EC 1351 – DIGITAL COMMUNICATION

UNIT I Pulse Modulation

Sampling Theorem for strictly band - limited signals

1.a signal which is limited to − W < f < W , can be completely
 n 
described by  g ( ).
 2W 
 n 
2.The signal can be completely recovered from  g ( )
 2W 
Nyquist rate = 2W
Nyquist interval = 1
2W
When the signal is not band - limited (under sampling)
aliasing occurs .To avoid aliasing, we may limit the
signal bandwidth or have higher sampling rate.
Let gδ (t ) denote the ideal sampled signal
∞
gδ (t ) = ∑ g (nT ) δ (t − nT )
n = −∞
s s (3.1)

where Ts : sampling period

f s = 1 Ts : sampling rate
From Table A6.3 we have
∞
g(t ) ∑ δ (t − nTs ) ⇔
n = −∞
∞
1 m
G( f ) ∗
Ts
∑ δ( f
m = −∞
−
Ts
)
∞
= ∑ f G( f
m = −∞
s − mf s )
∞
g δ (t ) ⇔ f s ∑ G( f
m = −∞
− mf s ) (3.2)

or we may apply Fourier Transform on (3.1) to obtain

∞
Gδ ( f ) = ∑ g (nT ) exp(− j 2π nf T )
n = −∞
s s (3.3)
∞
or Gδ ( f ) = f s G ( f ) + f s ∑ G( f
m = −∞
− mf s ) (3.5)
m≠0

If G ( f ) = 0 for f ≥ W and Ts = 1
2W
∞
n jπ n f
Gδ ( f ) = ∑ g ( ) exp(− ) (3.4)
n = −∞ 2W W
With
1.G ( f ) = 0 for f ≥W
2. f s = 2W
we find from Equation (3.5) that
1
G( f ) = Gδ ( f ) , − W < f < W (3.6)
2W
Substituti ng (3.4) into (3.6) we may rewrite G ( f ) as
1 ∞
n jπnf
G( f ) =
2W
∑ g ( 2W ) exp(−
n = −∞ W
) , − W < f < W (3.7)

n
g (t ) is uniquely determined by g ( ) for − ∞ < n < ∞
2W
 n 
or  g ( ) contains all information of g (t )
 2W 
  n 
To reconstruct g (t ) from  g ( ) , we may have
 2W 
∞
g (t ) = ∫ G ( f ) exp( j 2πft ) df
−∞

1 ∞
n jπ n f
∑
W
=∫ g( ) exp(− ) exp( j 2π f t )df
−W 2W n = −∞ 2W W
∞
n 1  n 
= ∑ g(
W

n = −∞
)
2W 2W ∫−W 
exp j 2π f (t − )  df (3.8)
2W 
∞
n sin( 2π Wt − nπ )
= ∑ g( )
n = −∞ 2W 2π Wt − nπ
∞
n
= ∑
n = −∞
g(
2W
) sin c(2Wt − n) , - ∞ < t < ∞ (3.9)

(3.9) is an interpolation formula of g (t )

Figure 3.3 (a) Spectrum of a signal. (b) Spectrum of an

undersampled version of the signal exhibiting the aliasing
phenomenon.
Figure 3.4 (a) Anti-alias filtered spectrum of an
information-bearing signal. (b) Spectrum of instantaneously
sampled version of the signal, assuming the use of a
sampling rate greater than the Nyquist rate. (c) Magnitude
response of reconstruction filter.
3.3 Pulse-Amplitude Modulation

Let s (t ) denote the sequence of flat - top pulses as

∞
s (t ) = ∑m(nT ) h(t − nT )
n =−∞
s s (3.10)

 1, 0<t <T
1
h (t ) =  , t = 0, t = T (3.11)
2
 0, otherwise

The instantaneously sampled version of m(t ) is
∞
mδ (t ) = ∑m(nT )δ (t − nT )
n =−∞
s s (3.12)
∞
mδ (t ) ∗ h(t ) = ∫ mδ (τ )h(t −τ )dτ
−∞

