SZE 3533

Topic IV – Pulse Modulation

4.0 Digital Communication System

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 4.0 Introduction
•In the early 90’s, telecommunication networks is changing towards
digital world. With the rapid advancement in the fields of VLSI and
microprocessor, several telecommunication components such as
transmission line and switching has been using digital signals in
their operation.

•Therefore, information signals must be changed to digital form so

that it can be transmitted through this network.

•Several techniques requiring full coding of the original signal will

be used:
• Pulse Code Modulation (PCM)
-      
• Differential PCM (DPCM)
• Adaptive Differential PCM (ADPCM)
• Delta Modulation (DM)
• Adaptive Delta Modulation (ADM)
 4.1 Digital Modulation
• Advantages :
– Immunity to noise
– Easy storage and processing: MP, DSP, RAM, ROM, Computer

– Regeneration
– Easy to measure
– Enables encryption
– Data from several sources can be integrated and transmitted using
the same digital communication system
– Error correction detection can be utilized

• Disadvantages :
– Requires a bigger bandwidth
– Analog signal need to be changed to digital first
– Not compatible to analog system Voice : Analog : 4 kHz
– Need synchronization Digit : 2 x 4 kHz x 8 bit = 64 kb/s
BWmin  32 kHz

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 4.2 TRANSMISSION METHOD FOR ANALOG &
DIGITAL SIGNALS
Analog Analog channel Analog
input Baseband output
Analog Analog De Analog
input Modulator
channel modulator output

Digital Digital decoder Digital

encoder
input channel output

Digital Analog Digital

Modem Modem
input channel output

Analog ADC & Digital Decoder Analog

input encoder channel & DAC output

Analog ADC & ADC & Analog

Modem Analog Modem output
input encoder channel decoder
 4.3 Pulse Modulation
Pulse Modulation consists of:

• PAM (Pulse Amplitude Modulation) => VPAM  Vm

• PWM (Pulse Width Modulation) =>   Vm
• PPM (Pulse Position Modulation) => d (pulse delay)  Vm
• PCM (Pulse Code Modulation)

Less susceptible to
noise

Less susceptible to
noise compared to
PAM
Easily effected by
noise
 4.3 Sampling Theorem
m (t)
m(t) s

m(t) X ms(t)
t t

s(t)
Digital signal
s(t) t
fs  2 fm
Fourier series for impulse train : Ts
1
s  t   1  2 cos  s t  2 cos 2 s t  2 cos 3 s t  ..... Nyquist theorem
Ts states that:
2 1
s   2f s where T s 
Ts fs
Therefore :
ms  t   m t  s  t 
1
  m  t   2m  t  cos s t  2m t  cos 2s t  2m  t  cos 3s t  .....
Ts
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 Time domain Frequency domain

m(t ) M ( )

t 
0  m 0 m

s (t ) S ( )

2
Ts

t 
 6Ts 0 Ts 6Ts  s 0 s

ms (t ) M s ( )
1
Ts

  s  m s  m
t 
 6Ts 0 Ts 6Ts  s  m 0 m s
 s  m  s  m

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 Reconstruction

TX RX
M ( ) M r ( )

1 ms(t) 1
m(t) X h(t) mr(t)
 
 m 0 m  m 0 m
Low pass filter
M s ( )
Pulse signal 1
H ( )

s(t) Ts
Ts
  s  m  s  m

 s  m 0  m s
 s  m  s  m 
 n 0 n

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• Sampling process shown previously uses an ideal pulse signal
• However, it is quite difficult to generate an ideal pulse signal practically
• The usual pulse signal generated is as shown below:

s(t)

A

t
Ts
 - pulse width
A 2 A 
2 nt
Ts – pulse period s (t ) 
Ts

Ts

n 1
cn kos
Ts
di mana
n
sin
Ts n
cn   sinc
n Ts
Ts

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 m(t)

Information signal
t

s(t)
Pulse signal 
t
Ts
Sampled signal (PAM)
ms(t)  ms(t)
   Ts 

t t
Ts Ts

Natural Sampling Flat-top Sampling

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 4.3.1 Difference in Sampling Methods
ms(t)
Natural Sampling

Ideal Sampling Flat-top Sampling

• In every sampling methods, the pulse amplitude is directly proportional to the

amplitude of the information signal
• Practically, an ideal sampling is difficult to generate
• However, by using an ideal and natural sampling, noise can be eliminated, which
is not the case for flat-top sampling

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Mathematical analysis:

m(t) ms(t)

m(t) X ms(t)
t t

s(t)
Pulse signal
s(t) t

Fourier series for pulse signal, s(t) :

