UNESCO-NIGERIA TECHNICAL &
VOCATIONAL EDUCATION
REVITALISATION PROJECT-PHASE II
NATIONAL DIPLOMA IN
ELECTRICAL ENGINEERING TECHNOLOGY
TELECOMMUNICATION
ENGINEERIN (I)
COURSE CODE: EEC 128
YEAR I- SEMESTER II
THEORY
Version 1: December 2008
1
TABLE OF CONTENTS
Week1: Introduction to communication systems …………………….
1
Week2: Elements of a communication systems……………………………
6
Week3: Types of transducers……………………………………………….
11
Week4: Types of microphones ………………………………………………..
14
Week5: Loudspeakers………………………………………………………………………
21
Week6: Amplitude modulation……………………………………….
28
Week7: Frequency Modulation…………………………………………..
32
Week8: Amplitude modulator…………………………………………….
37
Week9: Frequency Modulators…………………………………………………………………..
40
Week10: Amplitude Demodulators………………………………………
42
Week11: Frequency demodulator………………………………………
43
Week12: Tuned radio receiver………………………………………
46
Week13: Superherodyne Receiver
.
……………………………………………
52
Week14: AM Superherodyne Receiver ……………………………………………...
58
Week15: FM Receiver………………………………………………….
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3 Amp 1. Principles of Communication
Week 1
1.1 Introduction to communication systems
Communication is the transmission of information /message from one point to
another. Communication enters our lives in many ways:
- Telephone – makes us talk to any person anywhere.
- Radio and television – entertain and educate.
- Communication signals as navigational aids – ships, aircrafts and satellites.
- Weather forecasting – conditions measured by a multitude of sensors are
communicated to forecasters.
- Videophones, voicemail and satellite conferencing – enable seeing live images
instantly and communicate directly with people located far away.
- Digital data transmission and retrieval – has made realization of e-mail, FAX and
internet possible.
We communicate through speech. In modern communication systems, the
information is first converted into electrical signals and then sent electronically.
Communication system
Two persons talking to each other constitute the simplest communication system. The
person who speaks is the source, the person listening is the receiver and the intervening
air is the communication link between them. A communication system consists of three
basic components:
- Transmitter (source)
- Communication channel (link – medium)
- Receiver
Nature/details of these components depend on:
1. Nature of the signal/ message to be communicated.
2. Distance which separates the source and the receiver.
Direct talking is possible over short distances – sound waves attenuate fast. Long
distance communication requires the signal/message to be converted into an electrical
signal/a set of signals /electromagnetic waves.
- Long distance communication requires a link between the source and the
receiver
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Figure1.1:
Week 1
Communication system
Communication Channel:
Provides a link between the transmitter and the receiver. It can be a transmission line
(telephone and telegraphy), an optical fibre (optical communication) or free space in
which the signal is radiated in the form of electromagnetic waves.
Designing a Communication System
In designing a communication system we have to focus our attention to the following
questions:
- In what form is the information that is to be conveyed.
- How can the transmitter use this information?
- How does the transmitter feed the information to the communication
channel?
- What effects does the communication channel have on the information?
- In what form the receiver should present the information to the outside world?
- How does the received information differ from the original information?
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What cause the difference? To what extent can the two be allowed to
Figure1.2:
Transmitter
Receiver
In its simplest form, the transmitter has following problems:
1. Size of the antenna or aerial
For transmitting a signal we need an antenna. It should have a size comparable
to the wavelength of the electromagnetic wave representing the signal ( at least
/4) so that the time variation of the signal is properly sensed by the antenna.
For an electromagnetic wave of frequency 20 kHz, the wavelength  is 15 km.
Obviously such a long antenna is not possible. Therefore, direct transmission of
such a signal is not possible. If the frequency of the signal is 1MHz, the
corresponding wavelength is 300m and transmission of such a signal is
possible. Therefore, there is a need of translating the information contained in
the original low frequency signal into high or radio-frequencies before
transmission.
2. Effective power radiated by an antenna
The power radiated from a linear antenna  l /  2 For a good transmission we need
high power hence there is need for high frequency transmission.
3. Mixing up of signals from different transmitters Direct transmission of baseband signal
leads to interference from multiple transmitters. Thus multiple user friendly
communication is not possible. A possible solution is provided by employing
communication at high frequencies and then allotting a band of frequencies to each user.
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The above arguments suggest that there is a need for translating the original signal ( low
frequency) into a high frequency wave before transmission such that the translated signal
continues to possess the information contained in the original signal. The high frequency
wave carrying the information is called the carrier wave. The process of transformation
is called Modulation. Modulation Transformation of the signal into a form suitable for
transmission through a given communication channel
Figure2:
Transmitter
Receiver
Transmitter:
Transmits the message/signal over the communication channel. Quite often the original
signal is not suitable for transmission over the communication channel to the receiver. It
requires to be modified to a form suitable for transmission. A transmitter, in its simplest
form, is a setup which boosts the power of message signal and feeds it into the
communication channel
Antenna
An antenna is a vital component of any communication system. It is employed both at the
transmitting end as well as at the receiving end. An antenna is a length of conductor, its
length is such that it acts as a resonant circuit at the frequency of operation. l = /2.
It acts as a conversion device. The first conversion takes place at the transmitter where
electrical energy is converted into electromagnetic waves. The second conversion occurs
at the receiving end where the electromagnetic waves are transformed into electrical
signal that is applied to the input of the receiver.
Two types of antenna:
1. Dipole antenna – Length of dipole = /2 ; Omni directional.
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2. Dish antenna – A spherical or parabolic dish is employed as a reflector
or collector. The resonant element is placed at the
focus. It is highly directional.
Communication Channel
In a communication system, the communication channel or the transmission medium is
the physical path between the transmitter and receiver. Transmission media can be
classified into two broad categories:
(a) Guided - Point – to – point communication
(i) Twisted pair
(ii) Coaxial cable
(iii) Optical fibre
(b) Unguided – Free space
Characteristics and quality of transmission are determined both by the nature of the signal
as well as the medium. In guided media, the nature of the medium is more important; in
unguided media, the spectrum or the frequency band of the signal transmitted by the
transmitter is more important. Characteristics of a Communication Channel : Band width,
Modulation and Data rate.
Receiver
Reconstructs the original message or data after its propagation through
the communication channel, the process consisting of decoupling of the carrier wave
and the modulating signal is broadly termed as demodulation. The design of the receiver
depends on the modulation process employed in the transmitter. The antenna receives the
modulated wave transmitted from the transmitter, which is then amplified by a suitable
amplifier and fed to the demodulator or decoder. The demodulator or decoder extracts the
original signal. The process of demodulation provides a means of recovering the original
signal from the modulated wave. In effect, demodulation is reverse of modulation:
therefore, it depends on the modulation process used.
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1.2 Elements of communication systems
Communication systems consist of:
1. Transmitter
- Convert the original signal to be suitable for transmission.
2. Receiver
- Accepts the transmitted signals and convert back to original form.
3. Transmission Medium (Channel)
- Provide means of transporting signals from Transmitter to Receiver such as
copper wires, fiber optic or free space.
What is baseband?
Without any shift in the range of frequencies of the signal.The signal is in its original
form, not changed by modulation.
WHAT IS CARRIER?
Transferring information at high frequency
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Figure1.3 Carrier modulated waveforms
WHAT IS MODULATION ?
MODULATION IS THE PROCESS OF CHANGING SOME PROPERTYOF THE
INFORMATION SOURCES INTO SUITABLE FORM FOR TRANSMISSION
THROUGH THE PHISICAL MEDIUM/CHANNEL. It is performed in the Transmitter
by a device called Modulator.
WHAT IS DEMODULATION ?
DEMODULATION IS THE REVERSE PROCESS OF MODULATION BY
CONVERTING THE MODULATED INFORMATION SOURCES BACK TO ITS
ORIGINAL INFORMATION (IT REMOVES THE INFORMATION FROM THE
CARRIER SIGNAL). It is performed in the Receiver by a device called
Demodulator.
THE NEED OF MODULATION:
¢ Channel assignment (various information sources are not always suitable for direct
transmission over a given channel)
¢ Reduce noise &interference
¢ Overcome equipment limitation
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TYPE OF MODULATION:
¢
Amplitude Modulation (AM)
¢
Frequency Modulation (FM)
¢
Phase Modulation (PM)
ANALOG AND DIGITAL SIGNAL
The information can be in term of :
 Analog form such as Human Voice or Music
 Digital form such as binary-coded number.
