Pulse Modulation Lecture Notes 02
Pulse Modulation Lecture Notes 02
INSTITUTION
PULSE MODULATION
MODULE TITLE
EENG524
MODULE CODE
Information Theory
Information Content, Entropy or Average Information Content,
Entropy Rate, Entropy of Joint Event, Redundancy & Information
Efficiency or Relative Entropy, Channel Representation, Transition
Probability Matrix, Noise-free Channel, Channel Probability or Joint
Probability Matrix, Information Transmission Rate, Channel Capacity
MODULE DESCRIPTION C = Max { I (X : Y)} bits/symbol, where I (X : Y) = Trans-information
or Mutual Information = H (X) – H (X/Y) = H (Y) – H (Y/X), Channel
Capacity and Information Transmission Rate, i.e. C = Max {H’(X) –
H’(X/Y)} bits/sec., Hartley – Shannon Law of Information, Shannon’s
Law for Channel Capacity C = 2W log 2 {(S + N)/N}½ = B log2(1+S/N)
bits/sec., Fundamental Theorems of Information Theory:
R ≤ C … (1);
R > C … (2).
Coding:
Basic Principles, Binary Coding, Fano Coding Scheme, Huffman’s
Coding Scheme, Coding Efficiency, Algebraic Codes/Block or (n, k)
Codes, Convolution Code, Error Detection and Correction, the
Syndrome (S) = [H][R], Parity Check matrix, Hamming Codes.
References
SUGGESTED 1. Ziemer and Tranter. Principles of Communications. Systems,
Modulation and Noise. Houghton Miffin
READINGREFERENCE
2. A Bruce Carlson. Communication Systems. An introduction to
TEXTS/MANUALS/WEBSITES Signals and Noise in Electrical Communications. McGraw Hill.
02 – Pulse Analogue Modulation
LECTURE NUMBER
LECTURE DURATION 01
(HOURS)
At the end of this lecture the students should be able to :
SPECIFIC INSTRUCTIONAL 1. Differentiate between the various forms of pulse modulation
OBJECTIVES AND LEARNING 2. Apply the basic theoretical expressions to solve related
OUTCOMES engineering problems
3. Analyse the concepts of bandwidth and signal recovery in pulse
analogue modulation schemes
For this analysis our sampling signal is the unit impulse ∂(t) which is represented in Figure 15.
v(t)
t
2To To 0 To 2To
Let us first of all recall the properties of the delta function (t)
(i) (t) = 0 except at t = 0
(ii) t dt 1
i.e. area under the delta function is equal to its weight or strength
v t ) Vn e jnws t
n
where
Ts
2
1 1
V n= ∫
T s −T
∂(t ) e
− jn w t
dt= ; ∀ n
s
Ts
s
2
...2
2π 1
Also w s= ∧fs=
Ts Ts
∞
v s (t )=v m (t) ∑ δ (t−k T s)
k=−∞
∞ ∞
1
¿ v m (t) ∑ V n e jn w t =¿❑ v m (t)
s
T
∑ e jn w t ¿
s
k=−∞ k=−∞ ❑
..3
Taking a general term say
1 jnw t
v (t) e s
Ts m
and invoking the Frequency shift theorem we can write its spectrum as
1 jnw t 1
v (t) e s
↔ V m (w−n ws )
Ts m Ts
...4
Giving
∞
1
V s ( w )= ∑ V (w−w s )
T s n=−∞ m
...5
Spectrum of Vm(w)
|V m (f )|
f
-fs -B B fs
Fig. 17. a. Baseband signal spectrum
|V s (f )|
Hence provided fs ≥ 2B, we will obtain the spectrum as above. We note that a true replica of the
baseband spectrum centred at ±nfs is obtained unlike the case of using a pulse train as the switching
or gating signal discussed earlier. It can be seen that filters can be designed centred at any ±nfs to
recover the original signal. This is therefore the ideal situation.
S(t)
Balanced Modulator
1.0 τ
v(t) xs(t) = v(t)s(t)
t
T
s(t) v(t)
∞
S ( t )= ∑ f s τSinc (n ¿ f s τ )e j 2 πn f τ ¿
s
n=−∞
...6
Alternatively we can write
∞
S ( t )=C 0+ ∑ 2C n cosn w s t ..
n=1
7
where
C n=f s τsinc( n f s τ ); ωs =2 π f s
...8
Hence noting that the output
we obtain
The circuit in Fig. 19a shows a schematic representation of a Bipolar Chopper, where the baseband
signal and its inverted form are alternatively transmitted using a switching function sampling at a rate
of fs. This effect can be realised by applying a bipolar switching waveform alternating between ±1.
