VINAYAKA MISSION’S UNIVERSITY
AARUPADAI VEEDU INSTITUTE OF TECHNOLOGY
DEPARMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING
UNIT – I, 2 MARKS
PART-A
1. Define modulation?
Modulation is a process by which some characteristics of high frequency carrier signal is varied in accordance with the instantaneous value of the modulating signal.
2. What are the types of analog modulation?
Amplitude modulation.
Angle Modulation
1. Frequency modulation
2. Phase modulation.
3. What are the types of AM modulators?
There are two types of AM modulators. They are
Linear modulators
Non-linear modulators
4. What is single tone and multi tone modulation?
If modulation is performed for a message signal with more than one frequency component then the modulation is called multi tone modulation.
If modulation is performed for a message signal with one frequency component then the modulation is called single tone modulation.
5. Compare AM with DSB-SC and SSB-SC.
6. What are the advantages of VSB-AM?
1. It has bandwidth greater than SSB but less than DSB system.
2. Power transmission greater than DSB but less than SSB system.
3. No low frequency component lost. Hence it avoids phase distortion.
7. Define amplitude Modulation
Amplitude Modulation is the process of changing the amplitude of a relatively high frequency carrier signal in proportion with the instantaneous value of the modulating signal.
8. Define Modulation index and percent modulation for an AM wave
Modulation index is a term used to describe the amount of amplitude change present in an AM waveform .It is also called a coefficient of modulation. Mathematically modulation index is
m = Em/Ec
Where m = Modulation coefficient
Em = Peak change in the amplitude of the output waveform voltage.
Ec = Peak amplitude of the unmodulated carrier voltage.
Percent modulation gives the percentage change in the amplitude of the output wave when the carrier is acted on by a modulating signal.
9. Define Low level Modulation.
In low level modulation, modulation takes place prior to the output element of the final stage of the transmitter. For low level AM modulator class A amplifier is used.
10. Define AM Vestigial sideband
AM vestigial sideband is a form of amplitude modulation in which the carrier and one complete sideband are transmitted, but only part of the second sideband is transmitted.
11. What are the advantages of single sideband transmission?
The advantages of SSBSC are
1. Power conservation: Normally ,with single side band transmission only one sideband is transmitted and the carrier is suppressed. So less power is required to produce essentially the same quality signal.
2. Bandwidth conservation: Single sideband transmission requires half as much bandwidth as conventional AM double side band transmission.
3. Noise reduction: Because a single side band system utilizes half as much bandwidth as conventional AM,the thermal noise power is reduced to half that of a double side band system.
12. Define demodulation
Demodulation or detection is the process by which modulating voltage is recovered from the modulated signal. It is the reverse process of modulation.
PART-B
1. Derive amplitude modulated wave equation and explain each term with the help of frequency spectrum.
v In amplitude modulation, the amplitude of a carrier signal is varied according to variations in the amplitude of modulating signal.
v The carrier frequency remains same, but its amplitude varies according to amplitude variations of the modulating signal.
The instantaneous value of the amplitude modulated wave can be given as,
(a) Sinusoidal modulating signal (b) Sinusoidal high frequency carrier (c) Amplitude modulated signal
The ratio of maximum amplitude of modulating signal to maximum amplitude of carrier signal is called Modulation Index.
v Value of Em must be less than value of Ec to avoid any distortion in the modulated signal. Hence maximum value of modulation index will be equal to 1
v when Em = Ec. Minimum value will be zero. When modulation index is expressed in percentage, it is also called percentage modulation.
v The modulated carrier has new signals at different frequencies, called side frequencies or sidebands. They occur above and below the carrier frequency.
We know that bandwidth of the signal can be obtained by taking the difference between highest and lowest frequencies. From above figure we can obtain bandwidth of AM wave as,
Thus bandwidth of AM signal is twice of the maximum frequency of modulating signal.
2. Explain AM modulator circuits.
BJT Collector Modulator
v The modulated output can be obtained by making the voltage on output electrode to vary according to input modulating signal.
v The transistor is biased well beyond cutoff so that it operates in class C mode. The class C mode is used because of its high efficiency. The RF drive is a carrier signal used for AM.
v This carrier amplitude is such that it drives transistor in conduction over part of its cycle. It is applied to the base of transistor. The modulating signal is passed through the power amplifier and applied to the collector through a low frequency transformer.
v This voltage is shown as Vm (t) in figure. This modulating voltage is In series with the supply voltage Hence the collector voltage becomes
3. Draw the block diagram of AM Transmitter and explain each block briefly.
v The block diagram of AM transmitter. The crystal oscillator generates carrier frequency. The buffer amplifier and driver amplifier amplify the power level of the carrier to required value. The amplified carrier is given to class C
4. Derive the expression for AM & its Power and Efficiency calculation:
5. Explain in detail about DSB SC-AM with frequency spectrum.
UNIT – II, 2 MARKS
PART-A
1. What do you mean by narrowband FM?
When the modulation index is less than 1, the angle modulated systems are called low index. The bandwidth requirement of low index systems is approximately twice of the modulating signal frequency. Therefore low index systems are called narrowband FM
2. Why Armstrong method of FM is superior to reactance modulator?
Reactance modulator is direct FM, where as Armstrong method is indirect FM. Armstrong method generates FM from PM. Hence crystal oscillators can be used in Armstrong method. Therefore frequency stability is better than reactance modulator.
3. What are the advantages of FM over AM?
FM has following advantages over AM.
v The amplitude of FM is constant. It is independent of depth of modulation. Hence transmitter power remains constant in FM whereas it varies in AM.
v Since amplitude of FM constant, the noise interference is minimum in FM. Any noise superimposing amplitude can be removed with the help of amplitude limits. Whereas it is difficult to remove amplitude variations due to noise in AM.
v The depths of modulation have limitation in AM. But in FM the depth of modulation can be increased to any value by increasing the deviation. This does not cause any distortion in FM signal.
v Since guard bands are provided in FM, there is less possibility of adjacent
channel interference.
4. Define PM.
Phase modulation is defined as the process of changing the phase of the carrier signal in accordance with the instantaneous amplitude of the message signal.
5. Differentiate between narrowband fm and wideband FM
6. What is mean by indirect method of FM generation?
In this type of angle modulation, FM is obtained by phase modulation of the carrier. This means, an instantaneous phase of the carrier directly proportional to amplitude of the modulating signal
7. What is modulation index of FM?
Modulation index is defined as the ratio of maximum frequency deviation to the
Modulating frequency.
8. State Carson’s rule of FM bandwidth.
Carson rule states that the bandwidth required transmitting an angle Modulated wave as twice the sum of the peak frequency deviation and the highest Modulating signal frequency. Mathematically Carson’s rule is
B=2(f +fm) Hz.
