
Microphone
Mic-preamp
Output impedance Output Level
ECM 77 150 Ω +- 20%(bal) -52 dB ±2dB
M88 200 Ω -52dB
117

Line Level
– The audio signal source output impedance is 600 Ω , balanced and floating
– A standard operating level (SOL) for old equipment's was chosen +4 dBm into 600 Ω
– The loading input impedance of broadcast audio equipment is 600 Ω , balanced and floating.
– Modern studies operate in voltage matching configuration.
– The signal level is expressed in dBu and the SOL is +4dBu.
– The output impedance is 50 Ω and less.
– The input impedance is 20K Ω and Higher.
– For professional and consumer products, the SOL is about – 10 dBu and less (sometimes -6 dBu )
– The output impedance is Low and input impedance is high.
300Ω
300Ω
LoadSource 600Ω
25Ω
25Ω
20KΩ
118
Audio Signal Source
Audio Equipment
Audio Equipment
Audio Equipment
50Ω

N
S
Output Diaphragm
Moving coil
attached to
Diaphragm
Diaphragm
(front plate)
Spacer
Back plate
(fixed)
Output
Ribbon Microphone
Capacitor Microphone
How Microphones “Hear”?
Dynamic Microphone
119
A type of microphone design that uses
acoustical energy and mechanical
vibration as the means for producing
the electromagnetic signal required for
analog recording
A type of microphone design using an
electronic device to record variations in
amplitude and frequency; require an
external power source to operate.
Small ribbon (rather than coil)
Similar to condenser mics in pickup
quality and sensitivity

Microphones – Transducer Type
A) A dynamic microphone is less
sensitive to sound because the
transducer is self-powered by the
sound of the subject’s voice.
B) A condenser microphone is more
sensitive to sound because the
transducer is powered by a battery
or phantom power source.
120

Dynamic microphone
It Uses a small coil that moves within a magnetic field when
activated by sound
Pros:
• Rugged and durable
• Can withstand weather and rough handling
• Great for ENG (electronic news gathering)
• Low Cost
• No power necessary
Cons:
• Larger and heavier
• Not the best choice for quality audio
121

Condenser microphone
It Has a plate that moves against another plate to produce a signal
Pros:
• More sensitive
• Better quality
• Can be small
Cons:
• Fragile and sensitive to shock and temperatures
• Needs power
• Battery
• Phantom power supplied by camera or audio mixer
• Expensive
122

Condensers microphones –
Battery powered versus
Phantom Powered…
123

Ribbon microphone
– Uses a small ribbon (rather than coil) moving in a
magnetic field
– Similar to condenser mics in pickup quality and
sensitivity
– Produce warm, rich sound
– Very sensitive to rough handling and moving air
– Rarely used outside an audio booth
LuckyLouie. RCA 44 Ribbon
Microphone. 4 November 2007.
<http://en.wikipedia.org/wiki/File:Rca
44.png>.
124

Microphones – Form Factor
Microphones are classified by form factor depending on their general purpose and application.
125

Microphones – Polar Pattern
– A visual reference to how well a microphone picks
up sound within 360° of its central axis.
– The most common polar patterns:
A) Omnidirectional
B) Bidirectional;
C) Cardioid
D) Supercardioid
E) Hypercardioid
F) Ultracardioid (or shotgun)
126

Omnidirectional Microphone Patterns
– The omnidirectional microphone hears equally well from all directions.
127

Cardioid Microphone Patterns
– The unidirectional microphone favors sounds that are in front of it.
– Its pickup pattern is heart shaped, hence the term cardioid.
128

Hypercardioid Microphone Patterns
– The hypercardioid pickup pattern is narrower than the cardioid and has a longer reach.
– Hypercardioid mics can also hear sounds coming from behind the mic.
129

Microphone sensitivity specification
This specification tells how much electrical
output (in thousandths of a volt or
"millivolts") a microphone produces for a
certain sound pressure input in dBSPL
(usually 94dBSPL)
130

How mics are used?
• Handheld mic
• Lavalier (lapel) mic
• Boom (shotgun) mic
• Desktop mic
• Headset mic
131

Handheld Microphone
• Held by the talent
• Allows for talent to control mic’s location
• Reporter can easily move it closer to interview
• Singer can control intimacy of the sound by holding it closer
• Field reporters - need dynamic handheld (rugged, durable)
• Singers - need condenser handheld (better sound quality)
• Disadvantages
• Can be distracting (takes up visual space)
• Talent’s hands are not free
• Quality of sound depends on proper placement by talent
132