∞ ∞
=∫
−∞
∑m(nT )δ (τ − nT )h(t −τ )dτ
n =−∞
s s

∞ ∞
= ∑m(nTs )∫ δ (τ − nTs)h(t −τ )dτ (3.13)
n =−∞
−∞

Using the sifting property , we have

∞
mδ (t ) ∗ h(t ) = ∑m(nT )h(t − nT )
n =−∞
s s (3.14)
 The PAM signal s (t ) is
s(t ) = mδ (t ) ∗ h(t ) (3.15)
⇔ S ( f ) = Mδ ( f ) H ( f ) (3.16)
∞
Recall (3.2) gδ (t ) ⇔ fs ∑ G( f − mf )
m = −∞
s (3.2)
∞
Mδ ( f ) = f s ∑M( f −k f )
k = −∞
s (3.17)
∞
S( f ) = fs ∑ M ( f − k f )H ( f )
k = −∞
s (3.18)

Pulse Amplitude Modulation – Natural and Flat-Top

Sampling
 The most common technique for sampling voice in PCM systems
is to a sample-and-hold circuit.

The instantaneous amplitude of the analog (voice) signal is held

as a constant charge on a capacitor for the duration of the
sampling period Ts.

 This technique is useful for holding the sample constant while

other processing is taking place, but it alters the frequency
spectrum and introduces an error, called aperture error, resulting
in an inability to recover exactly the original analog signal.
The amount of error depends on how mach the analog changes
during the holding time, called aperture time.

 To estimate the maximum voltage error possible, determine the

maximum slope of the analog signal and multiply it by the aperture
time ∆T
Recovering the original message signal m(t) from PAM
signal

Where the filter bandwidth is W

The filter output is f s M ( f ) H ( f ) . Note that the
Fourier transform of h(t ) is given by
H ( f ) = T sinc( f T ) exp(− jπ f T ) (3.19)
amplitude distortion delay = T
2
⇒ aparture effect
Let the equalizer response is
1 1 πf
= = (3.20)
H ( f ) T sinc( f T ) sin(π f T )
Ideally the original signal m(t ) can be recovered completely.
Other Forms of Pulse Modulation

In pulse width modulation (PWM), the width of each

pulse is made directly proportional to the amplitude of
the
In pulse position modulation, constant-width pulses
are used, and the position or time of occurrence of
each pulse from some reference time is made directly
proportional to the amplitude of the information
signal.
Pulse Code Modulation (PCM)

Pulse code modulation (PCM) is produced by analog-to-

digital conversion process.
As in the case of other pulse modulation techniques, the
rate at which samples are taken and encoded must
conform to the Nyquist sampling rate.
The sampling rate must be greater than, twice the highest
frequency in the analog signal,

fs > 2fA(max)

3.6 Quantization Process

Define partition cell
J k : { mk < m ≤ mk +1} , k = 1,2,, L (3.21)
Where mk is the decision level or the decision threshold.
Amplitude quantization : The process of transforming the
sample amplitude m(nTs ) into a discrete amplitude
ν (nTs ) as shown in Fig 3.9
If m(t ) ∈ J k then the quantizer output is νk where νk , k = 1,2,, L
are the representation or reconstruction levels , mk +1 − mk is the step size.
The mapping ν = g(m) (3.22)
is called the quantizer characteristic, which is a staircase function.
Figure 3.10 Two types of quantization: (a) midtread and
(b) midrise

Quantization Noise

Figure 3.11 Illustration of the quantization process.