9. What is the advantage of Fm?
v Lesser distortion. Frequency modulated wave is less susceptible to interferences from buildings, traffic etc which provides improved signal to noise ratio (about 25dB) w.r.t. to man made interference.
v Waves at higher frequencies can carry more data than the waves at low frequency.
v Smaller geographical interference between neighboring stations.
v Less radiated power.
v Well defined service areas for given transmitter power.
1. Draw the circuit diagram of FET reactance modulator and explain its operation.
v Reactance-Tube Modulation. - In direct modulation, an oscillator is frequency modulated by a REACTANCE TUBE that is in parallel (SHUNT) with the oscillator tank circuit. (The terms "shunt" or "shunting" will be used in this module to mean the same as "parallel" or "to place in parallel with" components.) This is illustrated in figure. The oscillator is a conventional Hartley circuit with the reactance-tube circuit in parallel with the tank circuit of the oscillator tube.
v The reactance tube is an ordinary pentode. It is made to act either capacitively or inductively; that is, its grid is excited with a voltage which either leads or lags the oscillator voltage by 90 degrees.
Figure. Reactance-tube FM modulator.
v When the reactance tube is connected across the tank circuit with no modulating voltage applied, it will affect the frequency of the oscillator. The voltage across the oscillator tank circuit (L1 and C1) is also in parallel with the series network of R1 and C7. This voltage causes a current flow through R1 and C7. If R1 is at least five times larger than the capacitive reactance of C7, this branch of the circuit will be essentially resistive. Voltage E1, which is across C7, will lag current by 90 degrees. E1 is applied to the control grid of reactance tube V1. This changes plate current (Ip), which essentially flows only through the LC tank circuit. This is because the value of R1 is high compared to the impedance of the tank circuit. Since current is inversely proportional to impedance, most of the plate current coupled through C3 flows through the tank circuit.
v At resonance, the voltage and current in the tank circuit are in phase. Because E1 lags E by 90 degrees and I p is in phase with grid voltage E1, the superimposed current through the tank circuit lags the original tank current by 90 degrees. Both the resultant current (caused by Ip) and the tank current lag tank voltage and current by some angle depending on the relative amplitudes of the two currents. Because this resultant current is a lagging current, the impedance across the tank circuit cannot be at its maximum unless something happens within the tank to bring current and voltage into phase. Therefore, this situation continues until the frequency of oscillations in the tank circuit changes sufficiently so that the voltages across the tank and the current flowing into it are again in phase.
v This action is the same as would be produced by adding a reactance in parallel with the L1C1 tank. Because the superimposed current lags voltage E by 90 degrees, the introduced reactance is inductive. In NEETS, Module 2, Introduction to Alternating Current and Transformers, you learned that total inductance decreases as additional inductors are added in parallel. Because this introduced reactance effectively reduces inductance, the frequency of the oscillator increases to a new fixed value.
v Now let's see what happens when a modulating signal is applied. The magnitude of the introduced reactance is determined by the magnitude of the superimposed current through the tank. The magnitude of Ip for a given E1 is determined by the transconductance of V1. (Transconductance was covered in NEETS, Module 6, Introduction to Electronic Emission, Tubes, and Power Supplies.)
v Therefore, the value of reactance introduced into the tuned circuit varies directly with the transconductance of the reactance tube. When a modulating signal is applied to the grid of V1, both E1 and I p change, causing transconductance to vary with the modulating signal. This causes a variable reactance to be introduced into the tuned circuit. This variable reactance either adds to or subtracts from the fixed value of reactance that is introduced in the absence of the modulating signal. This action varies the reactance across the oscillator which, in turn, varies the instantaneous frequency of the oscillator. These variations in the oscillator frequency are proportional to the instantaneous amplitude of the modulating voltage. Reactance-tube modulators are usually operated at low power levels. The required output power is developed in power amplifier stages that follow the modulators.
v The output of a reactance-tube modulated oscillator also contains some unwanted amplitude modulation. This unwanted modulation is caused by stray capacitance and the resistive component of the RC phase splitter. The resistance is much less significant than the desired XC, but the resistance does allow some plate current to flow which is not of the proper phase relationship for good tube operation. The small amplitude modulation that this produces is easily removed by passing the oscillator output through a limiter-amplifier circuit.
2. Explain the direct and indirect method of FM transmitters.
v If a crystal oscillator is used to provide the carrier signal, the frequency cannot be varied too much (this is a characteristic of crystal oscillators). Thus, crystal oscillators cannot be used in broadcast FM, but other oscillators can suffer from frequency drift. An automatic frequency control (AFC) circuit is used in conjunction with a non-crystal oscillator to ensure that the frequency drift is minimal.
v Figure 4 shows a Crosby direct FM transmitter which contains an AFC loop. The frequency modulator shown can be a VCO since the oscillator frequency as much lower than the actual transmission frequency. In this example, the oscillator centre frequency is 5.1MHz which is multiplied by 18 before transmission to give ft = 91.8MHz.
When the frequency is multiplied, so are the frequency and phase deviations. However, the modulating input frequency is obviously unchanged, so the modulation index is multiplied by 18. The maximum frequency deviation at the output is 75kHz, so the maximum allowed deviation at the modulator output is
Since the maximum input frequency is fm = 15kHz for broadcast FM, the modulation index must be
o
The modulation index at the antenna then is = 0.2778 x 18 = 5.The AFC loop aims to increase the stability of the output without using a crystal oscillator in the modulator.
v The modulated carrier signal is mixed with a crystal reference signal in a non-linear device. The band-pass filter provides the difference in frequency between the master oscillator and the crystal oscillator and this signal is fed into the frequency discriminator. The frequency discriminator produces a voltage proportional to the difference between the input frequency and its resonant frequency. Its resonant frequency is 2MHz, which will allow it to detect low frequency variations in the carrier.
v The output voltage of the frequency discriminator is added to the modulating input to correct for frequency deviations at the output. The low-pass filter ensures that the frequency discriminator does not correspond to the frequency deviation in the FM signal (thereby preventing the modulating input from being completely cancelled).
v Indirect transmitters have no need for an AFC circuit because the frequency of the crystal is not directly varied. This means that indirect transmitters provide a very stable output, since the crystal frequency does not vary with operating conditions.
v Figure 5 shows the block diagram for an Armstrong indirect FM transmitter. This works by using a suppressed carrier amplitude modulator and adding a phase shifted carrier to this signal. The effect of this is shown in figure 6, where the pink signal is the output and the blue signal the AM input. The output experiences both phase and amplitude modulation. The amplitude modulation can be reduced by using a carrier much larger than the peak signal amplitude, as shown in figure 7. However, this reduces the amount of phase variation.
v The disadvantage of this method is the limited phase shift it can provide. The rest of figure 5 shows the frequency shifting to the FM broadcast band by means of frequency multiplication (by a factor of 72), frequency shifting and frequency multiplication again. This also multiplies the amount of phase shift at the antenna, allowing the required phase shift to be produced by a small phase variation at the modulator output.