 2 
n 2nt
s (t ) 
Ts

Ts

n 1
sinc
Ts
cos
Ts
Therefore, the sampled signal:

ms (t )  m(t ) s (t )

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  2  2nt 
ms (t )  m(t ).   cn cos 
 Ts Ts n 1 Ts 
m(t ) m(t )2  2nt
ms (t ) 
Ts

Ts

n 1
c n cos
Ts

For n = 1, 2 , 3 …..
m(t ) m(t ) 2 2t m(t ) 2 4t
ms (t )   c1 cos  c2 cos 
Ts Ts Ts Ts Ts
m(t )2 6t
c3 cos  ....
Ts Ts

The above expression shows that the frequency components of the

sampled signal is at fs , 2fs and 3fs . Components 2fs and 3fs is a replica of
the spectrum of the sampled signal.
 m(t ) m(t ) 2 2t m(t ) 2 4t
ms (t )   c1 cos  c2 cos 
Ts Ts Ts Ts Ts
m(t )2 6t
c3 cos  ....
Ts Ts

The spectrum of the sampled signal has sidebands fs  fm , 2fs  fm , 3fs 

fm and so on.

Spectrum of the sampled

ms(f) signal

f
fs-fm fs fs+fm 2fs-fm 2fs+fm 3fs-fm3f 3fs 3fs+fm
0 2s 2fs
The choice of sampling frequency, fs must follow the sampling theorem
to overcome the problem of aliasing and loss of information

Shannon sampling
(a) Sampling frequency=> fs1 < 2fm (max) theorem=> fs  2fm
ms(f) Aliasing
Nyquist frequency
 fs = 2fm= fN
f A bandlimited signal that
fm fs1 2fs1 3fs1 has a maximum
frequency, fmax can be
regenerated from the
(b) Sampling frequency=> fs2 > 2fm (max) sampled signal if it is
ms(f) sampled at a rate of at
least 2fmax .

f
fm fs2 2fs2 3fs2
 4.4 Detection of Sampled Signal
By using LPF to the sampled signal, ms(t)

ms(t) LPF m(t)

Cut-off frequency , fo for LPF must be within the range: fm  fo  fs - fm

• Eventhough the sampled signal can be detected easily at fs = 2fm , but usually
fs > 2fm . The main reason is to have a ‘guardband’ .
• Therefore, the maximum frequency that can be processed by the sampled
data using sampling frequency, fs (without aliasing) is:
=> fm = fs / 2 = 1 / 2Ts
Mathematical analysis:

From the sampling process, the sampled signal:

m(t ) m(t ) 2 
2nt
ms (t ) 
Ts

Ts

n 1
cn cos
Ts
n
where : cn  sinc
Ts
If:
  Ts n
therefore sinc 1
Ts
m(t ) m(t )2 
2nt
Therefore m s (t ) 
Ts

Ts

n 1
cos
Ts

Taking: m t   1  cos  m t 
replacing m t   1  cos  m t 
m(t ) m(t )2 
2nt
inside m s (t ) 
Ts

Ts

n 1
cos
Ts
 2 
m s  t   1  cos  m t   1  cos  m t   cos n s t
Ts Ts n 1

  
 1  cos  m t   1  cos  m t   2 cos n s t
Ts Ts n 1

  

 1  cos  m t   
1  2 cos n  t
s 
Ts  n 1 
  
 1  cos  m t  1  2 cos  t  2 cos 2 t  2 cos 3 t  ....
Ts  
s s s

 1  cos  m t  2 cos  s t  2 cos  s t cos  m t 
  
Ts   2 cos 2 s t  2 cos 2 s t cos  m t  ... 
 1  cos  m t  2 cos  s t  cos  s   m  t  cos  s   m  t 
  
Ts  2 cos 2 s t  cos 2 s   m  t  cos 2 s   m  t  .........
It can be shown that the output sampled signal is the same as the output
PAM signal when :
  Ts => ms(t) = VPAM

that is, the pulse width  is much smaller compared to the pulse
period Ts .