(Adapted from Bennett, 1948, with permission of AT&T.)
Let the quantization error be denoted by the random
variable Q of sample value q
q = m −ν (3.23)
Q = M −V , ( E[ M ] = 0) (3.24)
Assuming a uniform quantizer of the midrise type
2m max
the step - size is ∆ = (3.25)
L
− m max < m < m max , L : total number of levels
1 ∆ ∆
 , − <q≤
f Q (q) =  ∆ 2 2 (3.26)

 0, otherwise
∆
1 ∆2 2
σ = E[Q ] = ∫
2
Q
2 2
∆ q f Q (q )dq = ∫ ∆ q dq
2
−
2 ∆ −2
∆2
= (3.28)
12

When the quatized sample is expressed in binary form,

L =2R (3.29)
where R is the number of bits per sample
R = log 2 L (3.30)
2m max
∆= (3.31)
2R
1 2
σQ2 = mmax 2 −2 R (3.32)
3
Let P denote the average power of m(t )
P
⇒( SNR ) o =
σQ2
3P
=( 2
)2 2 R (3.33)
mmax
(SNR) o increases exponentially with increasing R (bandwidth).
Pulse Code Modulation

Figure 3.13 The basic elements of a PCM system.

Quantization (nonuniform quantizer)
µ - law
log(1 + µ m )
ν = (3.48)
log(1 + µ )
dm log(1 + µ )
= (1 + µ m ) (3.49)
dν µ
A - law
 A(m) 1
 1 + log A 0 ≤ m ≤
ν = A (3.50)
1 + log( A m ) 1
 ≤ m ≤1
 1 + log A A
 1 + log A 1
dm  0≤ m ≤
= A A (3.51)
d ν (1 + A) m 1
≤ m ≤1
 A
Compression laws. (a) m -law. (b) A-law.

Figure 3.15 Line codes for the electrical representations of

binary data.
(a) Unipolar NRZ signaling. (b) Polar NRZ signaling.
(c) Unipolar RZ signaling. (d) Bipolar RZ signaling.
(e) Split-phase or Manchester code.

3.8 Noise consideration in PCM systems

(Channel noise, quantization noise)
Time-Division Multiplexing

Digital Multiplexers
3.11 Virtues, Limitations and Modifications of PCM

Advantages of PCM
1. Robustness to noise and interference
2. Efficient regeneration
3. Efficient SNR and bandwidth trade-off
4. Uniform format
5. Ease add and drop
6. Secure

3.12 Delta Modulation (DM)

 Let m[ n] = m(nTs ) , n = 0,±1,±2,
where Ts is the sampling period and m(nTs ) is a sample of m(t ).
The error signal is
e[ n] = m[ n] − mq [ n − 1] (3.52)
eq [ n] = ∆ sgn(e[ n] ) (3.53)
mq [ n] = mq [ n − 1] + eq [ n] (3.54)
where mq [ n] is the quantizer output , eq [ n] is
the quantized version of e[ n] , and ∆ is the step size
The modulator consists of a comparator, a quantizer, and an
accumulator
The output of the accumulator is
 n
m q [ n] = ∆ ∑ sgn(e[ i ])
i =1
n
= ∑ eq [ i] (3.55)
i =1

Two types of quantization errors :

Slope overload distortion and granular noise

Slope Overload Distortion and Granular Noise

Denote the quantization error by q[ n ] ,

mq [ n ] = m[ n] − q[ n ] (3.56)
Recall (3.52) , we have
e[ n ] = m[ n] − m[ n −1] − q[ n −1] (3.57)
Except for q[ n −1], the quantizer input is a first
backward difference of the input signal
To avoid slope - overload distortion , we require
∆ dm(t )
(slope) ≥ max (3.58)
Ts dt
On the other hand, granular noise occurs when step size
∆ is too large relative to the local slope of m(t ).
Delta-Sigma modulation (sigma-delta modulation)

The modulation which has an integrator can

relieve the draw back of delta modulation
(differentiator)
Beneficial effects of using integrator:
1. Pre-emphasize the low-frequency content
2. Increase correlation between adjacent samples
(reduce the variance of the error signal at the
quantizer input )
3. Simplify receiver design
Because the transmitter has an integrator , the receiver
consists simply of a low-pass filter.
(The differentiator in the conventional DM receiver is
cancelled by the integrator )
Figure 3.25 Two equivalent versions of delta-sigma
modulation system.