3. Explain how FM can be generated from PM.
4. Draw the block diagram of FM demodulator using PLL and explain.
Phase locked loop (PLL)
v Another popular form of FM demodulator comes in the form of a phase locked loop. Like the quadrature detector, phase locked loops do not need to use a coil, and therefore they make a very cost effective form of demodulator.
v The way in which they operate is very simple. The loop consists of a phase detector into which the incoming signal is passed, along with the output from the voltage controlled oscillator (VCO) contained within the phase locked loop. The output from the phase detector is passed into a loop filter and then sued as the control voltage for the VCO.
v
Phase locked loop (PLL) FM demodulator
v With no modulation applied and the carrier in the centre position of the pass-band the voltage on the tune line to the VCO is set to the mid position. However if the carrier deviates in frequency, the loop will try to keep the loop in lock. For this to happen the VCO frequency must follow the incoming signal, and for this to occur the tune line voltage must vary. Monitoring the tune line shows that the variations in voltage correspond to the modulation applied to the signal. By amplifying the variations in voltage on the tune line it is possible to generate the demodulated signal.
v It is found that the linearity of this type of detector is governed by the voltage to frequency characteristic of the VCO. As it normally only swings over a small portion of its bandwidth, and the characteristic can be made relatively linear, the distortion levels from phase locked loop demodulators are normally very low.
5. Draw the frequency spectrum of an FM waveform for tone modulation and discuss how to arrive at the transmission bandwidth.
Since angle modulation is a nonlinear process, an exact description of the spectrum of an angle-modulated signal for an arbitrary message signal is more complicated than linear process. However if s(t) is sinusoidal, then the instantaneous phase deviation of the angle-modulated signal is sinusoidal and the spectrum can be relatively easy to obtained. If we
BAND WIDTH OF FM SIGNALS
UNIT-III, 2 Marks
1. What are the types of FM detectors?
Slope detector and phase discriminator
2. What are the types of phase discriminator?
Foster seely discriminator and ratio detector.
3. What are the disadvantages of balanced slope detector?
1. Amplitude limiting cannot be provided
2. Linearity is not sufficient
3. It is difficult to align because of three different frequencies to which various tuned circuits to be tuned.
4. The tuned circuit is not purely band limited.
4. What is TRF receiver?
Tuned Radio Frequency is also called straight receiver.Here the receiver operates in straight forward manner without frequency conversion
5. What are the advantages of superheterodyne receiver over TRF?
The advantages of superheterodyne receiver over TRF are high selectivity improved sensitivity throughout the carrier frequency band..It eliminates image frequency.
6. What are the characteristics of a receiver?
Sensitivity, selectivity, Fidelity.
7. Define sensitivity.
Sensitivity is a measure of receiver’s ability to re- produce very weak signals. The weaker the signal that can be applied and still produce a certain signal-to- noise (S/N) ratio, the better that receiver’s sensitivity rating. Usually, sensitivity is specified as the signal strength in microvolts necessary to cause a S/N ratio of 10 decibels, or 3.16:1
8. Define selectivity.
Selectivity of a receiver is defined as its ability to select the desired signals among the various signals.
9. Define super heterodyne principle.
It can be defined as the process of operation of modulated waves to obtain similarly modulated waves of different frequency. This process uses a locally generated carrier wave, which determines the change of frequency.
10. What are synchronous detectors?
The synchronous or coherent detector uses exact carrier synchronization for retrieving the message signal from the modulated signal. These detectors are mainly used for detecting DSB-SC or SSB-SC signals because of their complicated nature.
PART-B
1. Draw the block diagram of superhetrodyne receiver and explain its operation
v The superheterodyne is the type receiver most familiar to you. You probably see one daily in your home in the form of an AM and/or fm radio. We will discuss the basic workings of both AM and fm types and their differences. Amplitude Modulation Receiver Figure shows a block diagram with waveforms of a typical AM superheterodyne receiver developed to overcome the disadvantages of earlier type receivers. Let’s assume you are tuning the receiver. When doing this you are actually changing the frequency to which the RF amplifier is tuned. The RF carrier comes in from the antenna and is applied to the RF amplifier. The output of the amplifier is an amplified carrier and is sent to the mixer.
v The mixer also receives an input from the local oscillator. These two signals are beat together to obtain the IF through the process of heterodyning. (Heterodyning will be further discussed later in this chapter and was covered in NEETS, Module 12, Modulation Principles.) At this time you should note the dotted lines connecting the local oscillator, rf amplifier, and the mixer. This is used on block diagrams and schematics to indicate GANGED TUNING. Ganged tuning is the process used to tune two or more circuits with a single control. In our example, when you change the frequency of the receiver all three stages change by the same amount. There is a fixed difference in frequency between the local oscillator and the rf amplifier at all times. This difference in frequency is the IF. This fixed difference and ganged tuning ensures a constant IF over the frequency range of the receiver
The IF carrier is applied to the IF amplifier.
v The amplified IF carrier is then sent to the detector. The output of the detector is the audio component of the input signal. This audio component is then passed through an audio frequency amplifier. The amplified audio component is sent to a speaker for reproduction. This allows you to hear the signal. You should note that a superheterodyne receiver may have more than one frequency-converting stage and as many amplifiers as needed to obtain the desired power output. (Additional amplifiers are not shown.) HETERODYNING.As you know the intermediate frequency is developed by a process called heterodyning. This action takes place in the mixer stage (sometimes called a converter or first detector). Heterodyning is the combining of the incoming signal with the local oscillator signal.
v When heterodyning the incoming signal and the local oscillator signal in the mixer stage, four frequencies are produced. They are the two basic input frequencies and the sum and the difference of those two frequencies. The amplifier that follows (IF amplifier) will be tuned to the difference frequency. This difference frequency is known as the intermediate frequency (IF). A typical value of IF for an AM communications receiver is 455 kilohertz. The difference frequency is a lower frequency than either the rf input or oscillator frequencies. This lower frequency gives slightly better gain but does increase the chances of image frequency interference. Image frequencies will be discussed later in this chapter.
v DETECTION:Once the IF stages have amplified the intermediate frequency to a sufficient level, it is fed to the detector. When the mixer is referred to as the first detector, this stage would be called the second detector. The detector extracts the modulating audio signal. The detector stage consists of a rectifying device and filter, which respond only to the amplitude variations of the IF signal. This develops an output voltage varying at an audio-frequency rate. The output from the detector is further amplified in the audio amplifier and is used to drive a speaker or earphones.