Fig 19b shows the output of the chopper in the time domain.
X1 xs(t) m(t)
m(t)
fs
SPDT_OPEN
t
a. Bipolar Chopper xs(t)
x(t)
S(t)
1.0
t t
-1.0
Ts Ts/2
c Bipolar switching signal d. Bipolar modulation
PAM
Ts/2
Figure 19. Timing waveforms
Let us assume the spectrum of the baseband signal as shown in Fig. 20a and the bipolar switching
waveform as shown in Fig. 20b. We can therefore write in Fourier series expansion:
[ 4 1
3
1
5 ]
1
S ( t )= cos ω s t− cos 3 w s t+ cos 5 ws t− cos 7 w s t+ …
π 7
..13
From which we obtain the output xs(t) as
[ 4 1
3
1
5
1
]
x s ( t ) = x (t)cos ω s t− x (t )cos 3 ws t+ x (t)cos 5 w s t− x (t)cos 7 ws t+ …
π 7
..14
We note the DSBSC effect of the multiplication operation with the spectra centred on ±nfs shown in
Fig. 20c.
NB.
1. The Fourier series of the switching waveform has no DC term, hence the PAM waveform has its
first term centred on ± fs.
2. Only odd harmonics are present in the analysis
3. Bandpass filtering is required to extract the baseband signal from the modulation with the filters
centred on nfs where n is an odd integer.
4. This form of PAM forms the basis for the production of the baseband multiplexed signal for FM
stereo.
| X (f )| | X s (f )|
f f
fs 3fs 5fs
a. Baseband signal spectrum c. vPAM(t)
S(t)
1.0
t
Samples
v(t)
Hold
Hold period
Period Ts t
1. The signal is sampled in accordance with the Sampling Principle at intervals of Ts.
2. The samples are ‘held’ at their values for a period of time up to Ts seconds long, when the next
sample is taken.
3. The new value of the sample will either rise or fall to the instantaneous value of the baseband
signal at the sampling time.
A rudimentary circuit is shown in Fig 22 demonstrating the Sample and Hold principle.
In this circuit the switch S is momentarily closed, i.e. it samples v(t), and causes the capacitor C to
charge to the sample value at the sampling instant. S is kept open for Ts seconds before closing again
for an instant to take another sample of v(t) and the cycle repeats itself.
Ideally the source resistance R s → 0, e.g. a voltage follower circuit could be used for this purpose, and
R → ∞, i.e. a tantalum capacitor followed by another voltage follower circuit. S is an electronic switch,
i.e. a diode bridge or a FET as discussed earlier.
The S & H sample is then ready for conversion to binary form during the time interval available
between samples, i.e. Ts using analogue to digital techniques.
S
R1
V1
C1
R2
Q2 vPAM(t) = xp(t)
C1 x(t) x(kTs)
x(t) G1(p(t-kTs)) v0(t) =
G2 xp(t)
τ
kTs t
Noting that an n-channel FET is triggered by a positive signal, the circuit is activated by the closing of
the sampling switch by a gate pulse on G1, applied briefly thereby charging capacitor C to the sampled
value. This value is held until discharged by a pulse applied at gate G2. In both instances when a
positive gating signal is applied to the FETs the Drain –Source presents a near short circuit. In the case
of G1, the sampled value is transmitted to the capacitor C1. In the case of G2 the short circuit causes
the capacitor to discharge quickly to zero, awaiting the next sample value.
Periodic gating of the S & H circuit generates the sampled wave given by
❑
x p (t )=∑ x(k T s¿ ) p(t−k T s )¿
k
..15
Each output pulse of duration τ, represents the instantaneous value of the sampled signal.
Let us now analyze the waveforms using Signal theory.
Now
p ( t−k T s )= p ( t )∗∂ (t−k T s)
...16
i.e. the convolution of a function with the delta function produces the original function at the location of
the delta function
The sampled output becomes
❑
x p (t )=∑ x ( k T s ) [ p ( t )∗∂ ( t−k T s ) ]
k
...17
❑
¿ p ( t )∗∑ x ( k T s ) ∂ ( t−k T s )
k
¿ p ( t )∗x ∂ (t)
...18
Taking the Fourier transforms we obtain the output transform as
❑
X p ( f )=P ( f ) {f s ∑ X ( f −n f s ) }=P ( f ) X ∂ (f )
n
...19
Note: The Fourier transform of the convolution of two functions is the product of their individual
transform.