Lavalier Microphone
• Small, rugged, omnidirectional
• Usually clipped to shirts six to eight inches below talent’s chin
• Advantages
• Talent can move freely
• Mic never moves
• Sound is consistent
• Disadvantages
• Each on-camera subject requires mic
• Requires proper setup and placement
• Clothing may rub mic or muffle sound
A low-profile microphone designed with an
electret condenser transducer element. May
be attached directly to the talent’s clothing.
133

Lavalier Microphone
“Dressing” lapel mics (hiding wires)
• Under clothing
• Have talent run microphone under the shirt
• Clip mic to tie, lapel, collar or button flap
• Careful of material rubbing mic or obstructing sound
• Outside clothing
• Run mic up the back of talent (outside clothing) and over
the talent’s shoulder near the collar (under collar if
possible)
• Clip mic to lapel, collar, or button flap
• Use gaffers tape (on talent’s back) to keep the mic cable
from moving
134

Boom Microphone
• A commonly used term for an ultracardioid
microphone with an extremely narrow pickup pattern.
• Unidirectional mics intended to pick up sound without
being seen on camera
• “Shotgun” microphone mounted to the end of a
telescoping boom pole
• Pole has shock-mount to absorb all
movement
• Can be held above talent or at low angle
below
135

Desktop Microphone
• Microphone mounted on a small stand
• Used for fixed positions
• Often seen in:
• Game shows
• Speeches
• Board rooms
• Churches
136

Desktop Microphone
Setup for Multiple Desk Microphones
To avoid multiple-microphone interference when using several desk mics for a panel
show, place them at least three times as far apart as any mic is from its user.
The 3 to 1 rule:
The distance between
microphones should be at least
3 times the distance from each
microphone to its intended
sound source.
137

Headset Microphone
• Gives talent the freedom of movement while keeping the mic close
to the mouth
• Sometimes includes an earphone for talkback
• Important for loud arenas and stadiums
• Used by sportscasters, musicians/performers and public speakers
• Small, wireless headsets with no earphone are sometimes called
“countryman” mics
138

Wireless Microphones
• Involves a microphone with a transmitter and a wireless
receiver
• Can come in any style
• Lav, headset, hand
• Allows complete freedom of movement without wires
• Transmits through frequency bands
• UHF
• Least interference
• More expensive
• VHF
• Cheaper
• Most interference
139

Windscreens
• Designed to let normal sound frequencies through and
filter out wind noise
• Made of acoustic foam rubber
• Windsock
• Used for shotgun mics
• Ragged cloth that is pulled over the windscreen
140

Microphones – Techniques
Dynamic handheld microphones should be positioned no further than a few inches
from the subject’s mouth.
141

When using any microphone outdoors,
it’s a good idea to attach a windscreen.
142

A handheld microphone is supported by a
microphone stand for hands-free operation.
143

When shooting an on-camera
interview, neatly dress the
microphone to minimize any
visual distractions.
144

A boom pole is used to position
a shotgun microphone within a
few feet of the subjects in this
scene.
145

A pistol grip is used in lieu of
a boom pole for acquiring
the sound of footsteps on
leaves.
146

Different techniques can be used for
positioning a microphone with a boom
pole-either above the subjects or below
them. Beware of getting in too close
though!
147

The Audio-Technica ES961 is a
cardioid condenser boundary
microphone designed for use on
a flat surface such as a
conference table or floor.
148

When given a choice, use an external microphone rather than the built-in
microphone on the recorder or camera.
149

A wireless microphone like this one is
designed for remote field
production.
150

A student uses a USB
microphone directly
connected to a computer.
151

Professional microphones
and recording systems are
equipped with balanced
XLR connectors.
152

Nobody likes working with a
tangled cable. Here, a
student uses the over-under
wrap to properly coil a
microphone cable.
153

During a recording, keep the audio levels within the green portion of the scale,
or good range, at roughly 50%–80%.
154

Digital VU meters often include a clipping warning light that illuminates whenever
sound levels exceed the distortion threshold.
155

Microphones – Proximity Effect
The proximity effect is an acoustic
phenomenon that boosts the bass
frequencies of your voice as you
move progressively closer to the
microphone diaphragm.
156