2. Draw the Circuit diagram and explain the slope detector.
v SLOPE DETECTION To be able to understand the principles of operation for fm detectors, you need to first study the simplest form of frequency-modulation detector, the SLOPE DETECTOR. The slope detector is essentially a tank circuit which is tuned to a frequency either slightly above or below the fm carrier frequency. View (A) of figure is a plot of voltage versus frequency for a tank circuit. The resonant frequency of the tank is the frequency at point 4. Components are selected so that the resonant frequency is higher than the frequency of the fm carrier signal at point 2.
v The entire frequency deviation for the fm signal falls on the lower slope of the bandpass curve between points 1 and 3. As the fm signal is applied to the tank circuit in view (B), the output amplitude of the signal varies as its frequency swings closer to, or further from, the resonant frequency of the tank. Frequency variations will still be present in this waveform, but it will also develop amplitude variations, as shown in view (B). This is because of the response of the tank circuit as it varies with the input frequency. This signal is then applied to the diode detector in view (C) and the detected waveform is the output. T
Slope detector. DIODE DETECTOR
v This circuit has the major disadvantage that any amplitude variations in the RF waveform will pass through the tank circuit and be detected. This disadvantage can be eliminated by placing a limiter circuit before the tank input. (Limiter circuits were discussed in NEETS, Module 9, Introduction to Wave-Generation and Wave-Shaping Circuits.) This circuit is basically the same as an AM detector with the tank tuned to a higher or lower frequency than the received carrier.
Figure A.Slope detector. VOLTAGE VERSUS FREQUENCY PLOT.
Figure B.—Slope detector. TANK CIRCUIT.
3. Explain the foster seely discriminator with suitable circuit diagram.
v When circuits employing discrete components were more widely sued, the Ratio and Foster-Seeley detectors were widely used. Of these the ratio detector was the most popular as it offers a better level of amplitude modulation rejection of amplitude modulation. This enables it to provide a greater level of noise immunity as most noise is amplitude noise, and it also enables the circuit to operate satisfactorily with lower levels of limiting in the preceding IF stages of the receiver.
v The Foster Seeley detector has many similarities to the ratio detector. The circuit topology looks very similar, having a transformer and a pair of diodes, but there is no third winding and instead a choke is used.
The Foster-Seeley detector
v Like the ratio detector, the Foster-Seeley circuit operates using a phase difference between signals. To obtain the different phased signals a connection is made to the primary side of the transformer using a capacitor, and this is taken to the centre tap of the transformer. This gives a signal that is 90 degrees out of phase.
v When an un-modulated carrier is applied at the centre frequency, both diodes conduct, to produce equal and opposite voltages across their respective load resistors. These voltages cancel each one another out at the output so that no voltage is present. As the carrier moves off to one side of the centre frequency the balance condition is destroyed, and one diode conducts more than the other. This results in the voltage across one of the resistors being larger than the other, and a resulting voltage at the output corresponding to the modulation on the incoming signal.
v The choke is required in the circuit to ensure that no RF signals appear at the output. The capacitors C1 and C2 provide a similar filtering function.
v Both the ratio and Foster-Seeley detectors are expensive to manufacture. Wound components like coils are not easy to produce to the required specification and therefore they are comparatively costly. Accordingly these circuits are rarely used in modern equipment.
4.Explain envelope detector with necessary diagrams.
4. Explain the following performance of radio receiver.
(1)selectivity
(2)sensitivity
(3)fidelity
(4)Image frequency rejection
v Understanding receiver characteristics is manda- tory in determining operational condition and for com- paring receivers. Important receiver characteristics are sensitivity, noise, selectivity, and fidelity
o Sensitivity
v Sensitivity is a measure of receiver’s ability to re- produce very weak signals. The weaker the signal that can be applied and still produce a certain signal-to- noise (S/N) ratio, the better that receiver’s sensitivity rating. Usually, sensitivity is specified as the signal strength in microvolts necessary to cause a S/N ratio of 10 decibels, or 3.16:1
o Noise
v All receivers generate noise. Noise is the limiting factor on the minimum usable signal that the receiver can process and still produce a usable output. Ex- pressed in decibels, it is an indication of the degree to which a circuit deviates from the ideal; a noise figure of 0 decibels is ideal.
o Selectivity
v Selectivity is the ability of a receiver to distinguish between a signal at the desired frequency and signals at adjacent frequencies. The better the receiver’s ability to exclude unwanted signals, the better its selectivity. The degree of selectivity is determined by the sharp- ness of resonance to which the frequency determining components (bandpass filters) have been engineered and tuned. Measurement of selectivity is usually done by taking a series of sensitivity readings in which the input signal is stepped along a band of frequencies above and below resonance of the receiver’s circuits. As the frequency to which the receiver is tuned is approached, the input level required to maintain a given output will fall. As the tuned frequency is passed, the input level will rise. Input levels are then plotted against frequency. The steepness of the curve at the tuned frequency indicates the selectivity of the re ceiver.
o Fidelity
v Fidelity is a receiver’s ability to reproduce the in- put signal accurately. Generally, the broader the bandpass, the greater the fidelity. Measurement is taken by modulating an input frequency with a series of audio frequencies and then plotting the output measurements at each step against the audio input. The curve will show the limits of reproduction. Good selectivity requires a narrow bandpass. Good fidelity requires a wider bandpass to amplify the outer- most frequencies of the sidebands. Knowing this, you can see that most receivers are a compromise between good selectivity and high fidelity.
o image frequency.
v An image frequency is any frequency other than the selected radio frequency carrier that ,if allowed to enter a receiver and mix with the local oscillator ,will produce a cross product frequency that is equal to the intermediate frequency.
v The image frequency rejection ratio is the measure of the ability of preselector to reject the image frequency.
Mathematically, IFRR is
IFRR =(1+Q2r2)1/2
Where r= (fim/fRF)-(fRF/fim)
UNIT – IV, 2 MARKS
PART-A
1. Define multiplexing.
Multiplexing is defined as the process of transmitting several message signals simultaneously over a single channel.
2. Difference between TDM and TDMA
By TDM, the signals at one earth station are multiplexed into single channel. Such multiple channels from different earth stations share a satellite transponder aith the help of TDMA.
3. Define PM
Phase modulation is defined as the process of changing the phase of the carrier signal in accordance with the instantaneous amplitude of the message signal
4. What is meant by pulse modulation & name the methods of it?
Pulse modulation includes many different methods of converting information into pulse form for transferring pulses from a source to a destination.
Methods:
- Pulse Width Modulation
- Pulse Position Modulation
- Pulse Amplitude Modulation
- Pulse code Modulation
Methods:
- Pulse Width Modulation
- Pulse Position Modulation
- Pulse Amplitude Modulation
- Pulse code Modulation
5. Define Pulse Width Modulation (PWM)
The pulse width (active portion of the duty cycle) is proportional to the amplitude of the analog signal. This method is sometimes called as pulse duration modulation (PDM) or pulse length modulation (PLM).