Voltage
vm(t) translator vPAM(t)

vd(t)
(a) PAM generation

LPF
vPAM(t) vm(t)

(b) PAM detection

 4.5 Pulse Width Modulation (PWM)
•  (pulse width) follows the instantaneous value of the information signal
vm(t) :   vm (t )
o represents the width that is
   o   o cos  mt fixed according to the minimum
value of the information signal

The equation shows that the pulse

   o 1 cos  m t  width,  of the output signal
PWM varies according to the
instantaneous value of the
information signal.
Replacing  inside the general equation of the sampled signal:

vPWM    PWM   1  2 cos  s t  2 cos 2 s t  ...
Ts
  o 1  cos  mt 
vPWM    PWM   1  2 cos  st  2 cos 2 s t  ...
Ts
 o 1  2 cos  s t  2 cos 2 s t  cos  mt  2 cos  s t cos  mt 
vPWM   
Ts   2 cos 2 s t cos  m t  ...

 o 1  2 cos  s t  2 cos 2 s t  cos  mt  cos( s   m )t 

vPWM   
Ts   cos( s   m )t  cos( 2 s   m )t  cos(2 s   m )t ...

Generation of PWM signal is by changing the value of sample signal of the

PAM signal into a specific period

vPAM(t) 555 timer vPWM(t)

(a) PWM generation using voltage to time converter

vPWM(t) LPF vm(t)

(b) PWM detection using LPF

fs > 2fm

fs = 2fm fs < 2fm

 4.6 Pulse Code Modulation (PCM)
A method used to represent an analog signal in terms of digital word
Constitutes 3 process:
1. Sampling the analog signal
2. Quantization of the amplitude of the sampled signal
3. Coding of the quantized sample into digital signal

Sampling Quantization Coding

PCM process:
S/H : Sample and hold
circuit

Analog
signal LPF S/H ADC PCM

Anti aliasing
filter ADC : analog to digital converter
fs
4.6.1 Sampling
• An analog signal must be sampled at Nyquist rate to avoid
aliasing

4.6.2 Quantization & Coding

• Process of estimating the sampled amplitude into a value suitable for
coding (ADC).
• A fixed number of levels including the maximum and minimum value of
the analog signal
• Number of levels is determined by the number of bits used for coding
3 terms that are commonly used in the quantization
process:
• Quantization level, L = 2n
Quantization level depends on the number of binary bits, n used to
represent each sample.
For example:For = 3; Quantization level, L = 23 = 8 level.
In this example, first level (level 0) is represented by 000, whereas bit
111 represents the eigth level
• Quantization Interval
Represent the voltage value for each quantized level
For example: For a sampled signal that has 5V amplitude, Vpp = 10 V
divide by the quantized level, L = 8 level,
Therefore, quantized interval ,
10 V
V   1.25 V
8
• Quantization value, Vk
The middle voltage for each quantized level
For example: for n = 3, quantized level, L = 8 and a sampled sinusoidal
signal with +5 V ,
The middle quantized value for level 0,

1.25 V
V0  5 V   4.375 V
2

In this example, for a sample that is in level 0 segment will be

represented by bit 000 with a voltage value of –4.375 V. The difference
between the sampled value and the quantized value results in
quantization noise.

For voice communication 256 levels

are commonly used (i.e n = 8)
4.6.3 UNIFORM QUANTIZATION
Uniform quantization is a quantization process with a uniform (fixed)
quantization interval.

Example : n = 3 , L = 8 , signal +5 V ; => Vk = 1.25 V . Bit rate: f b  nf s

Quantization level & Quantized Sampled signal
binary representation value

+5.0V
Leve l 7 : 111 4.375V 4.3V

Level 6 : 110 3.125V

1.9V
Level 5 : 101 1.875V 1.9V

Level 4 : 100 0.625V

t
Level 3 : 011 -0.625V

Level 2 : 010 -1.875V

Level 1 : 001 -3.125V -3.2V

Level 0 : 000 -4.375V -4.5V

-5.0V
 4.6.3.1 Uniform Quantization using Folded Binary Code (sign bit)
The same code representing several
+mp samples with different amplitudes
0 11
0 10
∆ 0 01 Quantization error
Step size Qe
0 00
0 t
1 00
1 01
1 10

-mp 1 11

Sign bit
value
PCM code 000 001 011 011 011 010 001 100 110 111 111 110 100 001 010 010 010 000

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 4.6.3.2 Quantization error
Quantization error (Qe) is also called Quantization noise (Qn) . And its
maximum magnitude is one half of the voltage of the minimum step
size .