Speakers
– Speakers vibrate and push on the air
• pushing out creates compression
• pulling back creates rarefaction
– Speaker must execute complex motion according to desired waveform
– Speaker is driven via “solenoid” idea:
• Electrical signal (AC) is sent into coil that attached to speaker cone and surrounds a
permanent magnet
• Depending on direction of current, the induced magnetic field either lines up with magnet
or is opposite
• Results in pushing or pulling magnet in coil (attracting/repelling), and thus pushing/pulling on
center of cone
161

Push Me, Pull Me
– When the center of the speaker cone is kicked, the whole cone can’t respond instantaneously
• the fastest any mechanical signal can travel through a material is at the speed of sound in the material
– The whole cone must move into place well before the wave period is complete
• Otherwise, different parts of the cone might be moving in while others are moving out (thus canceling
the sound)
• If we require the signal to travel from the center to the edge of the cone in 1/N of a wave cycle (N is
some large-ish number):
• available time is t = 1/Nf = /Ncair
• ripple in cone travels cconet, so radius of cone must be < ccone/Ncair
• Basic point is that speaker size is related to wavelength of sound
• Low frequency speakers are big, high frequency small
162

Speaker Wiring
Speaker Wire Chart
Copper
Wire
Size
2
Ohm
4
Ohm
6
Ohm
8
Ohm
22 3 6 9 12
20 5 10 15 20
18 8 15 23 30
16 12 25 37 50
14 20 40 58 77
12 31 61 92 123
10 50 98 147 196
Maximum loss of 5% of nominal speaker
impedance.
Example 8 Ω speaker, 5% = .4 Ω.
22 gauge wire has resistance of 0.0164
Ω per foot
1 wire each way (2 wires) = 0.0328 Ω / Ft.
.4/.0328=12.195 or 12' for 5 % loss.
http://circuitcalculator.com/wordpress/2007/09/20/wire-parameter-calculator/
http://en.wikipedia.org/wiki/American_wire_gauge

– Physical analog media such as vinyl records have been supplanted by invisible digital
media files that can be purchased online and downloaded to a portable MP3 player
in a matter of seconds.
File-Based Recording – Audio Settings
164

File-Based Recording – Audio Settings
The quality of an digital audio stream is defined by three variable:
1. Sample Rate specifies the number of samples recorded every second as designated in
kilohertz (kHz) units. For example, a 48 kHz sample rate indicates that the recording was
sampled 48,000 times per second. 44.1 kHz is the industry standard sample rate for audio
compact disc.
2. Bit Depth specifies the number of bits used to encode the value of each sample. Digital
audio is usually recorded at either a 16-bit or 24-bit bit depth.
3. Bit Rate specifies the number of bits per second transmitted during playback for
compressed audio file formats such as MP3.
165

Transcoding
– It is the process of converting a file
from one format to another, for
example, converting a Microsoft
Word document into PDF or a
WAV audio file to MP3.
– Here, Adobe Media Encoder is
being used to transcode an
uncompressed AIFF audio file to a
WAV file.
File-Based Recording
166

File-Based Recording
Apple iTunes includes a built-in audio encoder for converting uncompressed AIF and WAV files
(from an audio compact disc) to a compressed file format such as ACC or MP3.

MIDI Standard
(Musical Instrument Digital Interface)
168

MIDI
– Digital sound is relatively straight forward to record a complex sound that consists of many
frequencies. However, it is quite difficult to generate (or synthesize) complex sound.
– There is a better way to generate high quality music. This is known as MIDI — Musical
Instrument Digital Interface.
– MIDI is a protocol that enables computer, synthesizers, keyboards, and other musical
devices to communicate with each other (1980s ).
– It specifies the hardware connection between equipments as well as the format in which
are transferred between the equipments.
– General MIDI is a standard specified by MIDI Manufacturers Association.
– To be GM compatible, a sound generating device must meet the General MIDI system
level 1 performance requirement.
169

Setup:
– MIDI OUT of synthesizer is connected to MIDI
IN of sequencer.
– MIDI OUT of sequencer is connected to MIDI
IN of synthesizer and "through" to each of the
additional sound modules.
Working:
– During recording, the keyboard-equipped
synthesizer is used to send MIDI message to the
sequencer, which records them.
– During play back, messages are sent out from
the sequencer to the sound modules and the
synthesizer which will play back the music.
Typical Sequencer setup
THRU
THRU
THRU
IN OUT IN OUT
IN
IN
MIDI Interface/Sound Card
(Sequencer)
Synthesizer/Keyboard
MIDI Module A
MIDI Module B
Etc.
MIDI
170
MIDI message
During recording
During play back