There are 2 basic type of communication :
 Analog Communication
 Digital Communication
Example of Analog signal is shown below:
 Analog comes in term of Sinusoid (Sine or Cosine wave)
 Analog signals are continuous electrical signals that vary in
amplitude and frequency .
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Example of Digital Signal is shown below:
WHAT IS BANDWIDTH ?
IT IS THE DIFFERENCE BETWEEN THE HIGHEST FREQUENCIES AND THE
LOWEST FREQUENCIES OF THE INPUT SIGNAL FREQUENCIES (fB = 2fm ). The
bandwidth of a communication signal  bandwidth of the information signal.
EXAMPLE 3:
If human voice frequencies contain signals between 300 Hz and 3000 Hz, a voice
frequency channel should have bandwidth equal or greater than 2700 Hz.a communication
channel cannot propagate a signal that contains a frequency that is changing at a rate greater than
the Channel Bandwidth.
PROPAGATION TECHNIQUES
A signal can be propagated in 3 ways:
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1. Ground-Wave Propagation
Frequency < 2 MHz
2. Sky-Wave Propagation
Frequency between 2 MHz and 30 MHz
3. Line-of-Sight Propagation
Frequency > 30 MHz
A propagation of radio frequencies are shown below:
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Week 3
2.1 Types of transducers
A transducer is a device, usually electrical, electronic, electro-mechanical,
electromagnetic, photonic, or photovoltaic that converts one type of energy or physical
attribute to another for various purposes including measurement or information transfer
(for example, pressure sensors).
The term transducer is commonly used in two senses; the sensor, used to detect a
parameter in one form and report it in another (usually an electrical or digital signal), and
the audio loudspeaker, which converts electrical voltage variations representing music or
speech, to mechanical cone vibration and hence vibrates air molecules creating sound.
Antenna
An antenna is a transducer designed to transmit or receive electromagnetic waves. In
other words, antennas convert electromagnetic waves into electrical currents and vice
versa. Antennas are used in systems such as radio and television broadcasting, point-topoint radio communication, wireless LAN, radar, and space exploration. Antennas
usually work in air or outer space, but can also be operated under water or even through
soil and rock at certain frequencies for short distances. Physically, an antenna is an
arrangement of conductors that generate a radiating electromagnetic field in response to
an applied alternating voltage and the associated alternating electric current, or can be
placed in an electromagnetic field so that the field will induce an alternating current in
the antenna and a voltage between its terminals. Some antenna devices (parabolic
antenna, Horn Antenna) just adapt the free space to another type of antenna.
Cathode ray tube
The cathode ray tube (CRT) is a vacuum tube containing an electron gun (a source of
electrons) and a fluorescent screen, with internal or external means to accelerate and
deflect the electron beam, used to form images in the form of light emitted from the
fluorescent screen. The image may represent electrical waveforms (oscilloscope),
pictures (television, computer monitor), radar targets and others. The single electron
beam can be processed in such a way as to display moving pictures in natural colors. The
CRT uses an evacuated glass envelope which is large, deep, heavy, and relatively fragile.
Display technologies without these disadvantages, such as flat plasma screens, liquid
crystal displays, DLP, OLED displays have replaced CRTs in many applications and are
becoming increasingly common as costs decline.
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Figure2.1: Cathode ray tube
Cutaway rendering of a color CRT: 1. Electron guns 2. Electron beams 3. Focusing coils
4. Deflection coils 5. Anode connection 6. Mask for separating beams for red, green, and
blue part of displayed image 7. Phosphor layer with red, green, and blue zones 8. Closeup of the phosphor-coated inner side of the screen
Galvanometer
A galvanometer is a type of ammeter; an instrument for detecting and measuring electric
current. It is an analog electromechanical transducer that produces a rotary deflection,
through a limited arc, in response to electric current flowing through its coil. The term
has been expanded to include uses of the same mechanism in recording, positioning, and
servomechanism equipment.
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Figure2.2: D'Arsonval galvanometer movement.
Loudspeaker
A loudspeaker, speaker, or speaker system is an electro acoustical transducer that
converts an electrical signal to sound. The term loudspeaker can refer to individual
transducers (known as drivers), or to complete systems consisting of a enclosure
incorporating one or more drivers and electrical filter components. Loudspeakers, just as
with other electro acoustic transducers, are the most variable elements in an audio system
and are responsible for the greatest degree of audible differences between sound systems.
Microphones
A microphone, sometimes referred to as a mic or mike ( pronounced /ˈ maɪk/), is an
acoustic-to-electric transducer or sensor that converts sound into an electrical signal.
Microphones are used in many applications such as telephones, tape recorders, hearing
aids, motion picture production, live and recorded audio engineering, in radio and
television broadcasting and in computers for recording voice, VoIP, and for non-acoustic
purposes such as ultrasonic checking.
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2.2 Types of microphones
A microphone, sometimes referred to as a mic or mike ( pronounced /ˈ maɪk/), is an
acoustic-to-electric transducer or sensor that converts sound into an electrical signal.
Microphones are used in many applications such as telephones, tape recorders, hearing
aids, motion picture production, live and recorded audio engineering, in radio and
television broadcasting and in computers for recording voice, VoIP, and for non-acoustic
purposes such as ultrasonic checking.
Figure2.3: A Neumann U87 condenser microphone
The most common design today uses a thin membrane which vibrates in response to
sound pressure. This movement is subsequently translated into an electrical signal. Most
microphones in use today for audio use electromagnetic induction (dynamic
microphones), capacitance change (condenser microphones) or piezoelectric generation
to produce the signal from mechanical vibration.
Condenser, capacitor or electrostatic microphones
In a condenser microphone , also known as a capacitor microphone, the diaphragm acts
as one plate of a capacitor, and the vibrations produce changes in the distance between
the plates. There are two methods of extracting an audio output from the transducer thus
formed: DC-biased and RF (or HF) condenser microphones. With a DC-biased
microphone, the plates are biased with a fixed charge (Q). The voltage maintained across
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Figure2.4: Inside the Oktava 319 condenser microphone.
the capacitor plates changes with the vibrations in the air, according to the capacitance
equation (C = Q / V), where Q = charge in coulombs, C = capacitance in farads and V =
potential difference in volts. The capacitance of the plates is inversely proportional to the
distance between them for a parallel-plate capacitor. (See capacitance for details.) A
nearly constant charge is maintained on the capacitor. As the capacitance changes, the
charge across the capacitor does change very slightly, but at audible frequencies it is
sensibly constant. The capacitance of the capsule and the value of the bias resistor form a
filter which is high pass for the audio signal, and low pass for the bias voltage. Note that
the time constant of an RC circuit equals the product of the resistance and capacitance.
Within the time-frame of the capacitance change (on the order of 100 μs), the charge thus
appears practically constant and the voltage across the capacitor changes instantaneously
to reflect the change in capacitance. The voltage across the capacitor varies above and
below the bias voltage. The voltage difference between the bias and the capacitor is seen
across the series resistor. The voltage across the resistor is amplified for performance or
recording. RF condenser microphones use a comparatively low RF voltage, generated by
a low-noise oscillator. The oscillator may either be frequency modulated by the
capacitance changes produced by the sound waves moving the capsule diaphragm, or the
capsule may be part of a resonant circuit that modulates the amplitude of the fixedfrequency oscillator signal. Demodulation yields a low-noise audio frequency signal with
a very low source impedance. This technique permits the use of a diaphragm with looser
tension, which may be used to achieve better low-frequency response. Condenser
microphones span the range from inexpensive karaoke mics to high-fidelity recording
mics. They generally produce a high-quality audio signal and are now the popular choice
in laboratory and studio recording applications. They require a power source, provided
either from microphone inputs as phantom power or from a small battery. Power is
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necessary for establishing the capacitor plate voltage, and is also needed for internal
amplification of the signal to a useful output level. Condenser microphones are also
available with two diaphragms, the signals from which can be electrically connected such
as to provide a range of polar patterns (see below), such as cardioid, omnidirectional and
figure-eight. It is also possible to vary the pattern smoothly with some microphones, for
example the Røde NT2000 or CAD M179.
