SOUND SYSTEMS

part - 1
Sound Engineering Course

Michele Begotti
E-mail: [email protected]
Tel.: +39 0522-274411
+39 335 6393929
1
 Books
Sound System Engineering D.Davis -
C.Davis, Second Edition, Focal Press,
1997.
Sound Systems: Design and
Optimization Bob McCarthy , Focal
Press, 2007.
Master Handbook of Acoustics F.
Alton Everest, Fourth Edition, McGraw-
Hill, 2001

2
 Program

1. Signal Path
2. Interactions with the real world
3. Reception
4. Design principles
5. Simulation

3
1. Signal Path

4
 1. Need for Amplified Sound

There are many different types of sound systems with purposes as diverse
as special effects and voice paging.

- Sound Reinforcement Systems : Supply a real-time amplification of a

live source to a listener ( Rental, Live Music, Church System, Theatre,
Lecture and Conference, Sports..)

- Reproducing Systems: Amplify sound from a signal storage ( Club,

Cinema, Home..)

- Public Address Systems: Supply both real time

annconcements and amplified sound from a signal storage (
Stations, Airports, Shopping Centers)
 1. Need for Amplified Sound
Yet most share common design criteria.

We can cover many of these design criteria using

soundreinforcement systems as design examples.

The aim of a sound reinforcement system is to enhance the sound of the

sources and guarantee the expected listening level .

Sound reinforcement system has to be designed in order to move the

talker closer to the listener and overcome the background noise.

Sound reinforcement system has to provide the same show to all the
listeners.
 2. Audio Chain
Music Source

Message
Signal AMP
Player
Processing

Mike

Electronic
Acoustic
Transmission
Sources Transmission
 3. Audio Sources

Live sources: microphones, pick up.

Reproduction sources ( cd/mp3/dvd player, tuner)

 3 Source: Microphones

What is a Microphone ?
Its an acoustic-to-electric transducer or sensor that converts sound
into an electrical signal

Microphone Characteristics

The most important characteristics of microphones for live sound applications

are their
- operating principle,
- frequency response and
- directionality.

Secondary characteristics are their

- electrical output and
- physical design.
 3 Source: Microphones

A transducer is a device that changes energy from one form into another,
in this case, it uses a thin membrane which vibrates in response to sound
pressure, this movement is translated into an electrical signal .

The operating principle determines some of the basic capabilities of the

microphone..

There is a lot of mikes types:

Dynamic
Condenser, capacitor or electrostatic
Electrect
Boundary
Piezo.
 3 Source: Microphones

Dynamic
Work via electromagnetic induction
Moving coil microphones as in a
loudspeaker, only reversed.
the diaphragm an induction coil moves
in a magnetic field, producing a varying
current

- relatively simple construction

- economical and rugged.
- excellent sound quality and good specifications in all areas of microphone
performance.
- can handle extremely high sound levels
- relatively unaffected by extremes of temperature or humidity
--- Dynamics are the type most widely used in general sound reinforcement.
 3 Source: Microphones
Condenser
Work on the principle of the
condenser
the diaphragm acts as one
plate of a capacitor, and the
vibrations produce changes in
the distance between the
plates
Need a battery or a phantom
power
All condensers contain additional active circuitry to allow the electrical output of
the element to be used with typical microphone inputs
- the electronics produce a small amount of noise
- there is a limit to the maximum signal level that the electronics can handle.
- adversely affected by extremes of temperature and Humidity
- condensers can readily be made with higher sensitivity
-can provide a smoother, more natural sound, particularly at high frequencies
 3 Source: Microphones
Phantom Power

Phantom power is a DC voltage (usually 12-48 volts) used to power the

electronics of a condenser Microphone and to provide the polariziing voltage
for the element tself. This voltage is supplied through the microphone cable
by a mixer equipped with phantom power or by some type of in-line
external source.

The voltage is equal on Pin 2 and Pin 3 of a typical balanced, XLR-type

connector.