MIDI: Data Format
– Information traveling through the hardware is encoded
in MIDI data format.
– The MIDI data format is digital
– The encoding includes note information like beginning
of note, frequency and sound volume; up to 128 notes
– The data are grouped into MIDI messages
– Each MIDI message communicates one musical event
between machines. An event might be pressing keys,
moving slider controls, setting switches and adjusting
foot pedals.
– 10 mins of music encoded in MIDI data format is about
200 Kbytes of data. (compare against CD-audio!)
171

MIDI: Data Format
– Information traveling through the hardware is encoded in MIDI data format.
– The encoding includes note information like beginning of note, frequency and sound volume;
upto 128 notes
– The MIDI data format is digital
– The data are grouped into MIDI messages
– Each MIDI message communicates one musical event between machines. An event might be
pressing keys, moving slider controls, setting switches and adjusting foot pedals.
– 10 mins of music encoded in MIDI data format is about 200 Kbytes of data. (compare against
CD-audio!)
172

MIDI files
– When using computers to play MIDI music, the MIDI data are often stored in MIDI files. Each
MIDI files contains a number of chunks. There are two types of chunks:
• Header chunk — contains information about the entire file: the type of MIDI file, number of
tracks and the timing.
• Track chunk — the actual data of MIDI track.
– The data section contains three 16-bit words, stored most-significant byte first. The first word,
<format>, specifies the overall organization of the file. Only three values of format are
specified:
• 0 single multi-channel track
• 1 one or more simultaneous track of a sequence
• 2 one or more sequentially independent single-track patterns
173

Tracks, channels and patches
– Multiple tracks can be played at the same time.
– Each track can be assigned to a different channel.
– Each channel can accept more than one track.
– Each channel is assigned a patch, therefore generates sound of a particular instrument.
174

How MIDI Sounds Are Synthesized
– A simplistic view is that:
• the MIDI device stores the characteristics of sounds produced by different sound sources;
• the MIDI messages tell the device which kind of sound, at which pitch is to be generated, how long the
sound is played and other attributes the note should have.
– There are two ways of synthesizing sounds:
• FM Synthesis (Frequency Modulation)—Using one sine wave to modulate another sine wave, thus
generating a new wave which is rich in timbre. It consists of the two original waves, their sum and
difference and harmonics.
• The drawbacks of FM synthesis are: the generated sound is not real; there is no exact formula for generating a particular
sound.
• Wave-table synthesis— It stores representative digital sound samples. It manipulates these samples, e.g.,
by changing the pitch, to create the complete range of notes.
175

MIDI Sound Attributes and The Amplitude Envelop
– The shape of the amplitude envelop has great influence on the resulting character of
sound. There are two different types of envelop:
• Diminishing sound — gradually die out;
• Continuing sound — sustain until turned off.
– The Amplitude Envelop
• Delay — the time between when a key is played and when the attack phase begins
• Attack — the time from no sound to maximum amplitude
• Hold—the time envelop will stay at the peak even before starting the decay phase
• Decay — the time it takes the envelop to go from the peak level to the sustain level
• Sustain — the level at which the envelop remains as long as a key is held down
• Release — the time is takes for the sound to fade to nothing
176

MIDI software
MIDI player for playing MIDI music. This includes:
• Windows media player can play MIDI files
• Player come with sound card— Creative Midi player
• Freeware and shareware players and plug-ins—Midigate, Yamaha Midplug, etc.
MIDI sequencer for recording, editing and playing MIDI
• Cakewalk Express, Home Studio, Professional
• Cubasis
• Encore
• Voyetra MIDI Orchestrator Plus
Configuration — Like audio devices, MIDI devices require a driver. Select and configure MIDI
devices from the control panel.
177

Summary - MIDI versus Digital Audio
Digital Audio
– Digital representation of physical sound waves
– File size is large if without compression
– Quality is in proportion to file size
– More software available
– Play back quality less dependent on the sound sources
– Can record and play back any sound including speech
Abstract representation of musical sounds and sound effects
– MIDI files are much more compact
– File size is independent to the quality
– Much better sound if the sound source is of high quality
– Need some music theory
– Cannot generate speech
178