Electret condenser microphones
An electret microphone is a relatively new type of capacitor microphone invented at Bell
laboratories in 1962 by Gerhard Sessler and Jim West[1]. The externally-applied charge
described above under condenser microphones is replaced by a permanent charge in an
electret material. An electret is a ferroelectric material that has been permanently
electrically charged or polarized. The name comes from electrostatic and magnet; a static
charge is embedded in an electret by alignment of the static charges in the material, much
the way a magnet is made by aligning the magnetic domains in a piece of iron. They are
used in many applications, from high-quality recording and lavalier use to built-in
microphones in small sound recording devices and telephones. Though electret
microphones were once low-cost and considered low quality, the best ones can now rival
capacitor microphones in every respect and can even offer the long-term stability and
ultra-flat response needed for a measuring microphone. Unlike other capacitor
microphones, they require no polarizing voltage, but normally contain an integrated
preamplifier which does require power (often incorrectly called polarizing power or bias).
This preamp is frequently phantom powered in sound reinforcement and studio
applications. While few electret microphones rival the best DC-polarized units in terms
of noise level, this is not due to any inherent limitation of the electret. Rather, mass
production techniques needed to produce electrets cheaply don't lend themselves to the
precision needed to produce the highest quality microphones.
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Figure2.5: Electret condenser microphone capsules
Dynamic microphones
Dynamic microphones work via electromagnetic induction. They are robust, relatively inexpensive
and resistant to moisture. This, coupled with their high gain before feedback makes them ideal for onstage use. Moving coil microphones use the same dynamic principle as in a loudspeaker, only reversed.
A small movable induction coil, positioned in the magnetic field of a permanent magnet, is
attached to the diaphragm. When sound enters through the windscreen of the microphone, the sound
wave moves the diaphragm. When the diaphragm vibrates, the coil moves in the magnetic field,
producing a varying current in the coil through electromagnetic induction. A single dynamic
membrane will not respond linearly to all audio frequencies. Some microphones for this reason utilize
multiple membranes for the different parts of the audio spectrum and then combine the resulting signals.
Combining the multiple signals correctly is difficult and designs that do this are rare and tend to be
expensive. There are on the other hand several designs that are more specifically aimed towards isolated
parts of the audio spectrum. The AKG D 112, for example, is designed for bass response rather than
treble[2]. In audio engineering several kinds of microphones are often used at the same time to get the best
result. Ribbon microphones use a thin, usually corrugated metal ribbon suspended in a magnetic field.
The ribbon is electrically connected to the microphone's output, and its vibration within the magnetic field
generates the electrical signal. Ribbon microphones are similar to moving coil microphones in the sense
that both produce sound by means of magnetic induction. Basic ribbon microphones detect sound in a
bidirectional (also called figure-eight) pattern because the ribbon, which is open to sound both front and
back, responds to the pressure gradient rather than the sound pressure. Though the symmetrical
front and rear pickup can be a nuisance in normal stereo recording, the high side rejection
can be used to advantage by positioning a ribbon microphone horizontally, for example
above cymbals, so that the rear lobe picks up only sound from the cymbals. Crossed
figure 8, or Blumlein stereo recording is gaining in popularity, and the figure 8 response
of a ribbon microphone is ideal for that application.
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Other directional patterns are produced by enclosing one side of the ribbon in an acoustic
trap or baffle, allowing sound to reach only one side. Older ribbon microphones, some of
which still give very high quality sound reproduction, were once valued for this reason,
but a good low-frequency response could only be obtained if the ribbon is suspended very
loosely, and this made them fragile. Modern ribbon materials, including new
nanomaterials[3] have now been introduced that eliminate those concerns, and even
improve the effective dynamic range of ribbon microphones at low frequencies.
Protective wind screens can reduce the danger of damaging a vintage ribbon, and also
reduce plosive artifacts in the recording. Properly designed wind screens produce
negligible treble attenuation. In common with other classes of dynamic microphone,
ribbon microphones don't require phantom power; in fact, this voltage can damage some
older ribbon microphones. (There are some new modern ribbon microphone designs
which incorporate a preamplifier and therefore do require phantom power, also there are
new ribbon materials available that are immune to wind blasts and phantom power.)
Figure2.5: US664A University Sound Dynamic Supercardioid Microphone
Carbon microphones
A carbon microphone, formerly used in telephone handsets, is a capsule containing
carbon granules pressed between two metal plates. A voltage is applied across the metal
plates, causing a small current to flow through the carbon. One of the plates, the
diaphragm, vibrates in sympathy with incident sound waves, applying a varying pressure
to the carbon. The changing pressure deforms the granules, causing the contact area
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between each pair of adjacent granules to change, and this causes the electrical resistance
of the mass of granules to change. The changes in resistance cause a corresponding
change in the voltage across the two plates, and hence in the current flowing through the
microphone, producing the electrical signal. Carbon microphones were once commonly
used in telephones; they have extremely low-quality sound reproduction and a very
limited frequency response range, but are very robust devices. Unlike other microphone
types, the carbon microphone can also be used as a type of amplifier, using a small
amount of sound energy to produce a larger amount of electrical energy. Carbon
microphones found use as early telephone repeaters, making long distance phone calls
possible in the era before vacuum tubes. These repeaters worked by mechanically
coupling a magnetic telephone receiver to a carbon microphone: the faint signal from the
receiver was transferred to the microphone, with a resulting stronger electrical signal to
send down the line. (One illustration of this amplifier effect was the oscillation caused by
feedback, resulting in an audible squeal from the old "candlestick" telephone if its
earphone was placed near the carbon microphone.
Piezoelectric microphones
A crystal microphone uses the phenomenon of piezoelectricity—the ability of some
materials to produce a voltage when subjected to pressure—to convert vibrations into an
electrical signal. An example of this is Rochelle salt (potassium sodium tartrate), which is
a piezoelectric crystal that works as a transducer, both as a microphone and as a slimline
loudspeaker component. Crystal microphones were once commonly supplied with
vacuum tube (valve) equipment, such as domestic tape recorders. Their high output
impedance matched the high input impedance (typically about 10 megohms) of the
vacuum tube input stage well. They were difficult to match to early transistor equipment,
and were quickly supplanted by dynamic microphones for a time, and later small electret
condenser devices. The high impedance of the crystal microphone made it very
susceptible to handling noise, both from the microphone itself and from the connecting
cable.
Piezo transducers are often used as contact microphones to amplify sound from acoustic
musical instruments, to sense drum hits, for triggering electronic samples, and to record
sound in challenging environments, such as underwater under high pressure. Saddlemounted pickups on acoustic guitars are generally piezos that contact the strings passing
over the saddle. This type of microphone is different from magnetic coil pickups
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commonly visible on typical electric guitars, which use magnetic induction rather than
mechanical coupling to pick up vibration.
Speakers as microphones
A loudspeaker, a transducer that turns an electrical signal into sound waves, is the
functional opposite of a microphone. Since a conventional speaker is constructed much
like a dynamic microphone (with a diaphragm, coil and magnet), speakers can actually
work "in reverse" as microphones. The result, though, is a microphone with poor quality,
limited frequency response (particularly at the high end), and poor sensitivity. In practical
use, speakers are sometimes used as microphones in such applications as intercoms or
walkie-talkies, where high quality and sensitivity are not needed.
However, there is at least one other practical application of this principle: using a
medium-size woofer placed closely in front of a "kick" (bass drum) in a drum set to act as
a microphone. The use of relatively large speakers to transducer low frequency sound
sources, especially in music production, is becoming fairly common. Since a relatively
massive membrane is unable to transducer high frequencies, placing a speaker in front of
a kick drum is often ideal for reducing cymbal and snare bleed into the kick drum sound.
Less commonly, microphones themselves can be used as speakers, almost always as
tweeters. This is less common since microphones are not designed to handle the power
that speaker components are routinely required to cope with. One instance of such an
application was the STC microphone-derived 4001 super-tweeter, which was successfully
used in a number of high quality loudspeaker systems from the late 1960s to the mid-70s.