Because the voltage is exactly the same on Pin 2 and Pin 3, phantom power
will have no effect on balanced dynamic microphones: no current will flow
since there is no voltage difference across the output.
 3 Source: Microphones
Condenser Vs Dynamic
Transient response refers to the
ability of a microphone to respond Condenser
to a rapidly changing sound
wave.

Here there are two studio

microphones responding to the
sound impulse produced by an
electric spark: condenser mic on
top, dynamic mic on bottom; Dynamic
it takes almost twice as long for
the dynamic microphone to It is this transient response
respond to the sound. It also difference that causes
takes longer for the dynamic to condenser mics to have a more
stop moving after the impulse has crisp, detailed sound and
passed dynamic mics to have a more
mellow, rounded sound
 3 Source: Microphones

Electret
is a type of condenser microphone, which
eliminates the need for a power supply by
using a permanently-charged material.
An electret is a stable dielectric material with
a permanently-embedded static electric
charge

Very common
High-quality
Used in lavalier, telephones and built-in mic in recording devices
 3 Source: Microphones

Frequency response diagram plots the microphone sensitivity in decibels

over a range of frequencies and generally for perfectly on-axis sound. Its
the output level or sensitivity of the microphone over its operating range
from lowest to highest frequency.

A microphone whose output is equal at all frequencies has a flat frequency

response.
Flat response microphones typically have an extended frequency range. They
reproduce a variety of sound sources without changing or coloring the
original sound.
 3 Source: Microphones

A microphone whose response has

peaks or dips in certain frequency
areas exhibits a shaped response.
A shaped response is usually designed
to enhance a sound source in a
particular application.

For instance, a microphone may have a peak in the 2 - 8

kHz range to increase intelligibility for live vocals. This shape
is called a presence peak or rise. A microphone may also be
designed to be less sensitive to certain other frequencies.
One example is reduced low frequency response (low end
roll-off) to minimize unwanted boominess or stage
rumble.
 3 Source: Microphones

Directivity expresses the sensitivity of the mike out of its axis ( its
shown by polar pattern); the sensitivity to sound relative to the
direction or angle from which the sound arrives.

The three basic directional types of microphones are

- omnidirectional
- unidirectional
- bidirectional.

Pressure Transducer Microphone

Pressure Gradient Microphone
 3 Source: Microphones
The omnidirectional microphone has equal output or sensitivity at all angles.

An omnidirectional
microphone will pick up
the maximum amount
of ambient sound.
In live sound
situations an omni
should be placed very
close to the sound
source to pick up a
useable balance
between direct sound
and ambient sound.

In addition, an omni cannot be aimed away from undesired sources such as PA

speakers which may cause feedback
 3 Source: Microphones
The unidirectional microphone is most sensitive to sound arriving from one
particular direction and is less sensitive at other directions.

The most common type is a

cardioid (heart-shaped)
response. This has the most
sensitivity at 0 degrees (on-axis)
and is least sensitive at 180
degrees (off-axis).

This mic picks up only about one-

third as much ambient sound as an
omni.
Unidirectional microphones isolate
the desired on-axis sound from both
unwanted off-axis sound and from
ambient noise.
 3 Source: Microphones
The bidirectional microphone has maximum sensitivity at both 0 degrees (front)
and at 180 degrees (back).

It has the least amount of output at 90

degree angles (sides).

The coverage or pickup angle is only about

90 degrees at both the front and the rear.

Bidirectional

Though rarely found in sound reinforcement

they are used in certain stereo techniques.
 3 Source: Microphones
Capsule Design and Directivity
Ambient sound rejection
- Since unidirectional
microphones are less
sensitive to off-axis sound
than omnidirectional types
they pick up less overall
ambient or stage sound.
Unidirectional mics should
be used to control ambient
noise pickup to get a
cleaner mix.

Distance factor- Because directional microphones pick up less ambient

sound than omnidirectional types they may be used at somewhat greater
distances from a sound source and still achieve the same balance
between the direct sound and background or ambient sound.
 3 Source: Microphones
Impedance The output impedance of a microphone is roughly equal to
the electrical resistance of its output: there are three classification,
- low <600,
- mid 600 to 6000 ,
- high>6000

The practical concern is that low impedance microphones can be used with
cable lengths of 300m or more with no loss of quality while high impedance
types exhibit noticeable high frequency loss with cable lengths greater than
about 7m .