Audio Tools and Connectors
179

Audio Connections
2 pin DIN connector
Phono connectors & 2.5mm (1/8”) jack
3.5mm (1/4”) jack
3 pin XLR plugs
3 pin XLR sockets
XLR to jack adaptor
XLR to Phono adaptor
180

+ Tip
+ Pin
Shield Shield
(Sleeve)
+ Tip
- Ring
- Pin 3
+ Pin 2
Shield
Pin 1
182
Audio Connections

183
Balanced and Un-Balanced Wiring

Balanced and Un-Balanced Wiring
– Broadcast equipment's use balance signals with 3-pin, XLP or canon, connectors, and
the standard operating level is +4dBu
– Professional and consumer equipment's use unbalance signal with RCA or phono
jack, connectors and the standard operating level is (-10 dBu) (sometimes -6dBu)
184

- Pin 3 N/C
+ Pin 2
Shield Pin 1
Shield
+ Pin 2
- Pin 3
Shield Pin 1 Jumper
Balanced Output
Balanced Input
Balanced to Unbalanced
185

Transformers
Isolation Step-down Step-up
Polarity
+
+
Polarity
+ +
186

Patch Panels
1 24
25 48
From Equipment (Outputs)
To Equipment (Inputs)
Normalling Jack Pair
188

189
Patch Panels
The right way to do it.

Audio Production and
Postproduction

Acoustics
• Most sound booths incorporate different types
of soundproofing materials
• Hard walls; tile floors will allow too much
reverberation
• Reverb - sound remaining after the original
sound stops
• Too much soundproofing causes a dead,
lifeless sound
• Ideal room for recording and listening has
free-standing, sound-absorbing items
(furniture, rugs)

Audio Postproduction Room
An audio postproduction room
normally has acoustically treated
walls and contains high-end
surround-sound speakers as well
as two stereo speakers for audio
monitoring, various amplifiers, an
audio mixing console, a large
video display, a computer
display, and a high-capacity
computer with audio processing
software.

Amplification
• The “gain stage” is the first stage in the console.
• Setting proper gain is critical for clear audio.
• Gain settings too low result in noise
• Gain settings too high result in distortion
• If gain is too low, turning up output levels will also
increase the level of the inherent noise.
• Proper gain settings maximize signal to noise ratio

Gain Setting
• Many recording and live sound consoles provide “clip
indicators” to assist in proper gain-setting
• When the “peak” indicator illuminates, distortion will
result.

Mixing
• Adjusting the relative levels of the audio signals leaving the mixer.
• Mixing is accomplished using the console’s faders or potentiometers.
• In most consoles, the potentiometers only cut or reduce levels.
• Faders full up = no cut
• Faders completely down = full cut
Linear faders

Meters Calibration
• All devices in the recording chain must be calibrated to assure proper levels.
• Using a steady tone, the console meter is set to 100% modulation or “0 VU.”
• Then all device inputs are adjusted so their meters match.
• When all devices are calibrated, then maintaining proper meter levels on the console will assure
that other devices are not over-driven.

Equalizer Types
• Graphic equalizer, Several varieties
• Shelving (highpass-lowpass) High or low cut or boost
• Parametric, Boost, cut, center freq & Q

Routing or Directing
– Sending the signal to a variety of destinations.
– Signals are transported via “buses.”
– Buses are simply wires or paths for the signal to travel.
Typical console outputs
• Main or stereo bus
• Subgroups
• Monitor or foldback send to headphones
• Effects sends to outboard effects devices
• Auxiliary sends
• Cue sends to headphones or cue speakers
• Different manufacturers use different terms
Stereo Master Fader
Subgroup Faders

Routing or Directing
Signals may be routed to
outboard devices either
“pre fader” or “post
fader.”
The pan control routes the
signal to the left and right
stereo channels.

The Mixing Console Connections

Audio Amplifier Types
• Pre amplifier
• Frequently used in low level applications
• Phonograph cartridge
• Microphone
• Intercom
• Buffer amplifier
• Often used to isolate one circuit from another and to match impedances
• Line amplifier/DA
• Often used to increase drive capability
• Split signals to feed several different pieces of equipment.
• Usually no change in level
• Power amplifier
• Usually used to increase current capability to drive speakers and headphones

Audio Software Terms
• Waveform: a visual representation of an audio signal.
• Clip: is a short segment of audio. It can be combined
with others to make an audio track.
• Track (Channel): an area upon which audio moves. It
may be a single waveform or could be made up of
clips.