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2.3 Loudspeakers
A loudspeaker, speaker, or speaker system is an electro acoustical transducer that
converts an electrical signal to sound. The term loudspeaker can refer to individual
transducers (known as drivers), or to complete systems consisting of a enclosure
incorporating one or more drivers and electrical filter components. Loudspeakers, just as
with other electro acoustic transducers, are the most variable elements in an audio system
and are responsible for the greatest degree of audible differences between sound systems.
To adequately reproduce a wide range of frequencies, most loudspeaker systems require
more than one driver, particularly for high sound pressure level or high accuracy.
Individual drivers are used to reproduce different frequency ranges. The drivers are
named subwoofers (very low frequencies), woofers (low frequencies), mid-range
speakers (middle frequencies), tweeters (high frequencies) and sometimes super tweeters
optimized for the highest audible frequencies. The terms for different speaker drivers
differ depending on the application. In 2-way loudspeakers, there is no "mid-range"
driver, so the task of reproducing the midrange sounds falls upon the woofer and tweeter.
Home stereos use the designation "tweeter" for high frequencies whereas professional
audio systems for concerts may designate high frequency drivers as "HF" or "highs" or
"horns". When multiple drivers are used in a system, a "filter network", called a
crossover, separates the incoming signal into different frequency ranges, and routes them
to the appropriate driver. A loudspeaker system with n separate frequency bands is
described as "n-way speakers": a 2-way system will have woofer and tweeter speakers; a
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3-way system is either a combination of woofer, mid-range and tweeter or
subwoofer, woofer tweeter.
Figure2.6 Loudspeaker
History
The modern design of moving-coil drivers was established by Oliver Lodge in (1898)[2].
The moving coil principle was patented in 1924 by Chester W. Rice and Edward W.
Kellogg. These first loudspeakers used electromagnets because large, powerful
permanent magnets were generally not available at a reasonable price. The coil of an
electromagnet, called a field coil, was energized by current through a second pair of
connections to the driver. This winding usually served a dual role, acting also as a choke
coil filtering the power supply of the amplifier to which the loudspeaker was connected.
AC ripple in the current was attenuated by the action of passing through the choke coil;
however, AC line frequencies tended to modulate the audio signal being sent to the voice
coil and added to the audible hum of a powered-up sound reproduction device. In the
1930s, loudspeaker manufacturers began to combine two and three bandpasses worth of
drivers in order to increase frequency response and sound pressure level. In 1937, the first
film industry standard loudspeaker system, "The Shearer Horn System for Theatres" (a
two-way system) was introduced by Metro-Goldwyn-Mayer. It used four 15" low
frequency drivers, a crossover network set for 375 Hz and a single sectoral horn with two
compression drivers providing the high frequencies. John Kenneth Hilliard, James
Bullough Lansing and Douglas Shearer all played roles in creating the system. At the
1939 New York World's Fair, a very large two-way public address system was mounted
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on a tower at Flushing Meadows. The eight 27" low-frequency drivers were designed by
Rudy Bozak in his role as chief engineer for Cinaudagraph. High frequency drivers were
likely made by Western Electric. Altec introduced their coaxial Duplex driver in 1943,
incorporating a high frequency horn sending sound through the middle of a 12-inch
woofer for near-point-source performance Altec's "Voice of the Theatre" loudspeaker
system arrived in the marketplace in 1945, offering better coherence and clarity at the
high power levels necessary in movie theaters. The Academy of Motion Picture Arts and
Sciences immediately began testing its sonic characteristics; they made it the film house
industry standard in 1955. Subsequently, continuous developments in enclosure design
and materials led to significant audible improvements. The most notable improvements in
modern speakers are improvements in cone materials, the introduction of higher
temperature adhesives, improved permanent magnet materials, improved measurement
techniques, computer aided design and finite element analysis.
Driver design
The most common type of driver uses a lightweight but sometimes heavy diaphragm
connected to a rigid basket, or frame, via flexible suspension that constrains a coil of fine
wire to move axially through a cylindrical magnetic gap. When an electrical signal is
applied to the voice coil, a magnetic field is created by the electric current in the voice
coil which thus becomes an electromagnet field. The coil and the driver's magnetic
system interact, generating a mechanical force which causes the coil, and so the attached
cone, to move back and forth and so reproduce sound under the control of the applied
electrical signal coming from the amplifier. The following is a description of the
individual components of this type of loudspeaker. The diaphragm is usually
manufactured with a cone or dome shaped profile. A variety of different materials may be
used, but the most common are paper, plastic and metal. The ideal material would be stiff
(to prevent uncontrolled cone motions), light (to minimize starting force requirements)
and well damped (to reduce vibrations continuing after the signal has stopped). In
practice, all three of these criteria cannot be met simultaneously using existing materials,
and thus driver design involves tradeoffs. For example, paper is light and typically well
damped, but not stiff; metal can be made stiff and light, but it is not usually well damped;
plastic can be light, but typically the stiffer it is made, the less well-damped it is. As a
result, many cones are made of some sort of composite material. This can be a matrix of
fibers including Kevlar or fiberglass, a layered or bonded sandwich construction, or
23
3 Amp
2. Sound Transducers
Week 5
simply a coating applied to stiffen or damp a cone. The basket or frame must be designed
for rigidity to avoid deformation, which will change the magnetic conditions in the
magnet gap, and could even cause the voice coil to rub against the walls of the magnetic
gap. Baskets are typically cast or stamped metal, although molded plastic baskets are
becoming common, especially for inexpensive drivers. The frame also plays a
considerable role in conducting heat away from the coil. The suspension system keeps the
coil centered in the gap and provides a restoring force to make the speaker cone return to
a neutral position after moving. A typical suspension system consists of two parts: the
"spider", which connects the diaphragm or voice coil to the frame and provides the
majority of the restoring force; and the "surround", which helps center the coil/cone
assembly and allows free pistonic motion aligned with the magnetic gap. The spider is
usually made of a corrugated fabric disk, generally with a coating of a material intended
to improve mechanical properties. The name "spider" derives from the shape of early
suspensions, which where two concentric rings of bakelite material, joined by six or eight
curved "legs". Variations of this topology included adding a felt disc to provide a barrier
to particles that might otherwise cause the voice coil to rub. Another German company
currently offers a spider made of wood. The surround can be a roll of rubber or foam, or a
ring of corrugated fabric (often coated), attached to the outer circumference of the cone
and to the frame. The choice of suspension materials affects driver lifetime, especially in
the case of foam surrounds which are susceptible to aging and environmental damage.
The wire in a voice coil is usually made of copper, though aluminum, and rarely silver,
may be used. Voice coil wire cross sections can be circular, rectangular, or hexagonal,
giving varying amounts of wire volume coverage in the magnetic gap space. The coil is
oriented coaxially inside the gap, a small circular volume (a hole, slot, or groove) in the
magnetic structure within which it can move back and forth. The gap establishes a
concentrated magnetic field between the two poles of a permanent magnet; the outside of
the gap being one pole and the center post (a.k.a., the pole-piece) being the other. The
pole piece and back plate are often a single piece called the pole plate or yoke. Modern
driver magnets are almost always permanent and made of ceramic, ferrite, Alnico, or,
more recently, neodymium magnet. A current trend in design, due to increases in
transportation costs and a desire for smaller, lighter devices (as in many home theater
multi-speaker installations), is the use of neodymium magnet instead of ferrite types.
Very few manufacturers use electrically powered field coils as was common in the
earliest designs. The size and type of magnet and details of the magnetic circuit differ,
depending on design goals. For instance, the shape of the pole piece affects the magnetic
interaction between the voice coil and the magnetic field, and is sometimes used to
24
3 Amp
2. Sound Transducers
Week 5
modify a driver's behavior. A "shorting ring" or Faraday loop may be included as a thin
copper cap fitted over the pole tip, or as a heavy ring situated within the magnet-pole
cavity. The benefits of this are reduced impedance at high frequencies providing
extended treble output, reduced harmonic distortion, and a reduction in the inductance
modulation that typically accompany large voice coil excursions. On the other hand, the
copper cap requires a wider voice coil gap, with increased magnetic reluctance, reducing
available flux, requiring a slightly larger magnet for equivalent performance. Driver
design, and the combination of one or more drivers into an enclosure to make a speaker
system, is both an art and science. Adjusting a design to improve performance is done
using magnetic, acoustic, mechanical, electrical, and material science theory, high
precision measurements, and the observations of experienced listeners. Designers can use
an anechoic chamber to ensure the speaker can be measured independently of room
effects, or any of several electronic techniques which can, to some extent, replace such
chambers. Some developers eschew anechoic chambers in favor of specific standardized
room setups intended to simulate real-life listening conditions. A few of the issues
speaker and driver designers must confront are distortion, lobing, phase effects, off axis
response and crossover complications. The fabrication of finished loudspeaker systems
has become segmented, depending largely on price, shipping costs, and weight
limitations. High-end speaker systems, which are heavier (and often larger) than
economic shipping allows outside local regions, are usually made in their target market
area and can cost $140,000 or more per pair.[7] The lowest-priced speaker systems and
most drivers are manufactured in China or other low-cost manufacturing locations.