Practical Sampling
The following differences are observed between ideal and practical sampling:
1. The sampled wave consists of pulses having finite amplitude and duration rather than impulses.
τ
t
4. Also it can be noted that the shape of the sampling pulses used is not very critical as the basic
process reduces to a multiplication operation.
Figure 24 provides a description of the limitations of practical filters in contrast to ideal filters and ideal
sampling. Let us assume a baseband spectrum as shown in Fig. 24a. The output PAM waveform is
shown in Fig 24 b where a practical reconstruction filter response is shown. It can be seen that in this
case the filter response does not drop off sharply as in the case of an ideal filter. In filtering out the
baseband spectrum at the dc level it is seen that additional spectral components will appear at the
output of the filter at frequencies greater than f s – W. This occurrence manifests itself as a hissing high
frequency noise in audio systems and is outside the message band. Although an undesirable
phenomenon, it is not critical since careful design of the roll off of the filter can reduce this situation,
which in any case occurs outside the message band of the baseband signal and is heavily attenuated
compared to the message signal. Furthermore these components are a function of the baseband
signal x(t). If x(t) = 0 they will disappear. For x(t) ≠ 0, the spectral amplitudes within W are large
enough to reduce the effect of the components at the tail end of the filter response and so making
their effect insignificant.
Effect of Aliasing
Figure 24c shows the spectrum of a message signal which has components well beyond the nominal
bandwidth, W, of the message. These are referred to as pre-cursor and post-cursor components.
Here we note that spectral amplitudes above W are insignificant hence the stated bandwidth of W for
the signal. The sampled output is shown in Fig. 24d.
Xs(f)
Spectrum of baseband
Filter response
w
-W W
a. Baseband spectrum f
-fs -W 0 W fs-W fs
b. Practical reconstruction filter
X(f)
Xs(f)
Bandlimiting the baseband
Prior to sampling fs
f f1
-f1 -W 0 W f1
c. Spectrum of time limited message -fs -WW fs f
f s - f1< W
d. Sampled message spectrum showing overlaps
It can be seen that a sidelobe which originally had a peak at ±f 1, now appears at fs – f1, well within the
bandwidth of the first sample centred at the zeroth frequency. This will lead to distortion of the
message signal, making it very difficult to extract the original spectrum using a BPF, without any
distortion. This phenomenon occurs whenever a frequency component is undersampled and when fs <
twice the largest significant frequency of the baseband signal, in this case f s < 2f1. Furthermore
frequencies such as f1 > W will now become fs – f1 < W as shown in Figures 24 c and d. The above
phenomenon of downward frequency translation, where frequencies which were originally outside
the nominal message band appear at the filter output in the form of much lower frequencies is referred
to as aliasing. Aliasing is more critical than the earlier situation where the unwanted frequency
components occurred outside the message band.
Remedy
1. The combination of careful filter design, as well as creating adequate guard bands between
samples by ensuring that fs > 2W makes practical filters approach the ideal case;
2. Introducing pre-filtering before sampling of the message or baseband signal and also if necessary
sampling at a rate much higher than the Nyquist rate combats aliasing and
3. Also although the amplitudes at f1 in the spectrum of the baseband signal may appear
insignificant it makes practical sense to apply a sampling rate fs >2f1 to ensure a reduction in
the effect of aliasing.
Xs(f)
f
-fs f1 fs
A practical example
The average voice signal normally extends beyond 10 kHz. However its energy is concentrated
between 100 and 600 Hz. In practice a bandwidth of 3 kHz is provided to transmit intelligible speech.
If a voice signal is pre-filtered by a 3.3 kHz LPF and then sampled at f s = 8 kHz, the aliased
components are typically less than 30 dB (i.e. less than1000 units) below the desired signal and will go
un-noticed by the listener. These are the standards in voice telephone system. Discuss.
In PCM systems, the effects of noise and distortion are combated by encoding the sampled values that
constitute the PAM signal.
In principle, each sampled value is converted into a corresponding binary code of ‘on-off’ pulses (i.e.
binary 1’s and zero’s), which must be transmitted during the interval between successive sampling
instants.
At the receiver, these On-Off pulses will also be corrupted by noise. However, the receiver has only to
decide whether a pulse is present or not and hence there is greater immunity to noise.