6. Define Pulse Position Modulation (PPM)
The position of a constant width pulse within a prescribed time slot is varied according to the amplitude of the analog signal is called PPM.
7. Pulse Amplitude Modulation (PAM).
The amplitude of a constant width, constant position pulse is varied according to the amplitude of the analog signal.
8. Define Pulse code Modulation (PCM).
The analog signal is sampled and converted to a fixed length, serial binary number for transmission. The binary number varies according to the amplitude of the analog signal.
9. State sampling theorem.
A band limited signal of finite energy, which has no frequency components higher than ‘W’ Hz can be completed in its samples and recovered back if the sampling frequency
10. Define Nyquist rate
When the sampling rate becomes exactly equal to 2W samples per second, for a signal bandwidth of W Hertz , then it is called Nyquist Rate.
PART-B
1. Briefly explain about pulse amplitude modulation.
v In communications PULSE-MODULATION SYSTEMS, the modulating wave must be SAMPLED at 2.5 times the highest modulating frequency to ensure accuracy. PULSE-AMPLITUDE MODULATION (PAM) is modulation in which the amplitude of each pulse is controlled by the instantaneous amplitude of the modulation signal at the time of each pulse
v To transmit intelligence using pulse modulation, you must provide a method to vary some characteristic of the pulse train in accordance with the modulating signal. Figure 2-40 illustrates a simple pulse train. The characteristics of these pulses that can be varied are amplitude, pulse width, pulse-repetition time, and the pulse position as compared to a reference. In addition to these three characteristics, pulses may be transmitted according to a code to represent the different levels of the modulating signal.
v To ensure maximum fidelity (accuracy in reproducing a modulating wave), the modulating signal has to be represented by enough pulses to restore the original wave shape. Logically, the higher the sampling rate (the more often sampled) of the pulse modulator, the more accurately the original modulating wave can be reproduced. Figure 2-41 illustrates the effectiveness of three pulse-sampling rates. View (A) shows a sampling rate of more than two times the modulating frequency. As you can see, this reproduces the modulating signal very accurately. However, the high sampling rate requires a wide bandwidth and increases the average power required of the transmitter. If less than two samples per cycle are made, you are not able to reproduce the original modulating signal, as shown in view (B). View (C) shows a sampling rate that is two times the highest modulating frequency. This is the minimum sampling rate that will give a sufficiently accurate reproduction of the modulating wave. The standard sampling rate is 2.5 times the highest frequency that is to be transmitted.
v This ensures the ability to accurately reproduce the modulating waveform. In military voice systems the bandwidth for voice signals is limited to 300 to 3,000 hertz, requiring a sampling frequency of 8 kilohertz. Although the pulse characteristic that is changed may vary for each type of pulse modulation, the sampling frequency will remain constant. We will now briefly discuss common types of pulse modulation.
Some characteristic of the sampling pulses must be varied by the modulating signal for the intelligence of the signal to be present in the pulsed wave. Figure 2-42 shows three typical waveforms in which the pulse amplitude is varied by the amplitude of the modulating signal. View (A) represents a sine wave of intelligence to be modulated on a transmitted carrier wave. View (B) shows the timing pulses which determine the sampling interval. View (C) shows PULSE-AMPLITUDE MODULATION (PAM) in which the amplitude of each pulse is controlled by the instantaneous amplitude of the modulating signal at the time of each pulse. Figure 2-42A.—Pulse-amplitude modulation (PAM). MODULATION. Figure 2-42B.—Pulse-amplitude modulation (PAM). TIMING. Figure 2-42C.—Pulse-amplitude modulation (PAM). PAM. Pulse-amplitude modulation is the simplest form of pulse modulation. It is generated in much the same manner as analog-amplitude modulation.
v The timing pulses are applied to a pulse amplifier in which the gain is controlled by the modulating waveform. Since these variations in amplitude actually represent the signal, this type of modulation is basically a form of AM. The only difference is that the signal is now in the form of pulses. This means that pam has the same built-in weaknesses as any other AM signal - high susceptibility to noise and interference. The reason for susceptibility to noise is that any interference in the transmission path will either add to or subtract from any voltage already in the circuit (signal voltage).
v Thus, the amplitude of the signal will be changed. Since the amplitude of the voltage represents the signal, any unwanted change to the signal is considered a SIGNAL DISTORTION. For this reason, pam is not often used. When PAM is used, the pulse train is used to frequency modulate a carrier for transmission. Techniques of pulse modulation other than PAM have been developed to overcome problems of noise interference. The following sections will discuss other types of pulse modulation
2. Explain the working principle of TDM with its relevant diagram.
v It's often practical to combine a set of low-bit-rate streams, each with a fixed and pre-defined bit rate, into a single high-speed bit stream that can be transmitted over a single channel. This technique is called time division multiplexing (TDM) and has many applications, including wireline telephone systems and some cellular telephone systems. The main reason to use TDM is to take advantage of existing transmission lines. It would be very expensive if each low-bit-rate stream were assigned a costly physical channel (say, an entire fiber optic line) that extended over a long distance.
v Consider, for instance, a channel capable of transmitting 192 kbit/sec from Chicago to New York. Suppose that three sources, all located in Chicago, each have 64 kbit/sec of data that they want to transmit to individual users in New York. As shown in Figure 7-2, the high-bit-rate channel can be divided into a series of time slots, and the time slots can be alternately used by the three sources. The three sources are thus capable of transmitting all of their data across the single, shared channel. Clearly, at the other end of the channel (in this case, in New York), the process must be reversed (i.e., the system must divide the 192 kbit/sec multiplexed data stream back into the original three 64 kbit/sec data streams, which are then provided to three different users). This reverse process is called demultiplexing.
v Figure 7-2—Time division multiplexing.
Choosing the proper size for the time slots involves a trade-off between efficiency and delay. If the time slots are too small (say, one bit long) then the multiplexer must be fast enough and powerful enough to be constantly switching between sources (and the demultiplexer must be fast enough and powerful enough to be constantly switching between users). If the time slots are larger than one bit, data from each source must be stored (buffered) while other sources are using the channel. This storage will produce delay. If the time slots are too large, then a significant delay will be introduced between each source and its user. Some applications, such as teleconferencing and videoconferencing, cannot tolerate long delays. As shown in Example 7-2, the sources that are multiplexed may have different bit rates. When this occurs, each source is assigned a number of time slots in proportion to its transmission rate.