Driver types
An audio engineering rule of thumb is that individual electrodynamic drivers provide
quality performance over at most about 3 octaves. Multiple drivers (e.g., subwoofers,
woofers, mid-range drivers, tweeters) are generally used in a complete loudspeaker
system to provide performance beyond 3 octaves.
25
3 Amp
2. Sound Transducers
Week 5
Figure2.7: Exploded view of a dome tweeter
Full range drivers
A full-range driver is designed to have the widest frequency response possible, despite
the rule of thumb cited above. These drivers are small, typically 3 to 8 inches (7 to 20
cm) in diameter to permit reasonable high frequency response, and carefully designed to
give low distortion output at low frequencies, though with reduced maximum output
level. Full range (or more accurately wide range) drivers are most commonly heard in
public address systems, and in televisions, although some models are suitable for hi-fi
listening. In hif-fi speaker systems, the use of wide range drive units can avoid
undesirable interaction between multiple drivers, caused by non-coincident driver
location, or crossover network issues. Fans of wide range driver hi-fi speaker systems
claim a coherence of sound, said to be due to the single source and a resulting lack of
interference, and likely to the lack of crossover components. Detractors typically cite the
wide range driver's limited frequency response and their modest output abilities, together
with their requirement for large, elaborate, expensive enclosures, such as transmission
lines, or horns, to approach optimum performance. Full range drivers often employ an
additional cone called a whizzer: a small, light cone attached to the joint between the
voice coil and the primary cone. The whizzer cone extends the high frequency response
of the driver and broadens its high frequency directivity, which would otherwise be
greatly narrowed due to the outer diameter cone material failing to keep up with the
central voice coil at higher frequencies. The main cone in a whizzer design is
manufactured so as to flex more in the outer diameter than in the center. The result is that
the main cone delivers low frequencies and the whizzer cone contributes most of the
higher frequencies. Since the whizzer cone is smaller than the main diaphragm, output
dispersion at high frequencies is improved relative to an equivalent single larger
diaphragm.
Limited-range drivers are typically noted in computers, toys, and clock radios. These
drivers are less elaborate and less expensive than wide range drivers, and they may be
severely compromised to fit into very small mounting locations. In this application, sound
quality is a low priority. The human ear is remarkably tolerant of poor sound quality, and
the distortion inherent in limited range drivers may enhance their output at high
frequencies, increasing clarity when listening to spoken word material.
26
3 Amp
2. Sound Transducers
Week 5
27
3.
Modulation Techniques
Week 6
3.1 Amplitude modulation
Amplitude modulation (AM) is a technique used in electronic communication, most
commonly for transmitting information via a radio carrier wave. AM works by varying
the strength of the transmitted signal in relation to the information being sent. For
example, changes in the signal strength can be used to reflect the sounds to be reproduced
by a speaker, or to specify the light intensity of television pixels. (Contrast this with
frequency modulation, also commonly used for sound transmissions, in which the
frequency is varied; and phase modulation, often used in remote controls, in which the
phase is varied). In the mid-1870s, a form of amplitude modulation—initially called
"undulatory currents"—was the first method to successfully produce quality audio over
telephone lines. Beginning with Reginald Fessenden's audio demonstrations in 1906, it
was also the original method used for audio radio transmissions, and remains in use today
by many forms of communication—"AM" is often used to refer to the medium wave
broadcast band (see AM radio).
Figure3.1: Signals in AM
We shall now develop the mathematical expressions that represent AM signals.
Assuming that the modulating signal is a sine wave in fm frequency:
Vm(t) = Vm cos m t
28
3.
Modulation Techniques
Week 6
and the carrier wave is in fc frequency:
Vc(t) = Vc cos c t
The expression representing the modulated wave is:
VAM(t) = (Vc + Vm cos m t)cos c t = A(t)cos c t (1.1)
where:
Vc
-
amplitude of carrier wave
Vm
-
amplitude of signal wave
c
-
2π  f C
m
-
2π  f m
Equation (1.1) describes a sine wave whose frequency fc (c) is that of the carrier wave
and whose amplitude A(t) is behaving as another sine wave whose frequency is fm with
the average value of VC.
By extracting Vc from the brackets we obtain:


V
VAM (t )  Vc  1  m cos m t  cos c t  Vc (1  m cos m t ) cos c t



Vc


A( t )
(1.2)
where m is called the modulation coefficient:
m
Vm
Vc  mVc = Vm
The modulation coefficient shows the relationship between the original carrier wave
amplitude and the signal wave amplitude. Figure 1-2 shows AM waves with the same
carrier and signal frequencies but with different modulation coefficients.
29
3.
Modulation Techniques
Week 6
Amplitude
m=0
t
m=0.3
t
m=0.8
t
m=1
t
m=3
t
Figure3.2 AM Waveforms for Various Modulation Coefficients
30
3.
Modulation Techniques
Week 6
Figure 3.3 AM signal containing not only the carrier and sidebands but also the
modulating signal.
31
3 . Modulation Techniques
Week 7
3.2 Frequency Modulation
The frequency modulated wave has a constant power and its frequency is changing as a function
of time:
V(t )  VC  cos2f (t )
Another description of a frequency modulated wave is as follows:
V(t )  VC  cos2(f C  fd(t )t )
(3.1)
fd(t) indicates the frequency deviation from the carrier wave frequency, which comes from the
modulated wave. The frequency deviation depends on the modulated wave power.
f d  K f  Vm
(3.2)
Kf is the constant, which describes the connection between the frequency deviation and the
modulated wave voltage.
Vm (the maximum power of the modulating wave) will cause a maximum frequency deviation.
Thus, fd (not fd(t)) is called a maximum frequency deviation.
The wave described in equation (1.8) is hard to analyze. Mathematical dismantling of this
wave will give the customary equation of a frequency modulated wave:
V(t )  VC  cos2fCt  (2fm t )
(3.3)
Where:
Vc
fc
fn

-
The power of the carrier wave.
The frequency of the carrier wave.
The frequency of the modulating wave.
The modulation coefficient.
The modulation coefficient is the relation between the maximum frequency deviation and
the frequency modulating wave.
32
3 . Modulation Techniques

Week 7
fd
fm
There is no connection between fd and fm.
fm is the frequency modulating wave. It determines the frequency change rate, but not the
maximum deviation frequency.
fd is the maximum frequency deviation, which is determined by the modulating wave
amplitude and not its frequency.
The change in the modulated wave frequency is in the range fcfd. Thus, the frequency
change range is 2fd.
 allows us to analyze the wave mathematically and its spectrum.
Observing equation (1.10) we can see that it looks as follows:
V(t )  VC 2f Ct  (t )
(1.11)
Thus, this equation also suits phase modulation.
This is true, because phase modulation and frequency modulation, even though they are
created differently, they create a wave, which behaves similarly (mathematically and
practically).
The main advantages of FM over AM are:
1. Improved signal to noise ratio (about 25dB) w.r.t. to man made interference.
2. Smaller geographical interference between neighboring stations.
33
3 . Modulation Techniques
Week 7
3. Less radiated power.
4. Well defined service areas for given transmitter power.
Disadvantages of FM:
1. Much more Bandwidth (as much as 20 times as much).
2. More complicated receiver and transmitter.
Figure3.4: Frequency modulation
34
3 . Modulation Techniques
Week 7
35
3 . Modulation Techniques
Week 7
36
3. Modulation Techniques
Week 8
otaludom edutilpmArs
There are two types of amplitude modulators. They are low-level and highlevel modulators. Low-level modulators generate AM with small signals and
must be amplified before transmission. High-level modulators produce AM at
high power levels, usually in the final amplifier stage of a transmitter.
