INTRODUCTION

TO DIGITAL MODULATION
The purpose of analog modulation is to impress
an
information-bearing analog waveform onto a
carrier for transmission.
The purpose of digital modulation is to convert an
information-bearing discrete-time symbol
sequence into a continuous-time waveform
(perhaps impressed on a carrier).
Key concerns : bandwidth efficiency and
implementation complexity.


Digital Modulation is defined as the transmission
of digitally modulated analog signals (carriers)
between 2 or more points.

If the information signal is digital and the carrier
has the following expression:
v(t) = V

sin(2tft + u)

Ac(t): amplitude modulation AM ASK
fc(t) : frequency modulation FM FSK
u(t) : phase modulation PM PSK

Ac(t) and u(t) QAM (Digital)


Digital Analog
Application of Digital Modulation :
1. Modem in personal computer
2. Digital Subscriber Lines (DSL)
3. Digital Microwave
4. Satellite Communication system
5. Cellular telephone Personal Communication
Systems (PCS)

Digital Modulation involves the Digital Signal
Processing because it is much simpler to store
digital signals than analog signal.

There are 2 type of Digital Modulation:
1. Pulse Modulation PAM, PWM, PPM
2. Binary Modulation ASK, FSK, PSK, PCM


Model of Digital Comm.
The model of Digital System is shown below:




Input Signal
TRANSMITTER




Output Signal
RECEIVER



Source
Encoder
Digital
Demodulation
Trans. Medium
or Channel
Channel
Encoder
Digital
Modulation

Source
Decoder

Channel
Decoder
Block diagram of a digital radio
system
Signal Processing
Advantages & disadv. of Digital Transmission
Advantages
Noise immunity
(Time domain) Multiplexing
Regeneration
Simple to evaluate and measure

Disadvantages
More bandwidth
Additional encoding (A/D) and decoding
(D/A) circuitry
The information capacity of a communication system
represents the number of independent symbols that can be
carried through the system in a given unit of time.

By using Shannon limit for information capacity, the
relationship between Information capacity to the signal
bandwidth and SNR is defined below:





I = information capacity (bit/second)

B = system bandwidth (Hertz)

S/N =signal-to-noise power ratio (dimensionless)
2 10
log 1 3.32 log 1
S S
I B B
N N
| | | |
= + = +
| |
\ . \ .
Bandwidth and information capacity
M-ary Encoding

2
log N M =
2
N
M =
N = number of bits necessary

M = number of conditions, levels, or
combinations possible with N bits

For binary M = 2 N = 1
1
baud
s
s
f
t
= =
baud = f
s
= symbol rate (baud per second)
t
s
= time of one signaling element (second)

symbol = one signaling element

What is the difference between baud and bits per
second (bps)?
Baud and Minimum Bandwidth
Baud is rate of change of a signal on the
transmission medium after encoding and
modulation have occurred (symbols per seconds),
rate of change at output of the Modulator.

Bit is rate of change at the input to the Modulator
(bits per second, bps).

Binary signals; logic 1 => High, logic 0 => Low


Baud and Minimum Bandwidth
2
2 2 log
b s
f f N BN B M = = =
f
b
= bitrate (bps)
f
s
= baud (symbols per second)
B = minimum Nyquist bandwidth (hertz)
M = number of discrete signals or voltage levels
N = number of bits encoded into each symbol
2
2log 2
b b
f f
B
M N
| |
= =
|
\ .
EXAMPLE 1
A standard voice-band communication channels
have a SNR power of 1000 (30 dB) and signal
Bandwidth of 2.7 kHz. Determine the information
capacity.


Solution :
By using Shannons Limit, information
capacity is defined as:
I = (2.7 kHz)(3.32)log
10
(1 + 1000) = 26.9
kbps
Pulse Modulation
Pulse Modulation is a process of sampling
analog signal and then converting them
into discrete pulses and transporting the
pulses from a source to a destination over
a transmission medium. A device to
perform this is called ADC (Analog-to-
Digital Converter) & DAC (Digital-to-
Analog Converter).





PAM (Pulse Amplitude Modulation)
It is used to describe the conversion of analog
signal to pulse-type signal in which the amplitude
of the pulse denotes the analog information. In
addition, it is a series of pulses in which the
amplitude of each pulse represents the amplitude
of the information signal at a given time.

Pulse Modulation
PWM (Pulse Width Modulation)
It is a pulse duration modulation (PDM) or
pulse length modulation. The width of pulse is
varied proportional to the Amplitude of the
analog signal at the time signal is sampled.

PPM (Pulse Position Modulation)
It is a series of pulses in which the timing of
each pulse represents the amplitude of the
information signal at a given time.