Example 7.1—The T1 system for wireline telephone networks
v The T1 system is used for wireline long-distance service in North America and is an excellent example of TDM. Speech from a telephone conversation is sampled once every 125 msec and each sample is converted into eight bits of digital data (see Chapter 8 for more details). Using this technique, a transmission speed of 64,000 bits/sec is required to transmit the speech. A T1 line is essentially a channel capable of transmitting at a speed of 1.544 Mbit/sec. This is a much higher transmission speed than a single telephone conversation needs, so TDM is used to allow a single T1 line to carry 24 different speech signals between, say, two different telephone substations (called central offices) within a city. As shown in Figure 7-3, the 1.544 Mbit/sec bit stream is divided into 193-bit frames. The duration of each frame is
v
corresponding to the period between samples of the speech. Each frame is divided into 24 slots, which are each eight bits wide (corresponding to the number of bits needed to digitize a speech sample). One additional bit at the end of the frame is used for signaling. The eight bits of data corresponding to a sample of the speech are placed into one of the 24 slots in the frame.
v For longer distances (say, between two large cities) higher-capacity channels are used and multiple T1 lines are time division multiplexed onto the new channels. A T3 channel for example, has a transmission speed of 44.736 Mbit/sec and uses TDM to carry 28 T1 lines (a total of 672 different speech signals) plus signaling. For more information on this hierarchical multiplexing system, see BeIlamy [7.1].
v
v Figure 7-3—Time division multiplexing on a T1 line.
Example 7.2—TDM with sources having different data rates
v Consider the case of three streams with bit rates of 8 kbit/sec,16 kbit/sec, and 24 kbit/sec, respectively. We want to combine these streams into a single high-speed stream using TDM. The high-speed stream in this case must have a transmission rate of 48 kbit/sec, which is the sum of the bit rates of the three sources. To determine the number of time slots to be assigned to each source in the multiplexing process. we must reduce the ratio of the rates, 8:16:24, to the lowest possible form, which in this case is 1:2:3. The sum of the reduced ratio is 6, which will then represent the minimum length of the repetitive cycle of slot assignments in the multiplexing process.
v The solution is now readily obtained: In each cycle of six time slots we assign one slot to Source A (8 kbit/sec), two slots to Source B (16 kbit/sec), and three slots to Source: C (24 kbit/sec). Figure 7-4 illustrates this assignment, using “a” to indicate data from Source A, “b” to indicate data from Source B, and “c” to indicate data from Source C.
v
v Figure 7-4—Multiplexing input lines with different transmission speeds.
Example 7.3—A more complex TDM system
v Consider a system with four low-bit-rate sources of 10 kbit/sec, 15 kbit/sec, 20 kbit/sec, and 30 kbit/sec. determine the slot assignments when the data streams are combined using TDM.
Solution
v The rate ratio 10:15:20:30 reduces to 2:3:4:6. The length of the cycle is therefore 2 + 3 + 4 + 6 = 15 slots. Within each cycle of 15 slots, we assign two slots to the 10 kbit/sec source, three slots to the 15 kbit/sec source, four slots to the 20 kbit/sec source, and six slots to the 30 kbit/sec source.
v So far we have considered a form of TDM that is based on fixed slot assignments to each of the low-bit-rate data streams. In other words, each stream has predefined slot positions in the combined stream, and the receiver must be aware which slots belong to which input stream. Both transmission ends, the transmitter and the receiver, must be perfectly synchronized to the slot period. For this reason, the technique is usually called synchronous TDM.
v There is another important version of TDM, usually referred to as statistical TDM. Statistical TDM is useful for applications in which the low-bit-rate streams have speeds that vary in time. For example, a low-bit-rate stream to a single terminal in a computer network may fluctuate between 2 kbit/sec and 50 kbit/sec during an active connection session (we've all seen variable speeds during Internet connections, for instance). If we assign the stream enough slots for its peak rate (that is, for 50 kbit/sec), then we will be wasting slots when the rate drops well below the peak value. This waste can be especially significant if the system has many variable-speed low-bit-rate streams.
v Statistical TDM works by calculating the average transmission rates of the streams to be combined, and then uses a high-speed multiplexing link with a transmission rate that is equal to (or slightly greater than) the statistical average of the combined streams. Since the transmission rates from each source are variable, we no longer assign a fixed number of time slots to each data stream. Rather, we dynamically assign the appropriate number of slots to accommodate the current transmission rates from each stream. Because the combined rate of all the streams will also fluctuate in time between two extreme values, we need to buffer the output of the low-bit-rate streams when the combined rate exceeds the transmission rate of the high-speed link.
v With statistical TDM, we are no longer relying on synchronized time slots with fixed assignments for each input stream, as we did with synchronous TDM. So how does the demultiplexer in statistical TDM know which of the received bits belongs to which data stream? Prior to transmission, we divide each stream of bits coming from a source into fixed-size blocks. We then add a small group of bits called a header to each block, with the header containing the addresses of the source and intended user for that block. The block and the header are then transmitted together across the channel. Combined, the block and header are called a packet.
v Actually, the header may contain other information besides the source and user addresses, such as extra bits for error control (see Chapter 10) or additional bits for link control (used, for example, to indicate the position of a particular block in a sequence of blocks coming from the same user, or to indicate priority level for a particular message). Extra bits can also be added to the beginning and end of a block for synchronization; a particular pattern of bits, called a start flag, can be used in the header to mark the start of a block, and another particular pattern of bits, called an end flag, can be used to conclude the block. Each block transmitted across the channel thus contains a group of information bits that the user wants, plus additional bits needed by the system to ensure proper transmission. These additional bits, while necessary to system operation, reduce the effective transmission rate on the channel. Figures 7-5 and 7-6 present the statistical TDM technique and the structure of a typical packet.
v
v Figure 7-5—Statistical TDM.
v Figure 7-6—Structure of a typical statistical TDM packet
3. Explain the working principle of FDM with its relevant diagram.
v In many communication systems, a single, large frequency band is assigned to the system and is shared among a group of users. Examples of this type of system include:
v A microwave transmission line connecting two sites over a long distance. Each site has a number of sources generating independent data streams that are transmitted simultaneously over the microwave link. | |
v AM or FM radio broadcast bands, which are divided among many channels or stations. The stations are selected with the radio dial by tuning a variable-frequency filter. (We examined AM and FM in Chapter 6.) | |
v A satellite system providing communication between a large number of ground stations that are separated geographically but that need to communicate at the same time. The total bandwidth assigned to the satellite system must be divided among the ground stations. | |
v A cellular radio system that operates in full-duplex mode over a given frequency band. The earlier cellular telephone systems, for example AMPS, used analog communication methods. The bandwidth for these systems was divided into a large number of channels. Each pair of channels was assigned to two communicating end-users for full-duplex communications. |
v Frequency division multiplexing (FDM) means that the total bandwidth available to the system is divided into a series of no overlapping frequency sub-bands that are then assigned to each communicating source and user pair. Figures 7-7a and 7-7b show how this division is accomplished for a case of three sources at one end of a system that are communicating with three separate users at the other end.
v Note that each transmitter modulates its source's information into a signal that lies in a different frequency sub-band (Transmitter 1 generates a signal in the frequency sub-band between 92.0 MHz and 92.2 MHz, Transmitter 2 generates a signal in the sub-band between 92.2 MHz and 92.4 MHz, and Transmitter 3 generates a signal in the sub-band between 92.4 MHz and 92.6 MHz). The signals are then transmitted across a common channel.