Low-Level AM: Diode Modulator
Diode modulation consists of a resistive mixing network, a diode rectifier, and
an LC tuned circuit. The carrier is applied to one input resistor and the
modulating signal to another input resistor. This resistive network causes the
two signals to be linearly mixed (i.e. algebraically added). A diode passes half
cycles when forward biased and the coil and capacitor repeatedly exchange
energy, causing an oscillation or ringing at the resonant frequency.
Figure3.5:Low-Level AM: Diode modulator
Low-Level AM: Transistor Modulator
Transistor modulation consists of a resistive mixing network, a transistor, and
an LC tuned circuit. The emitter-base junction of the transistor serves as a
diode and nonlinear device. Modulation and amplification occur as base current
controls a larger collector current. The LC tuned circuit oscillates (rings) to
generate the missing half cycle.
37
Low-Level AM: PIN Diode Modulato
Variable attenuator circuits using PIN diodes produce AM at 
VHF, UHF, and microwave frequencies. PIN diodes are special
type silicon junction diodes designed for use at frequencies above
100 MHz. When PIN diodes are forward-biased, they operate as
variable resistors. Attenuation caused by PIN diode circuits
varies with the amplitude of the modulating signal.
High-frequency amplitude modulators using PIN diodes.
38
Low-Level AM: Differential Amplifier
The modulating signal is applied to the base of a constant-current source
transistor. The modulating signal varies the emitter current and therefore
the gain of the circuit.
(a) Basic differential amplifier. (b) Differential amplifier modulator.
High-Level AM
In high-level modulation, the modulator varies the voltage and
power in the final RF amplifier stage of the transmitter. The
result is high efficiency in the RF amplifier and overall highquality performance.

High-Level AM: Collector Modulator
The collector modulator is a linear power amplifier that takes the 
low-level modulating signals and amplifies them to a high-power
level. A modulating output signal is coupled through a
modulation transformer to a class C amplifier. The secondary
winding of the modulation transformer is connected in series with
the collector supply voltage of the class C amplifier.
39

A high-level collector modulator
40
3 Modulation Techniques
Week 9
3.4 Frequency Modulators
With frequency modulation the carrier signal frequency is varied at an audio rate. The
amount of frequency shift is based on the amplitude of the modulating frequency. A
voltage controlled oscillator can be used to perform frequency modulation. The
modulating signals amplitude is the frequency controlling voltage for the oscillator.
Figure
Principles of Oscillator operation
Every oscillator has at least one active device be it a transistor or even the old valve.
This active device and, acts as an amplifier. At turn on, when power is first applied,
random noise is generated within our active device and then amplified. This noise is
fed back positively through frequency selective circuits to the input where it is
amplified again and so on. Ultimately a state of equilibrium is reached where the
losses in the circuit are made good by consuming power from the power supply and
the frequency of oscillation is determined by the external components, be they
inductors and capacitors (L.C.) or a crystal. The amount of positive feedback to
sustain oscillation is also determined by external components.
40
3 Modulation Techniques
Week 9
Voltage controlled Oscillator
A voltage controlled oscillator (VCO) is an oscillator where the principal variable
or tuning element is a varactor diode. The VCO is tuned across its band by a
"clean" dc voltage applied to the diode to vary the net capacitance applied to the
tuned circuit.
PHASE-LOCKED LOOP
A phase-locked loop (PLL) is a feedback system that is used to lock the output frequency
and phase to the frequency and phase of a reference signal at its input. The reference
waveform can be of many different types, including sinusoidal and digital. PLLs have
been used for various applications, including filtering, frequency synthesis, frequency
modulation, demodulation, and signal detection. The basic PLL consists of a voltagecontrolled oscillator(VCO), a phase detector (PD), and a filter. In its most general form,
the PLL may also contain a mixer and a frequency divider, as shown in Figure 12.20.
A basic PLL (see Fig. 1) consists of a reference oscillator, phase/frequency detector,
charge pump, loop filter, voltage controlled oscillator (VCO) and divider. With a constant
divisor of N, the loop forces the VCO frequency to be exactly N times the reference
frequency. The phase/frequency detector and charge pump deliver either positive or
negative charge pulses depending on whether the reference signal phase leads or lags the
divided VCO signal phase. These charge pulses are integrated by the loop filter to
41
3 Modulation Techniques
Week 9
generate a tuning voltage to move the VCO frequency up or down until the phases are
synchronized.
PLLs are
Figure 3.1: Basic PLL Block Schematic
42
4. Demodulators
Week 10
4.1 Amplitude Demodulators
Demodulators, or detectors, are circuits that accept modulated signals and
recover the original modulating information.
Figure4.1: Diode detector
Principle of operation of a diode detector
On positive alternations of the AM signal, the capacitor charges quickly to the
peak value of pulses passed by the diode. When the pulse voltage drops to zero,
the capacitor discharges into the resistor. Since time constant of the capacitor
and resistor is long compared to the period of the carrier the capacitor discharges
only slightly when the diode is not conducting. The resulting waveform across the
capacitor is a close approximation to the original modulating signal:
 Because the diode detector recovers the envelope of the AM (modulating)
signal, the circuit is sometimes called an envelope detector.
 If the RC time constant in a diode detector is too long, the capacitor
discharge will be too slow to follow the faster changes in the modulating
signal.
 This is referred to as diagonal distortion.
42
Synchronous detector
Synchronous detectors use an internal clock signal at the carrier frequency in the
receiver to switch the AM signal off and on, producing rectification similar to that
in a standard diode detector. Synchronous detectors or coherent detectors have
less distortion and a better signal-to-noise ratio than standard diode detectors.
The key to making the synchronous detector work is to ensure that the signal
producing the switching action is perfectly in phase with the received AM carrier.
An internally generated carrier signal from an oscillator will not work.
Figure4.2: A practical synchronous detector
43
4. Demodulators
Week11
4.2Frequency demodulator
Two important characteristics that all FM detectors must provide:
•
•
FM demodulators must convert frequency variations of the input signal into amplitude
variations at the output
The amplitude of the output must be proportional to the frequency deviation of the input
Figure4.3
•
There are four major types of FM detectors:
– Foster-Seely discriminator
– Ratio detector
– Quadrature detector
– PLL detector
The schematic diagrams of the four FM detectors are shown below.
43
Figure4.4: Foster-seely detector
Ratio Detector
Figure4.5: Ratio Detector
44
Figure4.6 Quadrature Detector
3.8 The Phase Locked Loop
The PLL can be used for FM demodulation. The audio rate of frequency shift of the
carrier when compared with the steady VCO output would result in a changing of the
VCO control voltage at the audio rate. The carrier is filtered off by the low pass filter.
This is a typical internal block diagram of an IC PLL circuit. The amplifier is used to
boost the level of the VCO control voltage.
45
5. Radio Receivers
Week12
5.1 Tuned radio receiver
Two important specifications are fundamental to all receivers:
–
–
Sensitivity: signal strength required to achieve a given signal-to-noise ratio
Selectivity: the ability to reject unwanted signals
Selectivity
Selectivity refers to the ability of a receiver to differentiate between the desired signal and other
undesired frequencies.
46
Initial selectivity is obtained using LC tuned circuits like the parallel resonant circuit depicted
below. The filter characteristic of an RLC circuit does not provide ideal selectivity. An ideal filter
would provide constant gain across the passband.
Q=
fr
X
2 f r L
 L 
BW
R
R
fr 
1
2
LC
Selectivity Problems
Overly selective receiver results in a loss of fidelity due to clipping of upper frequencies.
47
Sensitivity
Sensitivity - the ability to receive weak signals with an acceptable signal-to-noise ratioOne
common specification for AM receivers is the signal strength required for a 10-dB signal-plusnoise-to-noise ratio at a specified power level.
Receiver Topologies
•
•
•
Nearly all modern receivers use the superheterodyne principle
The simplest receiver would consist of a demodulator connected directly to an antenna
Adding a tuned circuit would improve the performance
Simple Receiver
 The simplest of receivers, a “crystal radio,” consists of a tuned circuit, diode (crystal)
detector and earphones.