Anyway a mic has to be connected to an input with the same or higher

impedance)
 3 Source: Microphones
Output level or sensitivity is the level of the electrical signal from the microphone
for a given input sound pressure level.

It express the electrical output in Volt generated by the microphone when placed in
a known sound pressure (SP).

Sensitivity is commonly measured in mV/Pa while the output level in one of these
methods:

1. Open-circuit voltage where 0 dBV = 1 V/0.1Pa ( Lp=74dBSPL).

2. Maximum power output where 0 dBm = 1 mW/ 1 Pa (Lp=94dBSPL)

Note: there are other less common method to express it.

 3 Source: Microphones

For example one mic can have a sensitivity of 1mV/Pa and an impedance
of 200 and we can calculate its output level for a 94dBSPL signal as:

V0
SensdBV 20 log L P 74 80dBV
1V
V02 1W
SensdBm 10 log 10 log 94 74 6 59dBm
R0 0.001W

We can refer to the ouput of a mic in terms of level, so considering the

amount of dBm or dBV calculated when a given sound pressure level is
applied to the capsule
3 Source: Microphones
 3 Source: Audio Signal Levels
Mic Level: usually 0.25mV that corresponds to -72dBV ( about -55dBm for
500 ) typical of dynamic and condenser mic

AUX/Tape: usually 100mV thats about -20dBV ( typical of CD player,

Tuner)

Line Level: usually 1V or 0dBV (typical of mixers, EQ, DSP..)

 3 Source: Audio Signal Levels

Note on dB:
dB represents the level ratio between two powers
dBV represents the level compared to 1 Volt RMS.
0dBV = 1V. There is no reference to impedance
(open circuit).
dBu represents the level compared to 0.775 Volts
RMS with an unloaded, open circuit, source (u =
unloaded).
dBm represents the power level compared to 1
mWatt. This is a level compared to 0.775 Volts RMS
across a 600 Ohm load impedance. Note that this is
a measurement of power, not a measurement of
voltage.
dB SPL - A measure of sound pressure level referd to
20Pa.
dBFS - dB Full Scale

0 dBFS represents the highest possible level in digital device.

All other measurements expressed in terms of dBFS will always be less than 0 dB
(negative numbers).
dBFS indicates the digital number with all digits ="1", the highest possible sample.
The lowest possible sample is (for instance for 16 bit audio): 0000 0000 0000 0001,
which equals -96 dBFS. Therefore the dynamic range for 16-bit systems is 96 dB.
There is no such standardized reference. x dBFS is a digital voltage level (peak) and y dBVU
or dBu is an analog voltage level (RMS).

Digital and analogue are two totally different realms.

That's why there is no relation between dBFS and dBVU or dBu, whatsoever.
Analog meter (ppm): attack time 10 to 300 ms reading rms values.
Digital meter: attack time < 1 ms reading peak values. That is really some difference.

Never take the following funny guessing game for granted. Use it only as a rough guide:
European & UK calibration for Post & Film is 18 dBFS = 0 VU
BBC spec: 18 dBFS = PPM "4" = 0 dBu
American Post: 20 dBFS = 0 VU = +4 dBu
 4. Electronic Transmission: Line level Devices
Mixing consoles

Mixing consoles are the heart of a sound reinforcement system. This brings
signals in from the outside world, allows you to manipulate them and sends
them back out to another unit.

Multiple consoles can be used for different applications in a single sound

reinforcement system.

The Front of House (FOH) mixing console must be located where the
operator can see the action on stage and hear the output of the loudspeaker
system. Some venues with permanently installed systems such as religious
facilities and theaters place the mixing console within an enclosed booth but
this approach is more common for broadcast and recording applications.