Figure 7-7a—A system using frequency division multiplexing.
Figure 7-7b—Spectral occupancy of signals in an FDM system.
v At the receiving end of the system, band pass filters are used to pass the desired signal (the signal lying in the appropriate frequency sub-band) to the appropriate user and to block all the unwanted signals. To ensure that the transmitted signals do not stray outside their assigned sub-bands, it is also common to place appropriate pass band filters at the output stage of each transmitter.
v It is also appropriate to design an FDM system so that the bandwidth allocated to each sub-band is slightly larger than the bandwidth needed by each source. This extra bandwidth, called a guard band, allows systems to use less expensive filters (i.e., filters with fewer poles and therefore less steep rolloffs).
v FDM has both advantages and disadvantages relative to TDM. The main advantage is that unlike TDM, FDM is not sensitive to propagation delays. Channel equalization techniques needed for FDM systems are therefore not as complex as those for TDM systems.
v Disadvantages of FDM include the need for band pass filters, which are relatively expensive and complicated to construct and design (remember that these filters are usually used in the transmitters as well as the receivers). TDM, on the other hand, uses relatively simple and less costly digital logic circuits.
v Another disadvantage of FDM is that in many practical communication systems, the power amplifier in the transmitter has nonlinear characteristics (linear amplifiers are more complex to build), and nonlinear amplification leads to the creation of out-of-band spectral components that may interfere with other FDM channels. Thus, it is necessary to use more complex linear amplifiers in FDM systems.
4. Write short notes on
i) Sampling theorem
v Digital transmission of information and digital signal processing all require signals to first be "acquired" by a computer. One of the most amazing and useful results in electrical engineering is that signals can be converted from a function of time into a sequence of numbers without error: We can convert the numbers back into the signal with (theoretically) no error. Harold Nyquist, a Bell Laboratories engineer, first derived this result, known as the Sampling Theorem, in the 1920s. It found no real application back then. Claude Shannon, also at Bell Laboratories, revived the result once computers were made public after World War II.
v The theorem is commonly called the Shannon sampling theorem, and is also known as Nyquist–Shannon–Kotelnikov, Whittaker–Shannon–Kotelnikov, Whittaker–Nyquist–Kotelnikov–Shannon, WKS, etc., sampling theorem, as well as the Cardinal Theorem of Interpolation Theory. It is often referred to as simply the sampling theorem.
v In essence, the theorem shows that a band limited analog signal that has been sampled can be perfectly reconstructed from an infinite sequence of samples if the sampling rate exceeds 2B samples per second, where B is the highest frequency in the original signal. If a signal contains a component at exactly B hertz, then samples spaced at exactly 1/(2B) seconds do not completely determine the signal, Shannon's statement notwithstanding. This sufficient condition can be weakened, as discussed at Sampling of non-baseband signals below.
v More recent statements of the theorem are sometimes careful to exclude the equality condition; that is, the condition is if x(t) contains no frequencies higher than or equal to B; this condition is equivalent to Shannon's except when the function includes a steady sinusoidal component at exactly frequency B.
v The theorem assumes an idealization of any real-world situation, as it only applies to signals that are sampled for infinite time; any time-limited x(t) cannot be perfectly band limited. Perfect reconstruction is mathematically possible for the idealized model but only an approximation for real-world signals and sampling techniques, albeit in practice often a very good one.
v The theorem also leads to a formula for reconstruction of the original signal. The constructive proof of the theorem leads to an understanding of the aliasing that can occur when a sampling system does not satisfy the conditions of the theorem.
v The sampling theorem provides a sufficient condition, but not a necessary one, for perfect reconstruction. The field of compressed sensing provides a stricter sampling condition when the underlying signal is known to be sparse. Compressed sensing specifically yields a sub-Nyquist sampling criterion.
v A signal or function is bandlimited if it contains no energy at frequencies higher than some bandlimit or bandwidth B. A signal that is bandlimited is constrained in how rapidly it changes in time, and therefore how much detail it can convey in an interval of time. The sampling theorem asserts that the uniformly spaced discrete samples are a complete representation of the signal if this bandwidth is less than half the sampling rate.
The sampling process
v The theorem describes two processes in signal processing: a sampling process, in which a continuous time signal is converted to a discrete time signal, and a reconstruction process, in which the original continuous signal is recovered from the discrete time signal.
v The continuous signal varies over time (or space in a digitized image, or another independent variable in some other application) and the sampling process is performed by measuring the continuous signal's value every T units of time (or space), which is called the sampling interval. In practice, for signals that are a function of time, the sampling interval is typically quite small, on the order of milliseconds, microseconds, or less. This results in a sequence of numbers, called samples, to represent the original signal. Each sample value is associated with the instant in time when it was measured. The reciprocal of the sampling interval (1/T) is the sampling frequency denoted fs, which is measured in samples per unit of time. If T is expressed in seconds, then fs is expressed in Hz.
ii) Comparison of multiplexing
5. Explain about pulse position modulation with its relevant diagram
v Pulse-Time Modulation In pulse-modulated systems, as in an analog system, the intelligence may be impressed on the carrier by varying any of its characteristics. In the preceding paragraphs the method of modulating a pulse train by varying its amplitude was discussed.
v Time characteristics of pulses may also be modulated with intelligence information. Two time characteristics may be affected: (1) the time duration of the pulses, referred to as PULSE-DURATION MODULATION (PDM) or PULSE-WIDTH MODULATION (PWM); and (2) the time of occurrence of the pulses, referred to as PULSE-POSITION MODULATION (PPM), and a special type of PULSE-TIME MODULATION (PTM) referred to as PULSE-FREQUENCY MODULATION (PFM). Figure 2-43 shows these types of PTM in views (C), (D), and (E). Views (A) and (B) show the modulating signal and timing, respectively
Figure 2-43B.—Pulse-time modulation (ptm). TIMING
. Figure 2-43C.—Pulse-time modulation (ptm). PDM
. Figure 2-43D.—Pulse-time modulation (ptm). PPM.
UNIT –V, 2 MARKS
PART – A
1. Define noise.
Noise is defined as any unwanted form of energy, which tends to interfere with proper reception and reproduction of wanted signal.
2. Give the classification of noise.
Noise is broadly classified into two types. They are External noise and internal noise.
3. What are the types of External noise
External noise can be classified into
1. Atmospheric noise
2. Extraterrestrial noises
3. Man –made noises or industrial noises
4. What are types of internal noise
Internal noise can be classified into
1. Thermal noise
2. Shot noise
3. Transit time noise
4. Miscellaneous internal noise
5. Define flicker noise
Flicker noise is the one appearing in transistors operating at low audio Frequencies. Flicker noise is proportional to the emitter current and junction temperature and inversely proportional to the frequency.