 Tuning is accomplished by adjusting a variable capacitor C1 to change the resonant
frequency.
48
Example Problem 1
Consider simple AM radio receiver. Tuning this radio is accomplished by adjusting a variable
capacitor C. Say we want tune this radio for middle of the AM dial (1070 kHz). Also, we desire
a 3-dB bandwidth of 6 kHz. If R = 10 W, determine the require values of L and C.
Tuned radio receiver
In a receiver with multiple RF stages, all tuned circuits must track together, typically by
ganged-tuning methods as shown: In the TRF receiver below, selectivity is improved by
cascading several RF amplifiers.
TRF receiver problems
The biggest problem with the TRF design is that selectivity varies with frequency.
The LC filter is too narrow at low frequencies and too wide at high frequencies.
Another problem is in keeping all the stages of the RF amplifiers tuned to the exact same
frequency.
49
Gang variable capacitor
Example Problem 2
In
the
previous
example,
the
bandpass
filter
had
Q = 178.3 to provide a 6-kHz bandwidth at 1070 kHz. If Q remains a constant consider the filter
selectivity at the ends of the dial (535 and 1605 kHz). For these two frequencies determine the
resulting bandwidth.
The Superheterodyne Recever
•
The superheterodyne receiver was invented in 1918 by Edwin H. Armstrong and is still
almost universally used
•
A superheterodyne receiver is characterized by one or more stages of RF amplification
and the RF stage may be tuned or broadband
50
Receiver Characteristics

Sensitivity - the ability to receive weak signals with an acceptable signal-to-noise ratio

One common specification for AM receivers is the signal strength required for a 10-dB
signal-plus-noise-to-noise ratio at a specified power level

Adjacent channel sensitivity is another way of specifying selectivity

Techniques like alternate channel rejection are also used to specify selectivity
Distortion
•
Distortion comes in several forms:
– Harmonic distortion is when the frequencies generated are multiples of those in
the original signal
–
Intermodulation distortion occurs when frequency components in the original
signal mix and produce sum and difference signals
–
Phase distortion consists of irregular shifts in phase and is common when signals
–
pass through filters
Dynamic Range
•
The ratio between between the receiver’s response to weak signals and signals that are
overload one or more stages is referred to as Dynamic Range
•
Blocking may occur when two adjacent signals, one of which is much stronger than the other,
cause a reduction in sensitivity to the desired channel.
This is also referred to as
desensitization or desense
Spurious Responses
•
Superheterodyne receivers have a tendency to receive signals they are not tuned to
•
Image Frequencies are signals that are produced as a result of the generation of
intermediate frequencies
51
5. Radio Receivers
Week13
5.2 Superherodyne Receiver
Description of heterodyning
Heterodyning is the process used at the receiver that allows one of these individual carrier
frequencies to be selected, and then shifted to a pre-defined frequency that is suitable for
the detector in the receiver. The term heterodyning means “frequency translation”, and is
the process which mixes a signal generated by a local oscillator, with that of the received
RF signal. By ensuring a fixed frequency difference between fRF and fLO the resultant
“Intermediate Frequency” is held at a constant value – but contains all of the information
in it’s sidebands previously held by the RF signal’s sidebands. This makes it possible to
design a receiver that is optimised to operate at one frequency – the intermediate
frequency, and allows decoding of the information signal completely independent of the
frequency the receiver is tuned to. In the block diagram, the heterodyning process is
carried out in the “Mixer” – the output of which is the intermediate frequency, defined by
fIM=fRF- fLO. A typical value for fIM in commercial AM broadcast receivers is
455kHz, and because when tuning the receiver the frequency of the local oscillator is also
changed, the relationship between fRF and fLO is kept constant and therefore the
intermediate frequency always remains constant.
The Motivation behind the use of the superheterodyne receiver
Before the superheterodyne receiver, simple radio receiver’s had been of the “Tuned
Radio Frequency (TRF)” type, where the required frequency is selected by the tuned
circuits in the RF amplifier, and then applied directly to the detector stage. Problems
associated with this receiver were ensuring sufficient RF gain was provided to allow a
diode detector to be used. Greater still were the problems of instability at higher
frequencies caused by several stages of amplifiers operating at non-characteristic
frequencies, and the TRF’s lack of high selectivity to reject unwanted signals. It was
therefore required to design a receiver that had it’s characteristics and circuitry optimized
for use at a single frequency – thereby ensuring stability and improving on the receiver’s
selectivity and sensitivity – the heterodyning process allows this to be implemented. The
52
TRF’s problems of bandwidth variation, insufficient adjacent-channel rejection and
instability are all solved by the superheterodyne receiver, all because the superhet
performs all amplification and filtering at a fixed frequency – nomatter what carrier frequency
the receiver is tuned to.
Block diagram of the Superheterodyne receiver
Frequency conversion
Recall that in the transmitter, a mixer is used to translate a low frequency input to a
higher frequency.The same process can be used in reverse by the receiver to translate an
RF signal down to the IF.
Mixing principles
The inputs to the mixer are the radio signal fs and a sine wave from a local oscillator fo.The mixer
output consists of four signals:
 fo + fs
 fo – fs
 fs
53
 fo
This function is called heterodyning
Selective filters
The output of the mixer is filtered to eliminate everything but the IF signal.
Figure
54
Tuning a superhet receiver
In a TRF receiver, a station is tuned by adjusting the resonant frequency of a filter.In a
superhet receiver, a station is tuned by changing the frequency of the receiver’s local
oscillator fo and:
 The oscillator is set such that fo - fs = fIF
 fIF is a fixed value (typically 455-kHz for AM radio).
Superhet advantages
TRF receivers suffered because changing the resonant frequency of the filter produced a
changing filter bandwidth. Selectivity varies with frequency. In superhet receivers, all the
filtering (selectivity) occurs at a single, fixed intermediate frequency (IF).
IF selectivity
Since IF is typically a lower frequency than RF, it is easier to obtain a more selective IF
filter.
55
Multiple Conversion Superheterodyne
In receivers tuning the upper HF and the VHF bands, two (or even more) IF channels
are commonly used with two (or more) stages of frequency conversion. The lowest
frequency IF channel provides the selectivity or bandwidth control that is needed and the
highest frequency IF channel is used to achieve good Image rejection. A typical system
used in two metre FM amateur transceivers is shown in Figure 6. In this system, IF
channels of 10.7 MHz and 455 kHz are used with double conversion. The requirement Is
different to that of the wideband FM broadcasting system as frequency deviation is only 5
kHz with an audio frequency spectrum limited to below 2.5 kHz. Channel spacing is 25
kHz and bandwidth is usually limited to less than 15 kHz so that the narrower bandwidth
455 kHz IF channel is suitable.
Figure
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Differences between receiver designed for AM and FM
Both AM and FM receivers use the superheterodyne method, but the obvious
difference between these receivers is that an FM receiver uses an FM demodulator and
requires an “amplitude limiter” following the IF section to remove variations in the
carrier amplitude caused by noise and interference – ensuring the input to the FM
demodulator is sinusoidal and of constant amplitude. Furthermore with the FM receiver,
much higher operating frequencies are used, the bandwidths of the RF and IF stages are
different from those used in AM and because of this, different intermediate frequencies
are used.
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4.3 AM Superherodyne Receiver
AM Broadcasting
AM broadcasting technique includes:
 Allocated the band 530 kHz – 1600 kHz (with minor variations)
 10 kHz per channel. (9 kHz in some countries)
 More that 100 stations can be licensed in the same geographical area.
 Uses AM modulation (DSB + C)
 In radio communication systems, the transmitted signal is very weak when it reaches the
receiver, particularly when it has traveled over a long distance.
 The signal has also picked up noise of various kinds.
 Receivers must provide the sensitivity and selectivity that permit full recovery of the
original signal.
 The radio receiver best suited to this task is known as the superheterodyne receiver.
Sensitivity
A communication receiver’s sensitivity, or ability to pick up weak signals,
is a function of overall gain, the factor by which an input signal is
multiplied to produce the output signal. The higher the gain of a receiver,
the better its sensitivity.
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The more gain that a receiver has, the smaller the input signal necessary to
produce a desired level of output. High gain in receivers is obtained by
using multiple amplification stages.