Finally a means must be found for reconstructing the sampled values of the original baseband signal
from the binary coded Signal. Let us now discuss the following points.
t t t t
Figure 25. Effect of noise and distortion on a single channel PAM
It must be stated that it is impracticable to represent every possible sampled value of the baseband
signal in binary form as this would require an infinite-length binary code. However if we divide the
voltage range which encompasses all possible sample magnitudes of the baseband signal into a
suitably large number of levels, we can then ‘round off’ every sampled value to a nearby level. This
process of ‘rounding off’ the sampled values to a nearby pre-defined level is referred to as
quantising and the subsequent operation of generating the binary codeword appropriate to the
particular quantised sample is known as encoding. The inverse operation of recovering the quantised
sampled value from the coded binary message is called decoding. Due to the binary nature of the
codeword, there must be a definite relationship between the length of the codeword and the number of
quantising levels.
Thus for a 5-digit code the number of quantisation levels will be 2 5 = 32. Thus level 1 will be
represented by say, 00000, level 2 will be represented by 00001, and so on until level 32 which is
11111.
A block schematic of a single channel PCM system incorporating the above features is shown in Fig 26
The quantising and encoding operations are usually performed by a single circuit referred to as an
Analogue-to-Digital Converter (ADC). The decoder is also referred to as a DAC. The waveforms at
various points in the system of Figure 26 are demonstrated in the illustrative example. Here a 3 digit
code is used which means that the number of quantisation levels is 2 3 = 8.
We note that although at first sight we appear to overcome the problem of noise by encoding the
quantised PAM signal {v2(t)} into a digital form, the quantisation process performed on the original
analogue PAM signal [ v1(t)], results in some loss of signal definition. This is analogous to introducing
some ‘noise’ into the system. The induced noise is referred as quantisation noise. Also by increasing
the number of quantisation levels (and therefore the length of the binary codeword), the quantisation
noise will be reduced, and the decoded and filtered signal will approach a faithful replica of the original
baseband signal.
2022-07-21
Discuss Simulation Assignment (https://www.youtube.com/watch?v=rg6enk_PcOk;
https://www.youtube.com/watch?v=ibnz5UjQ4u0.)
Illustrative example
A single channel PCM system samples the input baseband waveform resulting in the following results:
No. Sampled
values
1 2.8 V
2 7.0 V
3 7.7 V
4 4.4 V
5 2.3 V
Solution
a. Block schematic
LPF
Rotating switch/Decommutator
Rotating Switch/Sampler/ vPAM(t) vPCM(t) vPAM(t)
Commutator
Figure 26. A single channel PCM system
b. Waveforms
Volts (v1)
Code Level No. Quantisation
Levels (Volts)
111 8 7.5 8
7.7 V
110 7 6.5 7 7.0 V
6
101 6 5.5
5
100 5 4.5
4 4.4 V
011 4 3.5
3 2.8 V
010 3 2.5
2 2.3V
001 2 1.5
1
000 1 0.5
0
T
110 7 6.5 7
101 6 5.5 6
100 5 4.5 5
011 4 3.5 4
3
010 3 2.5
2
001 2 1.5
1
000 1 0.5
0
t
T
Fig. Flat top PAM (Sampled values are rounded off to nearest quantisation level.)
v3(t)
0 1 0 1 1 0 1 1 110 0 0 1 0
t
Fig. Binary coded sampled values
QUANTISATION
Practically, the number of pulses used in coding any level of baseband signal v i(t) is finite and an
integer value.
Let the number of quantisation levels chosen = n
Let the number of pulses in a group or the length of the codeword = m
Let the number of possible amplitudes of pulses = μ.
We therefore obtain
m
μ ≥n
..20
Or
m ≥ log μ n
..21
Example
Given a binary system with the number of quantisation levels = 128, find the length of the code word
used.
Bank of LPFs
m’2(t)
m2(t) Demultiplexer/Decommutator
NB
The final output m1’(t) is slightly different from m1(t) due to the quantisation process. The same
applies to m’2(t).
Illustrative Example
A PCM multiplexing system permits the combining of two baseband signals, m 1(t) and m2(t). The
signals are band-limited to 4 kHz and 6 kHz respectively and have voltage amplitudes in the range of 0
– 8 Volts. Each signal is sampled, quantised into 8 levels and finally represented by a 3-digit code.