6. Define signal to noise ratio.
Signal to noise ratio is the ratio of signal power to the noise power at the same point in a system.
7. Define noise figure
Noise figure is defined as
S/N = Signal power / Noise Power
8. Explain White Noise.
Many types of noise sources are Gaussian and have flat spectral density over a wide frequency range. Such spectrum has all frequency components in equal portion, and is therefore called white noise. The power spectral density of white noise is independent of the operating frequency.
9. Explain thermal noise
Thermal noise is the name given to the electrical noise arising from the random motion of electrons in a conductor.
10. What is narrowband noise?
The receiver of a communication system usually includes some provision for preprocessing the received signal. The preprocessing may take the form of a narrowband filter whose bandwidth is large enough to pass modulated component of the received
11. Give the representation of narrowband noise in terms of envelope and phase components.
12. Give the expression for noise voltage when several sources are cascaded.
Enr = Sqrt (4 KTB (R1 + R2 + …..) )
Where R1 , R2 --- are the resistances of the noise resistors.
K – Boltz man constant
T – absolute temperature
B – Bandwidth
PART-B
1. Obtain an expression for SNR of square law detector.
Square-low Detection
An example of simple nonlinear detector that we can calculate SNR and thus the “threshold region” can be more precisely determined.
= + -
= - . + àhigh freq
LPF
=
= +2 a + (t)+2 +2 a + + àremove dc
+2 +2 a + -
+ -
Postdection noise power:
Note =
(Assume the cross term are either zero or can be neglected)
Note: =3 if is Gaussian
This can be shown using moment generator = (t)
( )= =
Example: = à not a random signal
If we assume this part can be neglected to be discussed below
To examine ’(t) and
+2 (1+a ). + +
remove dcè =2 a +0.5
+2 (1+a ) + +
The term 0.5 is called harmonic distortion---not random noise
Power: = ( .) =4 . =2
Compare to baseband system:
Now =2 =2 where =2
(1) If >>1
(2) If <<1
(<<2 when is small)
This illustrates threshold effect.
Remarks:
1. The performance of envelope detector is better than that of squaer law detector for high SNR by 1.8dB.
2. DD/SD=a2/16
3. The BW of is wider than W and thus part or it can be filtered out by LPF (M.Schwartz,”Info .Trams.Modu.and Noise”pp.505~509)
à General form (assume . is small)
Replace with E: E=
=
= =.{ .( )/(1+ )}
where =2
IF >>
IF <<
Special case: =
2.Derive the noise performance of DSB-SC-AM
The receiver model for coherent detection of DSB-SC signals is shown in Fig. 7.4. The DSB-SC signal is, . We assume to be sample function of a WSS process with the power spectral density, , limited to Hz.
Fig. 7.4: Coherent Detection of DSB-SC.
The carrier, , which is independent of the message is actually a sample function of the process where is a random variable, uniformly distributed in the interval 0 to . With the random phase added to the carrier term, , the autocorrelation function of the process (of which is a sample function), is given by,
where is the autocorrelation function of the message process. Fourier transform of yields given by,
Let denote the message power, where
Then, .
That is, the average power of the modulated signal is . With the (two sided) noise power spectral density of , the average noise power in the message bandwidth is . Hence,
To arrive at the , we require . The input to the detector is , where is a narrowband noise quantity. Expressing in terms of its in-phase and quadrature components, we have
Assuming that the local oscillator output is , the output of the multiplier in the detector (Fig. 7.4) is given by
As the LPF rejects the spectral components centered around , we have
From Eq. 7.7, we observe that,
i) Signal and noise which are additive at the input to the detector are additive even at the output of the detector
ii) Coherent detector completely rejects the quadrature component .
iii) If the noise spectral density is flat at the detector input over the passband , then it is flat over the baseband , at the detector output. (Note that has a flat spectrum in the range to .)
As the message component at the output is , the average message power at the output is . As the spectral density of the in-phase noise component is for , the average noise power at the receiver output is . Therefore,
From Eq. 7.6 and 7.8, we obtain
3. Write shorts notes on
1. Internal noise.
Internal noise is that type of noise which is generated internally or with in the communication system or receiver.internal noise may be treated quantitatively and can also be reduced or minimized by proper system design.since internal noise is randomly distributed over the entire frequency spectrum.
2. External noise.
External noise may be defined as that type of noise which is generated external to a communication system i.e whose sources are external to the communication system. The effect of external noise,the only way is to shift the communication system to other place or location which has comparatively smaller external noise. Thus due to this reason, the satellite earth station are generally located in noise free valleys.
3. Noise temperature
In electronics, noise temperature is a temperature (in kelvin) assigned to a component such that the noise power delivered by the noisy component to a noiseless matched resistor is given by
PRL = kTsBn
in Watts, where:
· = noise temperature (K)
· = noise bandwidth (Hz)
Engineers often model noisy components as an ideal component in series with a noisy resistor. The source resistor is often assumed to be at room temperature, conventionally taken as 290 K (17 °C, 62 °F).[1]
4. Thermal noise.
Thermal noise is the name given to the electrical noise arising from the random motion of electrons in a conductor
5. Short noise.
Shot noise normally occurs when there is a potential barrier (voltage differential). PN junction diode is an example that has potential barrier. When the electrons and holes cross the barrier, shot noise is produced. For example, a diode, a transistor, and vacuum tube will all produce Shot noise. On the other hand, a resistor normally does not produce Shot noise since there is no potential barrier built within a resistor. Current flowing through a resistor will not exhibit any fluctuations. However, current flowing through a diode produces small fluctuations. This is due to electrons (in turn, the charge) arriving in quanta, one electron at a time. The current flow is not continuous, but limited by the quantum of the electron charges.
4. Explain the noise of cascaded amplifier and noise figure from equivalent noise résistance.
5. Derive the noise performance of phase modulation.
For PM, . For convenience, let . Then,
Again, we treat to be a sample function of a WSS process . Then,
output signal power = (7.24)
Let (7.25)
To calculate the output noise power, we require the power spectral density of . This is made somewhat difficult because of in . The analysis becomes fairly easy if we assume . Of course, it is possible to derive the PSD of without making the assumption that . This has been done in Appendix A7.1. In this appendix, it has been shown that the effect of is to produce spectral components beyond , which are anyway removed by the final, LPF. Hence, we proceed with our analysis by setting on the RHS of Eq. 7.25. Then reduces to,
Hence,
But,
Post detection LPF passes only those spectral components that are within . Hence the output noise power , resulting in,
As,
we have,
We can express in terms of the RMS bandwidth. From Eq. A5.4.7, (Appendix A5.4), we have
Hence
Using this value in Eq. 7.27(a), we obtain
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