Large music productions often use a separate stage monitor mixing console
which is dedicated to creating mixes for the performers' on-stage or in-ear
monitors. These consoles are typically placed at the side of the stage so
that the operator can communicate with the performers on stage.
4. Electronic Transmission: Line level Devices

The two basic function of a mixer are

summing and routing audio signals

Bus: is the common means of mass

(data-signal) transpotation. Is the
place where the signals come
togheter
 4. Electronic Transmission: Line level Devices

The primary signal path through the mixer begins with a mic or line input and
ultimately ends up at the main outputs.
4. Electronic Transmission: Line level Devices
1. GAIN - The GAIN control is used to match the incoming signal
level to the internal operating level
2. Hi - The HI control cuts or boosts the high frequenc (12kHz).
3. MID - MID consists of two controls: the upper one adsust the
frequency (250Hz 6kHz); the lower one cuts or boosts the
selected frequency.
4. LO - The LO control operates in the same way as the HI control
except that it affects the low frequencies (60Hz).
5. AUX A - The AUX A control is assigned POST-FADER for use
with external effects units such as delays or reverbs.
6. AUX B - The AUX B control can be used when an additional or
different MIX of sound is required(such as using stage monitor
speakers).
7. PAN - Adjusts the position of the channel signal between the left or
right output channel.
9. PFL - Pressing the PFL button allows the selected channel (or
channels) to be monitored via the headphone output
10. FADER - The FADER adjusts the level from the channel to the
MAIN output. The FADER acts as a volume control for each
channel.
4. Electronic Transmission: Line level Devices
21. METERS - The meters have a PPM characteristic and
normally monitor the Left and Right stereo MIX

22. STEREO AUX RETURN - Controls the level of

signals sent to the MAIN MIX outputs from the stereo
aux input.
23. AUX A POST - Controls the level appearing atthe AUX A
POST.
24. AUX B PRE - Controls the level appearing at theAUX B PRE.

25. MAIN MIX L-R - These Faders control the level of the main
MIX to the left and rightoutputs .

27. 48V PHANTOM POWER LED - Indicates that +48V

PHANTOM POWER is available to microphone Channels.

28. RECORD LEVEL - Controls the level to the

RECORD Left/Right RCA.

29. PLAYBACK LEVEL - Controls the level from the

PLAYBACK Left/Right
4. Electronic Transmission: Line level Devices
4. Electronic Transmission: Line level Devices
4. Electronic Transmission: Line level Devices
 4. Electronic Transmission: Line level Devices
The Auto Mixer

An Auto Mixer is a good idea when there is more then a single open
microphone.

The function of an automatic mixer is twofold:

1) To automatically activate microphones as needed

2) to automatically adjust the system gain in a corresponding manner.

Auto Mixers combine the signals from multiple microphones and automatically
correct for the changing gain requirements as the NOM (Number of Open
Microphones) changes.

Threshold is a useful setting for all microphones used in a service

Unused microphones (input levelsare below threshold) are gated. When the
input of a microphone is above threshold then other inputs with a lower
assigned priority level are ducked.
 AUTOMATIC GAIN CONTROL
increases or reduces a signal
COMPRESSOR above the THRESHOLD to get
reduces the dynamics of a the desired average level
signal having a level higher (TARGET'), within the set time
than the THRESHOLD and according to the
according to the choosen compression/expansion RATIO.
RATIO.

NOISE GATE EQUALIZER

automatically reduces an audio input level Allows to filter (3 points) the frequency
when its signal is below the THRESHOLD. It response of the signal
allows to either mute a channel when no
signal is detected at its input or open it when
necessary.
The noise gate permits to count which inputs
have to be considered in use at a given
time (necessary information for automixer /
'ducking' functions).
 ON: toggles the audio input in the
correspondent bus. It is the main switch
DIGITAL TRIMMER
of the node.
adjusts the audio
A-MIX1/2: assigns the node to the
input sensitivity.
control of the automixer no.1/2, which
It can help optimize a
automatically adjusts the audio input
too low or too high
level according to the number of open
audio signal.
channels.

