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

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TABLE OF CONTENTS Week1: Introduction to communication systems …………………….

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Week2: Elements of a communication systems……………………………

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Week3: Types of transducers……………………………………………….

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Week4: Types of microphones ………………………………………………..

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Week5: Loudspeakers………………………………………………………………………

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Week6: Amplitude modulation……………………………………….

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Week7: Frequency Modulation…………………………………………..

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Week8: Amplitude modulator…………………………………………….

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Week9: Frequency Modulators…………………………………………………………………..

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Week10: Amplitude Demodulators………………………………………

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Week11: Frequency demodulator………………………………………

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Week12: Tuned radio receiver………………………………………

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Week13: Superherodyne Receiver

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……………………………………………

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Week14: AM Superherodyne Receiver ……………………………………………...

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Week15: FM Receiver………………………………………………….

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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:

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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? 2

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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. 3

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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. 4

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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: 9

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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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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. 11

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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 15

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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. 17

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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 18

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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 22

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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

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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

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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

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2. Sound Transducers

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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

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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

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3.

Modulation Techniques

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Figure 3.3 AM signal containing not only the carrier and sidebands but also the modulating signal.

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3 . Modulation Techniques

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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

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35

3 . Modulation Techniques

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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

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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.

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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

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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

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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

56

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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Week14

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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5. Radio Receivers

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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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5. Radio Receivers

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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 61

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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 63

5. Radio Receivers

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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: 65

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 67

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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