Selectivity
A receiver with good selectivity will isolate the desired signal and greatly
attenuate/eliminates other signals. To improve selectivity is to add stages of
amplification, both before and after demodulator
Superheterodyne receivers
Superheterodyne receivers convert all incoming signals to a lower frequency,
known as the intermediate frequency (IF), at which a single set of amplifiers is
used to provide a fixed level of sensitivity and selectivity. Gain and selectivity are
obtained in the IF amplifiers. The key circuit is the mixer, which acts like a simple
amplitude modulator to produce sum and difference frequencies. The incoming
signal is mixed with a local oscillator signal.
Figure Block diagram of the Superheterodyne AM receiver
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RF Amplifier Stage
The term RF stands for radio- frequency, and it is the RF stage of the receiver that
couples the aerial to the receiver and minimizes the number of frequencies that could
cause problems with the heterodyning process at the mixer. The RF itself is defined as the
carrier frequency of the desired audio signal that we wish to detect, and it is the
relationship of the RF’s sidebands to it’s carrier that we wish to preserve in the IF, and
demodulate – thus providing the baseband information. The antenna picks up the weak
radio signal and feeds it to the RF amplifier and provides some initial gain and selectivity
and are sometimes called preselectors.
Mixer Stage
The mixer carries out the heterodyning process, by mixing the wanted signal frequency
fRF and the local oscillator frequency fLO, and producing the intermediate frequency
(fLO- fRF), and the image frequency (fLO+fRF). The intermediate frequency is a
constant value, therefore, the local oscillator must be tuneable to the frequency of the
receiver plus the intermediate frequency. To allow the local oscillator and the RF stage to
be tuned simultaneously, they are connected together by mounting on a common spindle
– a process called ganging. The preservation of the correct intermediate frequency
between the local oscillator and RF stage is called tracking.
Figure13: Concept of a mixer
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Mixing Principles
Mixers accept two inputs: The signal to be translated to another frequency is applied to
one input, and the sine wave from a local oscillator is applied to the other input. Like an
amplitude modulator, a mixer essentially performs a mathematical multiplication of its
two input signals. The oscillator is the carrier, and the signal to be translated is the
modulating signal. The output contains not only the carrier signal but also sidebands
formed when the local oscillator and input signal are mixed.
Local Oscillator
The Local Oscillator frequency is used during the heterodyning process to produce the
Intermediate frequency, and is kept at a fixed value relative to the RF input frequency
commonly 455kHz higher than the RF. By gang-tuning the RF stage and the IF stage, the
relationship remains constant hence the intermediate frequency remains constant, no
matter what audio frequency we wish to listen to.
 What should be the frequency of the local oscillator used for translation from RF to IF?
fLO = fc + fIF
(up-conversion)
or
fLO = fc - fIF
(down-conversion)
 Tuning ratio = fLO, max / fLO, min
 Up-Conversion: (1600 + 455) / (530+455) ≈ 2
 Down-Conversion: (1600–455) / (530–455) ≈ 12
Easier to design oscillator with small tuning
Image Frequency
The image frequency is a disadvantageous by-product of the heterodyning
process, which makes the receiver susceptible to frequencies transmitted at twice
the intermediate frequency either above or below the actual RF frequency
(depending on whether the local oscillator tracks above or below the tuned RF
signal). If tuning to 5975kHz and the IF is 455kHz, images of the station would
appear at either 5065kHz or 6885kHz. If there so happens to be stations
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transmitting on either of these frequencies, audio information from this frequency
and the RF frequency will leave the mixer stage and enter the detector – leading to
a high pitched squealing effect as both these signals are demodulated.
IF Amplifiers
 The primary objective in the design of an IF stage is to obtain good selectivity.
The intermediate frequency is a fixed single frequency (commonly 455kHz) that
the detector
and IF amplifier of the circuit are optimized to operate at. By
heterodyning the RF signal to the IF frequency, the performance of the circuit
and the sensitivity of the radio are improved.The IF is found by subtracting the RF
from the local oscillator frequency.Since the intermediate frequency is usually
lower than the input frequency, IF amplifiers are easier to design and good
selectivity is easier to obtain.
Demodulators
The detector or demodulator recovers the original baseband signal from the IF by
standard FM/AM demodulation techniques, and passes this through the Audio Frequency
Amplifier and onto the speaker or audio transmission. The highly amplified IF signal is
finally applied to the demodulator, which recovers the original modulating information.
The demodulator may be a diode detector (for AM), a quadrature detector (for FM), or a
product detector (for SSB).
Advantages of Superhetrodyne
Advatages of superherodyne receivers are:
 Overcome equipment : cannot operate at high frequency
 Component operate at fixed frequency
 Optimize utilization

Reduce cost
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RF-to-IF conversion
AM Vs FM
Carrier range RF
Radio AM
Radio FM
0.535 – 1.605
88 – 108 MHz
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MHz
IF
0.455 kHz
10.7 MHz
Bandwidth IF
10 kHz
200 kHz
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5.4 FM Receiver
Principle of Operation
FM receivers, like AM receivers, utilize the super heterodyne principle, but they operate
at much higher frequencies (88 - 108 MHz).The frequency modulated FM signal enters
from the antenna and is applied to a mixer stage where another input signal is from the
local oscillator output. The signal is converted to lower frequency (intermediate
frequency IF, 455 KHz in this case) and amplified. It is then applied to a limiter stage,
which removes the amplitude variations contained in the FM signal. The next
discriminator demodulates the FM signal. The detected signal results composed of low
frequency modulating signal and continuous component, proportional to the shift
between the carrier frequency of the FM signal (after the intermediate frequency
conversion) and the frequency to which the discriminator is calibrated. Only the low
frequency component is integrated and used to control the frequency of the local
oscillator, so that an intermediate frequency equal to the central frequency of the
discriminator is obtained.
Figure13:
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Limiting Action
Limiting can be described as the action of overamplification where the signal is
overdriven in stages and subsequently "clipped". Looking at figure 5(a) below we can
imagine what happens when it is amplified and clipped (5b), amplified once again and
clipped again (5c).
Figure 5. - an a.m. modulated signal being clipped
Naturally we don't put a normal a.m. signal through a limiter, this is usually only done
with f.m. signals. I simply provided figure 5 above so you could get the general idea.
You should notice that all the amplitude modulation information (including noise) is
progressively being removed. BTW 5(b) and (c) were simply done graphically by
taking (a) resizing the height by 150% and cutting off the excess height (top and
bottom) and repeating that exercise for (c). This is exactly what happens in a limiter
only to a much greater amplification!. To give you some idea of the amplification
required for proper limiting go back to the old vacuum tube days where a good a.m. i.f. amplifier might contain three vacuum tubes. In the same period a good f.m. receiver
may have had twelve or more tubes in the i.f./limiter stage.
Means of Detection
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The FM demodulators must convert frequency variations of the input signal into
amplitude variations at the output. A number of f.m. detection schemes have evolved
over the years. The principal discrete ones Were:
F.M. Discriminator (figure 6)
This discriminator simply works on the principal that with no modulation
applied to the carrier there is no ouput at the detector. Briefly T1 converts the
f.m. signal to a.m. and when rectified the output is still zero because they
would be equal but opposite in polarity, if modulation is applied then there is a
shift in the phase of the input component with a corresponding difference in
the signals out of the diodes.
Figure 6. - an f.m. discriminator
The difference between these outputs is the audio. As an aside, this is somewhat
similar to some Automatic Fine Tuning (A.F.T.) schemes in some a.m. receivers,
notably early T.V. receivers. With no frequency variation there is no output, with
frequency drift there will be an output difference (in either direction) which is
amplified and applied to front end tuning diodes for correction.
(b) Ratio Detector
The schematic looks a little similar to figure 6 but has a third (tertiary) winding on the
secondary of T1, diode D2 has its polarity reversed and the two divider resistors are
replaced by capacitors. This scheme was quite popular in entertainment type receivers.
You detect f.m. but NOT a.m. and it placed some relaxation on the severe limiting
requirements.
(c) Crystal Discriminator
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Once favoured by radio amateurs but superseded by later I.C. designs
(d) Phase Lock Loops
Among the relatively newer designs and PLL's overcome many of the drawbacks and
costs associated with building and aligning LC discriminators.
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