PRIORITY FUNCTION
This function allows to either increase ('DUCKING')
(FORCE ON) or reduce (FORCE OFF) This function allows to get
an input level forcedly. automatic level attenuation
Its activation can be either automatic of one or more audio inputs
(when the audio input signal level is when a signal is detected
above the noise gate threshold) or (through the noise gate) at
through an external command linked to the input with the highest
a logic input (GPIN). priority level. The priority
level, from 0 (lowest) to 5
(highest).
 EQUALIZER
Allows to filter (5 points) the frequency
response of the output signal

AUTOMATIC LEVEL
CONTROL
adjusts the level of one or
more audio outputs according
to the detected ambient
noise;
LIMITER
limits the dynamics of a signal having
a level higher than the THRESHOLD
with high compression ratio. It can be
really useful to avoid signal distortion DELAY: applies an
due to too high levels. electronic delay time to the
output signal.
4. Electronic Transmission: Line level Devices
Signal Processors

Signal processors fall into three main categories based

on which property of the audio signal they affect:

1) equalizers affect frequency response

2) dynamics controllers affect amplitude

3) delays affect time properties such as phase.

 4. Electronic Transmission: Line level Devices

Equalizers:

Equalizers exist in sound reinforcement systems in two forms: graphic and

parametric.
 4. Electronic Transmission: Line level Devices
Graphic equalizers have faders (slide controls) which together resemble a
frequency response curve plotted on a graph. Sound reinforcement systems
typically use graphic equalizers with one-third octave frequency centers. These
are typically used to equalize output signals going to the main loudspeaker
system or the monitors on stage.
 4. Electronic Transmission: Line level Devices

Central frequencies of a 31 band Graphic EQ

20 25 31,5 40 50 63 80 100 125 160 200
250 315 400 500 600 800 1k 1,25k 1,6k 2k 2,5k
3,15k 4k 5k 6,3k 8k 10k 12,5k 16k 20k

Bandwidth Frequencies: 3 dB points = half

power points labeled as fH and fL
Center Frequency: fc f h fl (geometric
mean)
Bandwidth = reciprocal of Q (Selectivity Factor)

fC fC
Q
f H f L BW
 4. Electronic Transmission: Line level Devices

Parametric equalizer has the possibility to select the frenquncy that has to be
modifiyd, the width of the frenquency range thats affected by this variation,
and the amplitude of the variation. The narrow the filtering effect the most the
phase change we introduce.

A feedback suppressor is an automatically-adjusted band-reject or notch filter

which includes a microprocessor to detect the onset of feedback and direct the
filter to suppress the feedback by lowering the gain right at the offending
frequency.loloolp
4. Electronic Transmission: Line level Devices
Parametric EQ:
Freedom to chose
- Center Frequency
- Level
- Q ( band at -3dB)
 4. Electronic Transmission: Line level Devices
Electronic Crossover (Spectral Divider):

A crossover is a filter. It is used to block some

frequencies while allowing others to pass with little or
no effect. This allows different speakers to work
together minimizing the overlap.
It has one or more full-range inputs (20 Hz - 20.000 Hz) and two
or more filtered outputs.
low-pass filter is a filter that passes low-frequency signals but
attenuates (reduces the amplitude of) signals with frequencies
higher than the cutoff frequency
hi-pass filter is a filter that passes hi-frequency signals but
attenuates (reduces the amplitude of) signals with frequencies
lower than the cutoff frequency
band-pass filter is a device that passes frequencies within a
certain range and rejects (attenuates) frequencies outside that
range .
4. Electronic Transmission: Line level Devices
 4. Electronic Transmission: Line level Devices

The crossover point is the intersection between the low-hi pass filter,
A Filter does not stop completly the frequencies after or before the
crossover point, but attenuates of a certain ammount depending on the
frequency.
In fact such devices are featured by a attenuation slope measured in
dB/octave
Usually the slope is a multiple of 6 and it defines the order of
crossover ( or filter).
Most of the time electronic crossovers allow to select the family:
frequenti depending on the designer: Butterworth, Linkwitz, Bessel.
4. Electronic Transmission: Line level Devices
 4. Electronic Transmission: Line level Devices

Any filtering action introduce a phase shift of the signal, that has to be
considered
 4. Electronic Transmission: Line level Devices
Dynamic Controllers
Dynamic controllers or processors represent a class of signal processing
devices used to alter an audio sign based solely upon its frequency content
and amplitude level, thus the term dynamic since the processing is
completely program dependent.