PCM (Pulse Code Modulation)
It is a series of pulse in which the amplitude of
the information signal at a given time is coded
as a binary number. The pulses are of fixed length
and fixed amplitude. Refer to Figure 10-1 in the
textbook for PWM, PPM, PAM & PCM.

PCM is generated by 3 processes; Sampling,
Quantization & Encoding.

An Integrated circuit that perform PCM
encoding and decoding function is called
CODER OR DECODER.

Pulse Modulation
Analog signal
Sample pulse
Pulse width
modulation
Pulse position modulation
Pulse amplitude
modulation
Pulse code
modulation
8 bit
t
s

PCM system Block Diagram

Signal Sampling
A process of periodically sampling the
continually changing analog input voltage and
convert it to a series of constant amplitude
pulses, so that it is easier to convert to binary
PCM code

2 types of sampling:
1. Natural Sampling
tops of the sample pulses retain their natural
shape, making it difficult for ADC to convert to
PCM codes
2. Flat-top Sampling
input voltage is sampled with narrow pulses and
then held relatively constant until next sampling


Natural Sampling
Input analog signal
Sampling pulse
Sampled output
Sample-and-hold circuit
Flat-top sampling
Sample-and-hold waveforms
Sampling Rate
Sample & Hold circuit is a vital component in ADC to
convert the analog signal to digital signal at certain
sampling period, T
s
.

The analog signal should be sampled at the Nyquist
rate Sampling Frequency,

2
s a
f f >
f
s
= minimum Nyquist sample rate (Hz)
f
a
= maximum analog input frequancy (Hz)
Number of Level
The number of level depends on the number of bits
used to express the sample value. It is defined
as :

N = 2
n

EXAMPLE :
Calculate the number of level if the number of
bit per sample is 8-bit.

Solution :
1. N = 2
8
= 256
Output spectrum of a S&H circuit
S&H circuit is nonlinear which results in harmonics
Ideal
Aliasing
2
s a
f f =
2
s a
f f <
http://www.mustagh.com/alias/Alias.html
Quantization
Quantization is a process of rounding off the amplitude of
flat-top samples to a manageable number of levels

Assigning PCM codes to absolute magnitude is called
quantizing

Total range is sub-divided into a smaller number of sub
ranges.

Magnitude difference between adjacent steps is called the
quantization interval or quantum

The magnitude of a quantum is called the resolution: equal
to the voltage of the minimum step size = voltage of the
least significant bit. minimum voltage other than 0 that
can be decoded by ADC.




The binary codes used for PCM are n-bit codes (sign-
magnitude code) where the MSB bit is the sign bit. If PCM is
3-bit codes, then the sign and magnitude are shown below:













In terms of Voltage, the maximum signal voltages are 3 V or -3
V and the minimum signal voltages are 1 V or -1 V.
Sign Magnitude Decimal
value
Quantization range
(V)
1 1 1 +3 +2.5 to +3.5
1 10 +2 +1.5 to +2.5
1 01 +1 +0.5 to 1.5
1 00 +0 0 to +0.5
0 00 -0 0 to -0.5
0 01 -1 -0.5 to -1.5
0 10 -2 -1.5 to -2.5
0 11 -3 -2.5 to -3.5
Folded binary code
Input analog signal
Sampling pulse
PCM code
Quantization
PAM signal
What is the PCM code for 2.6 V??
Quantization Error
Folded PCM code = sample voltage
resolution
For input at 2.6 V, the PCM code is therefore:
2.6/1 = 2.6
But since there is no code for +2.6, the magnitude is
rounded off to the nearest valid code, which is 111
(+3V)
Thus there is difference of 0.4
QUANTIZATION ERROR (Q
e
)
or also known as quantization noise (Q
n
)
Maximum magnitude Q
e
is equal to one-half a
quantum


resolution
2
e
Q =
Linear input-output transfer curve
Linear
Quantization
Error
Input analog signal
Sampling pulse
PCM code
PAM signal
Question: What is the quantized voltage, quantization error and PCM
code for 1.75 V??
Dynamic Range
max max
min
2 1
resolution
n
V V
DR
V
= = =
DR = dynamic range (unitless)
V
min
= the quantum value
V
max
= the maximum voltage magnitude of the DACs
n = number of bits in a PCM code (excl. sign bit)
( )
( )
20log 2 1
n
dB
DR =
Ratio of the largest possible magnitude to the
smallest (other than 0) magnitude that can be
decoded by the digital-to-analog converter (DAC)
in the receiver
DR = 2
n
-1