Audio Software Terms (Continued)
• Channel
• Single Channel=Mono
• Two Channels=Stereo
• reproduction of sound using two or more
independent audio channels in a way that creates the
impression of sound heard from various directions, as in
natural hearing.
• Trim: removes everything from the left and right of
selected audio.
• Amplify: change the volume of selected audio.

Dolby Digital
Also known as AC3, a widely popular multichannel surround sound format
capable of supporting up to 7.1 channels of audio.
Container format OR wrapper
A unique kind of file format used for bundling and storing the raw digital bit
streams that codecs encode.
QuickTime
A multimedia container format released by Apple Computer in 1991.
Audio Software Terms (Continued)

Clipping
Waveform distortion that occurs when too much amplification
is applied to the audio signal during recording or mixing.
Equalization
The process of altering the frequency or pitch of an audio
signal.
Phantom Power
A 48-volt (+48V) electric current that’s applied by the
recording device or mixer to the capacitor of a condenser
microphone through an attached microphone cable.
Clipping, Equalization and Phantom Power

Analog
Digital
AMPLIFIER
SOUND PRESSUTRE
WAVE
MICROPHONE
SOUND PRESSUTRE
WAVE
VOLTAGE IN WIRE
ANALOG TO
DIGITAL
CONVERTER
COMUTER
MEMORY
DIGTAL TO
ANALOG
CONVERTER
AMPLIFIER
Analog vs. Digital
224

Idealized Model
Sampling / Reconstruction
f(t) f*(t)
quantizer
f(nT)
computer
g(nT)
Zero order hold
g(t)
smoothing
filter
(low-pass)
g(t)
A/D converter
D/A converter
sampler
Model of the sampling and de-sampling process
225

Deconstruction of Sampling
f(t)
analog
(a)
f(t)
quantized in amplitude
(b)
(c)
quantized in time
f*(t)
f(nT)
(d)
digital
Graphic representation of types of signals:
a) analog signal
b) quantized (continuous time) signal
c) sampled data signal
d) digital signal
Sampling Theorem
Must sample twice as high
as highest component
226

Foldover in time domain
Original
signalOriginal
signal
Sampling pulseSampling pulse
Resultant
Digitized
Waveform
Resultant
Digitized
Waveform
227

Digital Spectra and foldover in frequency domain
0
|H(f)|
digital
sampling rate
SR
Nyquist Frequency
SR
2
foldover sampling rate
foldover frequency =
1/2 sampling rate

– In any analog system, some of the voltage is what you want to measure (signal), and
some of it is random fluctuations (noise).
– SNR: Signal to Noise ratio captures the quality of a signal (dB)
SNR = 10 log
V2
signal
V2
noise
Vsignal
Vnoise
= 20 log
S/N Ratio
229

error
4
3
2
1
0
-1
-2
-3
-4
digital
Quantization levels
analog
0-1 1
maximum error
of ± 1/2
Error Distribution
230
Signal to Quantization Noise Ratio (SQNR)

– The quantization error (or quantization noise) is the difference between the actual
value of the analog signal at the sampling time and the nearest quantization interval
value.
– The largest (worst) quantization error is half of the interval (0.5).
– If we use N bits per sample, the range of the digital signal is: -2N-1 to 2N-1
– The worst-case signal to quantization noise ratio is given by:
– Each bit adds about 6 dB of resolution, so 16 bits enable a maximum SQNR = 96 dB.
SQNR = 20 log
Vsignal
Vquant - noise
= 20 log
2N-1
0.5
= N x 20 log 2 = 6.02N (dB)
231
Signal to Quantization Noise Ratio (SQNR)

96 dB
24 dB
16 bits
16 bits
4 bits
4 bits
232
Dynamic Range Problem

Low-level Signal Problem
+2
+1
0
-1
-2
intended
result
Folding Frequency
folded
a

Signal Reconstruction
SR/2 SR
f
a
Ideal reconstruction filter
hasn’t been found
Reconstruction Filter

Oversampling Converters
• Multibit oversampling
convert 44.1 K to
176.4 K (4x)
352.8 K (8x)
• 1-bit oversampling
1 bit at high rate
sigma-delta
MASH
128 x 1-bit =
8 * 16-bits

Multibit Oversampling
SR
2
SR
f
a
3 SR
2
f
a
a
t

Actual Converters
• Combine oversampling
first with 1-bit stream
second
• Resulting noise lower
4x 6dB
8x 12dB
• In computers ground isolation of
digital noise overwhelms everything else!