Compressors
are designed to manage the dynamic range of an audio signal. A
compressor accomplishes this by reducing the gain of a signal that is above
a defined level (threshold) by a defined amount (ratio).
 4. Electronic Transmission: Line level Devices
Most compressors available are designed to allow the operator to select a
ratio within a range typically between 1:1 and 20:1, with some allowing
settings of up to :1.
A compressor with an infinite ratio is typically referred to as a limiter.
The speed that the compressor adjusts the gain of the signal (called the
attack) is typically adjustable as is the final output of the device.
 4. Electronic Transmission: Line level Devices
Expanders
The expander is a compressor in reverse. There are two types of expander:
1) signals above the threshold remain at unity gain whereas signals below
the threshold are reduced in gain,
2) the signal above the threshold also has the gain increased.

An expander works like an automatic mixing engineer who pulls down

the signal when the signal falls below the threshold; the more it falls
below the threshold the more he pulls down the fader.

Expanders are most often used in Studio recording to provide the

best mix signal to noise ratio when producing final masters.
 4. Electronic Transmission: Line level Devices

Noise gates
sets a threshold where if it is quieter it will not let the signal pass and
if it is louder it opens the gate. A noise gate's function is in a sense the
opposite to that of a compressor. Noise gates are useful for microphones
which will pick up noise which is not relevant to the program, such as the
hum of a miked electric guitar amplifier or the rustling of papers on a
minister's podium.
4. Electronic Transmission: Line level Devices
 4. Electronic Transmission: Line level Devices
Delay
Since the sound travels through the air at 344 m/s there will be
delay of 2.9 ms for each travelled m. After 18m delay amounts to
53 ms.

A 53-ms delay is long enough for most persons to perceive the

combination as having an echo that would be annoying and tend
to be fatiguing. and have degraded intelligibility and sound
quality unless one of the signals is more than 10 dB larger.
 4. Electronic Transmission: Line level Devices

Delay is a device that introduce a shift in the time domain of the audio
signal . It has the job of adjusting the relative phase of signal, it acts as an
EQ in the time domain. Its mainly used for two pouposes:

-Time allignement of the sound coming from two different sources aimed to
the same area. In order to avoid echos and comb filtering;
- Used to create the correct directional image of sound, coming from the
talker and not from the loudspeaker.
 4. Electronic Transmission: Line level Devices
Key Specifications for line level active electronic devices:
Active balanced high-impedance input: 10k
Active balanced low-impedance output: 150
Frequency Response range: 8Hz-22kHz
Amplitude Response: 0.5dB, 20Hz-20kHz
Phase Response: <45 from 20Hz to 20kHz
Hun and Noise <-90dBV
Dynamic Range>100dB
THD<0.1%
 4. Electronic Transmission: Line level Devices-
Interconnections
Line level devices are usually active and transfer a limited ( even
negligible) amounts of power. The signal transfer is reduced to
voltage only, hence the term Voltage Source describes the transmission
system between line level devices. These sources transfer signal virtually
without loss as long as the driving impedance is very low compared to the
receiver.
Input Cable Output _ impedance
LineLoss ( dB ) 20 log
Input _ impedance

In Line level devices the input stage is usually balanced in order to reject
noise thats injected onto the cable cable.

Balanced line refers to the wiring configuration that uses two signal
conductor and one common, The two signal are fed into a differential
input stage which amplifies only the signal that are unmatched and are
sent out from a push-pull output stage that splits the signal in two
identical ones one of which is reverse polarity.
 4. Electronic Transmission: Line level Devices-
Interconnections

Active Balanced: Inputs and outputs

are tied together directly

Transformer Balanced: a balanced

transformer can be sobstituted for
the active input with the advantages
in isolation between the systems.
 4. Electronic Transmission: Line level Devices-
Interconnections

Unbalanced Connection: When an

unbalanced output drives a
balanced input the differential
inputs are fed by the signal and
common respectively. This allows
for the common mode rejection to
suppress any interference that is
injected into the line. The succes
of this configuration depends upon
the isolation between the grounds
of the two devices.