May add to or substract from the

actual signal
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4.6.3.2 Quantization error
Example : Uniform Quantization error
Binary Input voltage Input voltage range: –14 mV
number range (mV) to +14 mV
1 11 10 to 14
Qn = LSB voltage /2 = /2
1 10 6 to 10
1 01 2 to 6  14 mV = 28 mV with 8 steps and 8 codes.
1 00 0 to 2 Therefore  = 28/8 = 3.5 mV.
0 00 -2 to 0
Therefore : Qn = 3.5 mV / 2 = 1.75 mV
0 01 -6 to -2
0 10 -10 to -6 SNRq = [1.76 + 6.02n] dB : (for details, refer
to monograph page 122)
0 11 -14 to -10
Noise from quantization error can be
reduced by increasing the quantization
level i.e increase n.
3
Nonuniform quantization SNRq  10 log  6.02n dB
using  Law:  ln1     2
Example :

Vpp = 31.5 V PCM 6 bit code (5 bits for

system magnitude and 1 bit
for sign

(a) No of levels: 26 = 64
(b) LSB voltage,  : 31.5/64 = 0.492 V
(c) Maximum quantization level, /2 = 0.25 V
(d) Voltage value for 001101 ; +(13 x 0.492) = +6.4 V
(e) Voltage value for 111001 ; –(25 x 0.492) = -12.3 V
(f) Code for input +13.62 V
= 13.62/0.492 = 27.68  28 => 111100
(g)Code for input –9.37 V
= 9.37/0.492 = 19.04  19 => 010011
 4.6.4 Non uniform quantization
nonuniform: to improve SNR (SQR)

 More levels is available for low level amplitudes compared to high

amplitude
 Increase SNR for low level amplitude and decrease SNR for higher
amplitudes

analog compression is done to the input signal before sampling and

quantization at the transmitter
Expansion is done at the receiver
COMPANDING (compression and expanding)
4.6.4 Non Uniform Quantization
example : Non-Linear
Quantization

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Companding => Compress - Expanding
A method used to produce a uniform SNR for all input signal range is
compression-expansion (Companding).
Input signal is compressed at the transmitter and expanded at the
receiver
Companding => Compress - Expanding
=> Analog – Compression process is done on the input signal
before sampling and coding
=> Digital – compression process is done after the signal is
sampled
analog signal To digital channel
analog ADC
(input) compressor

Analog Analog signal

DAC (output)
expander

PCM with analog compress-expand

analog signal Digital To digital channel

ADC
(input) compressor

Digital Analog signal

DAC
expander (output)

PCM with digital compress-expand

2 Popular companding system (standardized by ITU)
• EUROPE => A - Law
• USA/NORTH AMERICA =>  - Law

1  log( Ax ) 1
 1  log A for  x1
y A
Ax 1
 for 0 x
 1  log A A

A - compressor paramater. Usually

the value of A is 87.6.

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USA/NORTH AMERICA =>  - Law

 Law is a standard compress-

expand that is used in America
and Japan. The value of  used
is 255 (8 bit).

log(1  x )
y
log1   

For both laws, the values of x and

y refers to the equation below:

Ei Eo
x y
Ei (mak ) Eo ( mak )

Pemodulatan Digit
 Example 4.3 :
A compress-expand system using  Law ( = 255) is used for a signal with
range 0 to 10V. Determine the output of the system if the input is 0 and
7.5V.
Solution :
Given  = 255 and Ei(mak)= 10 V
For Ei = 7.5 V
For Ei = 0 V Ei 7.5
x ; x  0.75
Ei 0 Ei (mak ) 10
x ; x 0
Ei (mak ) 10 Output :
Output : log(1  x)
y
log(1  x ) log(1  255(0)) log1   
y ;y
log1    log1  255 y
log(1  255(0.75))
log1  255
y0
y  0.948
Example 4.4 :
A random signal has gone through a 256 level quantization process.
Determine the quantization signal to noise ratio for this system.
Solution :
From the above statement, the number of sampling bits is not known.
But, given L=256
L = 2n
therefore, n = 8
Given SNRq

SNRq  1.76  6.02n dB

SNRq  1.76  6.02(8)  50 dB
 4.6.5 Bit rate for PCM transmission
Telephone Europe bit rate(Mb/s) Telephone North America bit
channel channel rate(Mb/s)
30 2.048 24 1.544

120 8.448 48 3.152

480 34.368 96 6.321

1920 139.264 672 44.736

7680 565.148 4032 274.176

SDH 2.5Gb/s
North American standard (NAS) : -Law

European standard : A-Law For every 24 sample, 1 bit is added for

30 + 2 control channel = 32
Bit rate= 32 x 8 bit/sample x 8000 sample/s
= 2.048 Mb/s
synchronization
 For 24 sampel => 24 x 8 bit/sample
+ 1 bit = 193 bits
 Bit rate= 193 x 8000 = 1.544 Mb/s

Needs Multiplexing – Process of transmitting two or

more signals simultaneously
 Example : PCM-TDM CEPT System
Frame structure and Timing : European standard PCM system : E Line
488 ns Bit duration

8 bits per
time slot
3.9 s
3.9 s
30 signal + 2 control = 32 channels = 1 frame

125 s
125 s Signalling & synchronization

2 ms
Duration of multiframe 16 frames = 1 multiframe
(a) bits per time slot (b) time slots per frame (c) frames per multiframe
CEPT system – 32 channels (30 signals + 2 control)
Frame structure and timing
Number of channel = 32
Number of bits in one time slot = 8
32 channels = 1 frame
Number of bits in a frame = 32 x 8 = 256 bits
This frame must be transmitted within the sampling period
and thus 8 x 103 frames are transmitted per second.