The binary pulses occupy 60% of their assigned time slots.
i. Draw a block schematic of the system and illustrate with waveforms.
ii. Determine the minimum sampling rate of the system
iii. What is the maximum bit duration allowable?
iv. Sketch the frame of the PCM output when the sampled values are 3.2 V and 6.8 V respectively.
v. Find the Gross bit rate of the system if one bit or marker pulse is assigned to each channel for
synchronisation.
Solution
i. Same diagram as in Fig. 27. The waveforms will be illustrated later and are as shown below.
Therefore the minimum sampling rate required for the multiplexed system = f smin = 12 kHz.
Refer to the general rule for determining fsmin for multiplexed signals having a common sampling rate.
vPAM(t)
T t
T/2 T/2 NB. T = common sampling interval
Equal assigned
Time slots for each binary symbol
Now we note that the assigned time slot for each binary digit can be written as
T
No. of channels T /2 T T
Assigned time slot = = = =
No of digits∈code No . of digits∈code 2 x 3 6
83.3
τ max=0.6 x =8.33 μs .
6
iv.The frame of the PCM system is the collection/arrangement of digits representing the samples of the
signals within the sampling interval, T, of the PCM system.
Waveforms
Volts
Code Level No. Quantisation 6.8 V: Rounded off as 6.5 V
Levels (Volts) encoded as 110
111 8 7.5 8
110 7 6.5 7 m2(t) m1(t)
6
101 6 5.5
5
100 5 4.5
4 m1(t) m2(t)
011 4 3.5
3
010 3 2.5
2
001 2 1.5
1
000 1 0.5
0 t
Figure P. Waveforms and quantisation process T
3.2 V: This is rounded off to 3.5 V and
Hence encoded as binary code 011
0 1 1 1 1 0
v. Find the Gross bit rate of the system if one bit or marker pulse is assigned to each channel
for synchronisation.
0 1 1 1 1 0
Marker
t
60% of assigned
Time slot
T
Assigned Time slot for binary digits
This is defined as
Total number of bits∈a frame (T ) 2 ( 3+1 )
GBR= = =96 kbits/ sec
Time intervalT 83.3 x 10
−6
NB.
1. When a marker pulse is inserted at the end of each rotation of the commutator for
synchronisation, the frame is given by the interval between the markers. This is referred to as
frame synchronization.
2. When markers are inserted after each channel this is referred to as channel synchronization.
3. In some cases, both frame and channel synchronisation are used to facilitate the separation or
decoding of the channels.
01 1 1 1 0
Frame Marker
(synchronization)
t
60% of assigned
Time slot
T
Assigned Time slot for binary digit
Although PCM is attributed with an impressive noise immunity due to the nature of the binary coded
data that is transmitted and the process of decision making over only two levels at the receiver, the
process of quantising introduces errors during the rounding off (quantisation noise), as well as the
additive noise due to the inherent imperfection of the transmission channels. Hence two categories of
noise are inherent in PCM systems.
For this analysis however we shall concentrate on the impact of the quantisation noise on the
transmission efficiency of the signal.
Quantisation Noise
Consider an analogue signal having a peak-to-peak voltage 2V quantised into q equal levels. Let the
voltage spacing be ∆ v volts which for linear quantisation is given by
2V
∆ v=
q−1
.22
Refer to Waveforms in Fig. 28
We note that the error ε between an analogue voltage and its quantised value is always less than ½∆V.
Now let us assume that all quantisation levels have an equal probability of occurrence over a long
period of observation. The mean square error voltage ϵ 2 is then given by
1
∆V
2
1
ϵ 2=
∆V
∫ ϵ 2 dϵ
−1
∆V
2
.23
[ ]
1
1 ε3 2
∆V
1 2 ∆V3 ∆V2
¿ = . . =
∆V 3 −1
2
∆V ∆V 8 3 12
..24
Waveforms
Volts
Code Quantisation. Quantisation
Level No. Levels (Volts) 4.0
111 8 3.5
3.0
110 7 2.5
2.0 ∆v
101 6 1.5 1.0
100 5 0.5
0 t
011 4 -0.5 m1(t)
- 1.0
010 3 -1.5
-2.0
001 2 -2.5
-3.0
000 1 -3.5
-4.0
Let us assume bipolar levels are used, i.e. the quantisation levels are [Refer to Fig. 28]
1 3 5 1
± ∆ V , ± ∆V , ± ∆ V … , ± ∆ V ( q−1 ) ,
2 2 2 2
Also we assume that in a long message, each level is equally likely to occur with a probability of 1/q.