Audio Quality vs. Data Rate
Quality Sample Rate
(kHz)
Bits per
Sample
Mono/
Stereo
Data Rate
(kBytes/sec)
(uncompressed)
Frequency
Band
Telephone 8 8 Mono 8 200-3400 Hz
AM Radio 11.025 8 Mono 11.0 540-1700 KHz
FM Radio 22.050 16 Stereo 88.2
CD 44.1 16 Stereo 176.4 20-20000 Hz
DAT 48 16 Stereo 192.0 20-20000 Hz
A simple and widely used audio compression method is Adaptive Delta Pulse Code Modulation
(ADPCM). Based on past samples, it predicts the next sample and encodes the difference
between the actual value and the predicted value.

Digital Audio
240
- 4 Preamble Bits
- 24 Data Bits
- 1 Validity
- 1 User
- 1 Channel status
- 1 Parity (Even parity)

Validity bit
The validity bit shall be logic “0” if the audio sample word is suitable for conversion.
The validity bit shall be logic “1” if the audio sample word is not suitable for conversion. (There is no default
value for validity bit)
User data format
User data bits may be used in any way desired by the user. (The default value of the user data bit shall be
logic "0")
Channel status Byte 1 bits 4-7 indicate possible formats for the user data channel.
Channel status format
The channel status for each audio signal carries information associated with that audio signal (Examples:
length of audio sample words, number of audio channels, sampling frequency, time code, alphanumeric
source and destination codes, and pre-emphasis), and thus it is possible for different channel status data to
be carried in the two sub-frames of the digital audio signal.
– Channel status information is organized in 92-bit blocks, subdivided into 24 Bytes .
– The first bit of each block is carried in the Frame with preamble "Z".
– The specific organization follows, wherein the suffix 0 designates the first Byte or bit. Where multiple bit states represent
a counting number, tables are arranged with most significant bit (MSB) first, except where noted as LSB first.
Digital Audio Sub-frame Format
241

AES/EBU digital audio signal structure
242

Frame repeat
– 1 Frame : 1/48kHz = 20.83us (44.1kHz=22.67us)
Block repeat
– 1 Block : 20.83us ×192frame = 4ms (44.1kHz=4.352ms)
AES/EBU Interface bit rate
– 48kHz × 2CH × 32Bit = 3.072Mbps
After BPM (Biphase Mark) encoding
– 3.072Mbps × 2 = 6.144Mbps
AES/EBU Frames & Sub-frames & Data rate
243

X, Y and Z Sync Words (Preambles)
The subframe preambles starting with a
transition from negative to positive.
The subframe preambles starting with a
transition from positive to negative.
– Each preamble must transition to a different level from that of the last state of the bit
before it.
245

Common Audio File Types
File Type Acronym For Originally
Created By
File Info & Compression Platforms
.wav IBM
Microsoft
• Compressed or uncompressed
• One of the HTML5 audio formats
• Windows
• Plays in Web browsers that
support the .wav format of
HTML5 audio (Firefox, Safari,
Chrome, and Opera)
.mp3 MPEG audio layer 3 Moving Pictures
Experts Group
• Good compression rate with
perceivably high quality sound
• Cross-platform
•Plays in Web browsers that
support the .wav format of
HTML5 audio (Safari and IE)
.m4a MPEG-4 format
without the video
data
Moving Pictures
Experts Group
•AAC compression; same
compression as the MPEG-4 H.264
without the video data
Plays in Web browsers that
support the AAC format of
HTML5 audio (Safari, IE, and
Chrome)
246

File Type Acronym
For
Originally Created
By
File Info &
Compression
Platforms
.ogg or .oga Xiph.Org Foundation • Usually referred to as Ogg
Vorbis format
• One of the HTML5 audio
formats
Plays in Web browsers that
support the Ogg
Vorbisformat of HTML5
audio (Firefox, Chrome,
and Opera)
.mov QuickTime
movie
Apple • Not just for video
• supports audio track and a
MIDI track
• a variety of sound
compressors
• files can be streamed
• "Fast Start" technology
Cross-platform;
requires QuickTime
player
247