Note: Unbalanced interconnections are one of the most

common sources of polarity reversal. Check that the
connection is made across the non-inverting source and
receiver terminals.
 4. Electronic Transmission: Speaker Level Devices

Speaker level transmission moves us into the realm of real power.

The speaker level voltage runs higher than line by a factor of 10:1 (
reaching 100V rms). Current levels can easily reach ratios of 250:1
respect to line level.

Now we are dealing with power transfers of 1kW magnitude.

Amplifier is the source of current and voltage and the speaker (RL)
and cabling (Rw) the resistance. Losses in the cables become
important.

RL
LineLoss ( dB ) 20 log
RL 2 Rw
Line loss for different loads
and different cable sections
4. Electronic Transmission: Speaker Level Devices

Line Loss and Wire Section

The load is considered to be
concentrated at the end of
the line.
In order to keep the loss within
1dB a certain cable section must
be provided.

If the speakers are distributed

along the line, the section can be
almost halved.
 4. Electronic Transmission: Speaker Level Devices
How to connect speakers when:
lot of speakers connected,
long speaker lines,

Constrains of a traditional low impedance connection

+
- Series: little power
+ delivered to speakers + + Series/Parallel:
+ - -
- 16 - + reasonable
+
-
- + + impedance but
+
- - complex
-

1
+ + + + + Parallel: Hi
- - - - - currents!!!

Heavy loss due to wiring resistance

 Drawbacks: in real live a speaker is not a linear DC
resistor. It has a freq varing impedance. When we are
+
- close to the resonance the speaker becomes a
current generator thats opposite to the one
+
+ generated by the amplifier.
- 16 -
+ In a series of speakers the first introduce a distortion
- on the circuit that alter the current circulating on the
+ second and so on. In parallel the current distortion of

-
the first will not affect the other ones.

The demands for hi

1
+ + + + + currents can cause a
- - - - - voltage sag and results
in SPL loss.
 4. Electronic Transmission: Speaker Level Devices
100V Line connection.
100V
Thanks to a step-up transformer
and step-down transformers
connected to speakers the
problem related to impedance
GND matching is solved
High transformer impedance and
high line voltage minimize the wiring
power loss
1 2
Z Line VLine 100 2

2 2
Z Sp
VLine

100 1
# Spea ker s
WLine
WSp WSp Z Sp Z Line Z Line
Failure of a speaker do not affect the line and the other speakers
 5. Acoustic Transmission

r1
SPL(r ) 20 log
r0
 5. Acoustic Transmission
Reverberation

DRY: Many factors influence speech intelligibility

RT 0.8 sec

RT 1.3 sec

RT 2.0 sec
 5. Acoustic Transmission
Frequency Response - The measured, or specified, output over a specified
range of frequencies for a constant input level varied across those frequencies.
It often includes a variance limit such as within "+/- 3 dB".
Maximum SPL - The highest output the loudspeaker can manage, short of
damage or not exceeding a particular distortion level. This rating is often
inflated by manufacturers and is commonly given without reference to
frequency range or distortion level.
Coverage Angle - Represents the effevtive behaviour of the speaker during
operation, indicationg how the sound is dispersed within the space.
 5. Acoustic Transmission

Rated Power - Nominal (or even continuous) and peak (or maximum short-
term) are the powers a loudspeaker can handle (i.e., maximum input power
before destroying the loudspeaker). It is never the sound output the
loudspeaker produces.
Sensitivity -The sound pressure level produced by a loudspeaker in a non-
reverberant environment, usually specified in dB, and measured at 1 meter with
an input of 1 watt or 2.83 volts, typically at one or more specified frequencies.
 6. Acoustic Gain

The aim of a sound reinforcement system is to enhance the sound of the

sources and guarantee the expected listening level .

Sound reinforcement system has to be designed in order to move the

talker closer to the listener and overcome the background noise.