We will now add the powers associated with each quantisation level to obtain
[ ]
2 2 2 2
2 1 3 5 1
S= ( ∆ V ) +( ∆ V ) +( ∆ V ) +… ( (q−1) ∆ V )
qR 2 2 2 2
..26
1
Where the factor of 2 occurs because of the bipolar nature of the levels (e.g. ± ∆ V ), and the load
2
resistance is R
NB
Vpp =( q-1)∆ V ..27
[ ]
2
2 2 2 q −1 2
1 +3 +5 + …(q−1) =q
6
Hence we obtain
2 2
∆V (q −1)
S=
12 R
.30
The Signal power to Quantisation noise ratio is therefore given by
2 2
S ∆ V (q −1)/12 R 2
= 2
=( q −1)
N ∆V /12 R
..31
which for a large number of quantisation levels we obtain
S 2
=q
N
..32
Exercise
Table.
No. of S/N(db) No. of bits
levels 10log[(S/N)] (Codeword)
16 24 4
32 30 5
64 36 6
128 42 7
In practice non-linear quantisation is often used, i.e. the quantising intervals are not equally spaced
because the amplitude levels in speech and noise are not equally distributed. Logarithmic laws are
used that dictate the quantisation levels and step or quantum size, such as the A-law and the μ-law.
Consider an n-digit codeword. Also let the Bandwidth or the highest frequency of the signal be B.
Sampling interval ( T ) T
No of channels 1 1 1
τ= = = =
No . of bits∈T n bits f s n 2 nB
…33
v PAM
The bandwidth W required to transmit a pulse of length τ is given by [See diagram below for
explanation]
1
W= Hz
2τ
..34
G(f)
gT(t)(i.e. binary pulse) Here trans. bandwidth
Aτ reqd. W = 1/τ. In practice W =
1/2τ is used.
A
2 1 1 2
f
- /2 /2 t
Giving
W = nB ….35
1
Noting that τ = from Eqn 33
2nB
It has been shown that
S 2
=q
N
This becomes
S n❑ 2
=( 2 ) =2
2n
N
..36
Resulting in
2W
s B
=2
N
..37
i.e. an exponential increases in S/N with required transmission bandwidth W. This implies that
increasing the transmission bandwidth W increases the S/N, exponentially. This is better than the case
for analogue modulation.
1
W =nB=
2τ
…38
2. Increasing W implies a reduction in τ, i.e. allowing more bits to be transmitted within the
assigned time slot and hence a larger codeword can be used.
3. A larger code word implies an increased number of quantisation levels which in turn reduces the
quantisation error in encoding the signal [Refer to Tutorial]. This implies an increase in the S/N
and hence greater transmission efficiency. However this is at the expense of equipment
complexity and cost, a trade-off that will have to be made by the engineers.
Three techniques are available for modulating PCM, now a sequential series of binary 1’s and 0’s on to
a carrier, i.e. PSK, ASK and FSK as shown in the schematic in Fig. 31
m(t) A B
Demodulator
Modulator
Error rates
10-3
x PSK FSK ASK
10-5
S/N
-2 0 4 6 10 12 14
NB
1. The probability of error of transmission is a function of the SNR and acceptable error limits
depends on the application.
2. An error rate of 10-6 implies 1bit error in 106 or 1,000,000 bits transmitted.
3. The average number of errors in a message of m bits length is given by:
Refer to Tutorial
ASSIGNMENT
1. 24 voice channels are to be transmitted via a multiplexed PAM system with a marker pulse for
frame synchronisation. The sampling frequency is 8 kHz and the TDM signal has a 50% duty
cycle. Calculate the Gross Signalling Rate, pulse duration and minimum transmission bandwidth.
2. The signal v(t) = 4cosw0t + 2cos2w0t + 2cos3w0t is to be converted into a PAM train by
multiplying v(t) by a unit amplitude impulse train given by:
2 1
s(t )= [ +coswt+ cos 2 wt + coswt+… .]
T 2
2π
Where T = is the period of theimpulse train .
w
The signal is to be reconstituted from the PAM train by passing through a Low Pass Filter.
a. If v(t) is to be recovered without distortion, determine the minimum value of w and the minimum
bandwidth of the LPF.
b. If v(t) is sampled at 5w0, and passed through an ideal LPF with cut-off frequency of 3w 0,
determine the expression for the output signal.
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