File Type Acronym For Originally
Created By
File Info &
Compression
Platforms
.aiff Audio Interchange
File Format
Apple compressed,
uncompressed
Mac, Windows
.au
.snd
Sun compressed Sun, Unix, Linux
.ra
.rm
Real Audio Real Systems compressed; can be
streamed with Real
Server
Cross-platform;
requires Real player
.wma Window Media Audio Microsoft
248

Domestic and Broadcast
Audio Delivery Formats
249

Domestic audio formats
CD
• The most popular domestic audio format
• Audio only, Stereo
• 44.1kHz sample rate
• Uncompressed PCM
Dolby Pro Logic
• Matrixed standard stereo
• Original domestic surround sound
Dolby Digital (AC-3)
• 5.1 surround sound
• Specified by DVD standard
• Very common in DVD’s
• Requires high end DVD player or separate
processor
DTS (Digital Theatre System)
• 5.2 surround sound
• Specified as a DVD option
• Requires a DTS decoder
• Not popular
250

Broadcast audio formats
AES/EBU
• Very high quality possible (Uncompressed)
• Multiple sample rates (48kHz, 44.1kHz, 32kHz)
• Multiple sample sizes (24, 20, 16 bits)
• Multiple channel types (Mono, stereo, dual channel, primary/secondary)
• Can be used to hold data (Dolby E for instance)
Dolby E
• Compressed digital multi-channel format
• Up to 8 audio channels
• Can be used for surround sound or multi-versioning
• Can hold metadata
• Can use AES/EBU in non-audio mode.(One whole AES/EBU connection for one Dolby E connection)
Dolby E
251

AES3: Audio Engineering Society Standard #3
– AES3: Audio Engineering Society Standard #3
– The AES3id interface (alternatively named AES-BNC
or AES-75 Ohm) applies unbalanced cable
connections
– This is the only difference to AES3 which applies
balanced cable connections.
252

Broadcast needs for audio
– Good quality audio is essential for high definition video.
– Stereo is a basic requirement.
– Basic surround sound is a good target to achieve.
• 5.1 surround sound is the obvious choice.
– Complex sound field generation is unnecessary.
22.2 multichannel sound system 253

4 Generations of audio reproduction
1st generation: Mono audio
– One speaker
– Emphasis on quality
– No attempt at sound field generation
– Easy and simple recording
Speaker
254

2nd generation: Stereo audio
– Two speakers
– Basic sound field generation
– Basic unidirectional realism
– Easy production & recording
Left speaker
Right speaker
255

3rd generation: Surround sound
– 4, 6, 8 or more speakers
– Good sound field generation
– Good all-round realism
– Complex production & recording
– Many different arrangements
– Most popular is 5.1
Left speaker
Right speaker
Centre speaker
Sub speaker
Left surround speaker
Right surround speaker
5.1 arrangement
256

4th generation : Advanced sound field generation
– Undefined number of speakers
– Intricate delays to create complex sound fields
– Expensive
– Very specific listening point
– Impractical for cinema use
– Impractical for broadcast use
– Greater possible use with headphones
– ‘Out of head’ experience

Up to 8 discrete audio channels.
– Commonly 5.1 : L, C, R, SL, SR, LF
– Uses AC-3 (Audio Coder 3) compression
– First used in “Batman Returns” in 1992.
Left Centre Right
Screen
Left surrounds
Low freq.
Right surrounds
258
Film Formats
Dolby Digital

Co-developed with Lucas Film.
– Provides THX compliance
– Greater realism over Dolby Digital
– Uses the same 5.1 as Dolby Digital
– Used the same data blocks
– Back surrounds mixed in cinema
Left Centre Right
Screen
Left surrounds
Low freq.
Right surrounds
Back surrounds
259
Film Formats
Dolby Digital Surround EX (Extended)

4 channels : L, C, R, S
– Basic mono surrounds
Left Centre Right
Screen
Mono surrounds
Film Formats
Dolby SR (Spectral Recording)
260

Up to 8 discrete audio channels.
– Normally 5.1
– Sometimes 7.1
– Channels can be used for any
configuration
– No defined channels for surround or LF
Left
Centre
Right
Screen
Right centre
Left centre
Left surrounds
Right surrounds
Low freq.
261
Film Formats
SDDS (Sony Dynamic Digital Sound)

Questions??
Discussion!!
Suggestions!!
Criticism!!
262

Sound and Hearing Functional Aspects

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