NAG 20 log D0 20 log EAD

 6. Acoustic Gain

Possible Gain for a simple outdoor sound reinforcement system (

free field condition)

SPL0L SPL0 20 log D0

*3 SPLSR
L SPLSR
20 log D2

*1 SPL0S SPL0 20 log Ds

*2 SPLSR
S SPLSR
20 log D1
 6. Acoustic Gain

SR
Feedback Condition SPL SPLS
0
S

From *1 and *2 we get

SPL0 20 log Ds 20 log D1 SPLSR
Putting into *3
SPLSR
L SPL0
20 log Ds 20 log D1 20 log D2
Defining the Possible Acoustig Gain: PAG SPLSR
L SPL0
L

PAG 20 log D0 20 log Ds 20 log D1 20 log D2

PAG 20 log D0 20 log Ds 20 log D1 20 log D2 FSM 10 log NOM

 6. Acoustic Gain
PAGDirect PAGOmni ( LM T L L LL LL )
M l M
S S S
Microphone to
talker angle
Loudspeaker to
Microphone to
Loudspeaker to Microphone angle
Loudspeaker angle
listener angle
 6. Acoustic Gain

NAG 20 log D0 20 log EAD

The system can work properly if :

PAG NAG
This, due to the ipothesys,
does not ensure that the
system will work,
but if we do not achieve, at
least, in this condition the
system cannot work
 6. Acoustic Gain

NAG 20 log D0 20 log EAD

QMe 4N
L x 10 log Q= directivity factor
4D x MaS a
2
Me= electro-acoustic Modifier
( change the interaction
between the speaker and the
Modified Hopkins-Striker for microphone but not affecting
semireverberant (Ltot=LDir+Lrev) the Lrev)
Ma= architectural modifier (
takes care of the ratio
between the total absorbtion
coefficient and the absorbtion
coefficient of the wall
affected by the first
reflections)
N= ratio between the number
of radiating devices producing
LDir and the total number of
sources
 6. Acoustic Gain
Possible Gain for a simple indoor sound reinforcement system (
semi reverberant condition)

QMe 4N
x g ( x) Lx 10 log

SPLx SPLx0 20 log 10 log 4D 2
x MaS a

x0 g ( x0 )

g ( x) Dc2 x 2 Dc
QS a
16N
D0 D1 g ( D0 ) g ( D1 )
PAG 20 log 10 log 10 log NOM FSM
s 2
D D g ( Ds ) g ( D2
)
g(x) tells us that
g ( D0 ) after the Dc
D
NAG 20 log 0 10 log there is no more
EAD g ( EAD ) attenuation with
distance
 6. Acoustic Gain: Headroom

Crest Factor: represent the ratio x peak

between the peak value and the
C
xrms
RMS.
 6. Acoustic Gain: Headroom

The lowest possible is 0dB for DC

signals. Sine wave has 3dB.
Typical for speech and rock music
is 9-12dB.
Typical for Classical Music is 20-
25dB Great Big Sea - Old Brown's Daughter- uncompressed

x peak Igor Stravinsky - Rite Of Spring - uncompressed

C
xrms

Typical MP3 pop music 128kbps 44 kHz

 Power Required and Reccomended Amplifier size:

Starting from the consideration of the system has to provide a certain

dBSPL at the last listener in the area we can say that the power required
for a given loudspeaker is

dBW = Lp - Lsens + 20 * Log (D2/Dref) + HR

W = 10 ^ (dBW / 10)
Where:

Lreq = required SPL at listener

Lsens = loudspeaker sensitivity (1W/1M)
D2 = loudspeaker-to-listener distance
Dref = reference distance
HR = desired amplifier headroom/crest factor
dBW = ratio of power referenced to 1 watt
W = power required
 6. Acoustic Gain: Electrical Power Requirement
Power Required and Reccomended Amplifier size

Rule of Thumb
For a rule of thumb the best answer is found using what is commonly called
the loudspeakers RMS (root mean square) power rating. Use an amplifier
that is twice the RMS rating ( to take into account the crest factor of your
signal). If you cant find an amplifier with that exact rating, multiply this
power rating by 0.8 and also by 1.25 to find a range of acceptable power.

Note: For 100/70.7V lines Wtot iN1Wi

Where Wi is set by the tapping of the transformer

 End of Part 1

THANKS for your kind attention!