Therefore :
Transmission rate = 8 x 103 x 256 = 2.048 Mb/s
Bit duration = 1 / 2.048 x 106 = 488 ns
Duration of a time slot = 8 x 488 ns = 3.9 s
Duration of a frame = 32 x 3.9 s = 125 s => (= 1 / 8 kHz = 125 s)
Duration of a multi frame = 16 x 125 s = 2 ms
CEPT telephone system hierarchy

. E1 line
30 .
2.048 Mbps
Voice . MUX
.
channels . 1
.

E2 line
E1 MUX 8.448 Mbps
E1 2
E1

E3 line
E2 MUX 34.368 Mbps
E2 3
E2

E4 line
E3 MUX 139.264 Mbps
E3 4
E3
 4.7 Delta Modulation (DM)
Pulse signal
s(t)

comparator
+Δ
e(t) d(t)
m(t) ∑ X xDM(t)
+ -Δ
-
~ (t )
m
integrator


• There are 2 main components in the DM generator circuit, i.e comparator and
integrator.

Pemodulatan Digit
• Comparator will compare the error signal e(t), where
~ (t )
e(t )  m(t )  m
• Output signal from comparator has the following function:
  e(t )  0
d (t )   sgn[e(t )]  
  e(t )  0
• The output from the comparator will be sampled with a pulse signal at a rate of
1/Ts.
• Next, DM signal will be generated with the equation below:

xDM (t )   sgn[e(t )]   (t  nTs )
n  

   sgn[e(nTs )] (t  nTs )
n  
• The DM signal will be feed back, but before that this signal will be integrated

first ~
m(t )    sgn[e(nTs )]
n  

• This signal will determine the error value e(t).

Pemodulatan Digit
 4.7.1 Delta Modulation (DM) signal

m(t )

~ (t ) Effects of steep
m slope
Δ

Ts

t
0001010111111101100010000000

If e(t) < 0 or -∆ , it will be coded as 0

If e(t) > 0 or +∆, it will be coded as 1

A steep slope results in noise in DM signal. To avoid this from

happening, it has to follow the following condition:

 dm(t )
 mak
Ts dt
Pemodulatan Digit
 4.8 Line Coding
• Binary 1 and 0 in PCM signal can be represented by several formats
known as line coding.

information Line channel

PCM
coder

Reasons for line coding:

1. Synchronization
2. Error detection
3. Error correction

Pemodulatan Digit
4.8.1 Line code format
A. NRZ (Non Return to Zero)
- Popular method
- easy
- Data does not return to 0 in one
clock interval
- No synchronization. Can use ‘start
bit’ for synchronization purposes

1. NRZ-L (NRZ-Level)
1 => High level
0 => Low level
2. NRZ-M (NRZ-Mark)
1 => transition at the starting interval
0 => no transition

3. NRZ-S (NRZ-Space)
1 => no transition
0 => transition at the starting interval
Digital Signal Encoding Formats
 B. RZ (Return to Zero)
• Return to 0 at the half bit interval
• The same
advantages/disadvantages with
NRZ
• Overcome by using bipolar signal
and alternating pulse for
synchronization

4. RZ (Unipolar)
1 => High level
0 => Low level
5. RZ (Bipolar)
1 => Alternately +ve
0 => Alternately –ve
6. RZ (AMI – Alternately Mark Inversion)
1 => Alternately +ve and -ve
Digital Signal Encoding Formats
0 => Low level
 C. Bi phase
• Used in optical communication
system, satellite and video
recorder
• Self synchronizing
7. Bi phase M
1 => transition at the middle of the
interval
0 => no transition at the middle of the
interval
8. Bi phase L (Manchester Coding)
1 => transition from HI to LO at the
middle of the interval
0 => transition from LO to HI at the
middle of the interval
used in Ethernet IEEE 802.3 standard in
LAN
9. Bi phase S – inverse of Bi phase M
Digital Signal Encoding Formats
1 => no transition in the middle of the
interval
0 => transition in the middle of the