Thus 2
n
= DR + 1

And therefore, The minimum number of bit used:
n = log ( DR + 1 )
log 2
2 1 2
n n
DR = ~
For n > 4
( )
( )
20log 2 1 20 log2 6
n
dB
DR n n ~ = ~
No of Bits No of Levels DR (dB)
1 2 6.02
2 4 12
3 6 18.1
4 16 24.1
5 32 30.1
6 62 36.1
7 128 42.1
8 256 48.2
9 512 54.2
10 1024 60.2
11 2048 66.2
12 4096 72.2
13 8192 78.3
14 16348 84.3
15 32768 90.3
16 65536 96.3
Dynamic Range
Coding Efficiency
minimum number of bits
coding efficiency= 100
actual number of bits

Coding efficiency is a numerical indication of
how efficiently a PCM code is utilized
EXAMPLE
A PCM systems has the following specification:
Maximum Analog Input Frequency = 4 kHz
Maximum decoded voltage at the receiver = 2.55 V
The dynamic range = 46 dB

Determine the following :
(a) Minimum Sampling Rate
(b) Minimum number of bits used in PCM code
(c) Resolution
(d) Quantization Error
Solution
(a) The minimum sampling rate:
f
s
= 2f
a
= 2(4 kHz) = 8 kHz

(b) Calculate the Dynamic range :
46 = 20log(V
max
/ V
min
)
V
max
/ V
min
= antilog (46/20) = 199.5
Thus, the minimum number of bit used:
n = log (199.5 + 1) / Log 2 = 7.63

(c) Resolution is defined as:
V
max
/ 2
n
- 1 = 0.01 V

(d) Quantization Error :
Q = resolution / 2 = 0.01 V / 2 = 0.005 V
Signal-to-Quantization Noise Efficiency
( )
min
min
resolution
2
e e
V
SQR
Q Q
= = =
resolution
2
e
Q =
V
2
e
SQR
Q
= =
( )
max
max
e
V
SQR
Q
=
SQR is not constant
Linear vs. Nonlinear coding
Linear Nonlinear
What is the advantage of Nonlinear coding???
Idle channel noise
Companding
Higher amplitude
analog signals are
compressed


Dynamic range is
improved
PCM system with analog companding
-law compression
( )
max
max
ln 1
ln 1
in
out
V
V
V
V

| |
+
|
\ .
=
+
The 12-bit PCM can be compressed to 8-bit PCM.
The 8-bit compressed code consists of a sign bit,
3 segment identifier and 10-bit magnitude code
that specifies the quantization level
a) Sign Bit => 1 = +
=> 0 = -
b) No. of leading 0s, subtract from 7:-Bit Segment
Identifier
000 111
c) 4-Bit Quantization Interval
A B C D 0000 to 1111
Digital Companding
Refer to Figure 10-18(a), (b), (c) for the conversion of
12-bit PCM to 8-bit compressed code (Encoded PCM)
and vise versa (Decoded PCM).

EXAMPLE :
Convert the 12-bit PCM below to an 8-bit compressed
code
100110100100


Solution :
1. From Figure 10-18 (b), use segment 5,
ABCD will be 1010.
2. From segment 5, the 8-bit compressed code is
s101ABCD where s is a sign bit either 1 or 0.
3. Thus the 8-bit compressed code will be
Sign bit
Leading
zeros
ABCD
11011010
Example
(i) Determine the 12-bit code for analog signal of
+0.32V using a PCM system with 0.01V resolution.
Then convert the 12 bit code to 8-bit compressed
code
(ii) Repeat the question with -0.072V analog input
ANSWER:
(i) 10100000
(ii)00000111
Delta Modulation PCM
Delta Modulation use a single-bit PCM code to achieve
digital transmission of analog signal.

Refer to Figure 10-20 in the textbook for Delta
Modulation Transmitter and Figure 10-22 for
Receiver.

When the analog signal is sampled by Sample &
Hold circuit, it will create PAM (Pulse Amplitude
Modulation) and compared with initial condition
(zero volts => up-down converter is zeroed).
The output of Comparator is logic 1. On the next
clock pulse, up-down counter is incremented to a
count of 1.

Each time up-down converter is incremented, logic
1 is transmitted. Each time up-down converter is
decremented, logic 0 is transmitted.
It is a delta modulation system where the step
size of the DAC is automatically varied according
to previous values (refer to Figure 10-25 in the
textbook)

An Adaptive Delta Modulation can transmit voice
at about bit rate of the PCM system.

The advantage of using Adaptive Delta
Modulation:
Reduce Slope Overload The slope of the analog
is greater than the delta modulator can maintain
Reduce Granular Noise The variation of the
constructed signal compared to original signal.


Adaptive Delta Modulation PCM
PCM Advantages
PCM systems allow the regeneration of the
signal.

PCM systems allow the analog information to be
stored digitally (digital storage).

PCM systems allow the source and channel
coding.

PCM systems allow successive time
multiplexing of lower-rate digital streams