CSE4214 Digital Communications CSE4214 Digital Communications

Chapter 2 Formatting
Formatting and Baseband
Modulation

Formatting & Baseband Formatting and Baseband

3 4

Page 1
1
What is Formatting? Block Diagram
Block diagram representing formatting and transmission of baseband signals.
§  Information can take either of the three forms:
1.  Textual information
2.  Analog signals
3.  Digital data
§  Before the signals are transmitted over a digital
communication channel, an information bearing signal
must be converted to digital symbols (Formatting).
§  The resulting digital symbols are then represented by
baseband waveforms (Pulse Modulation or Line
Coding).
5 6

Textual Data (1) Textual Data (2)

American Standard Code for Information Interchange (ASCII) for encoding alphanumerics Extended Binary Coded Decimal Interchange Information (EBCDIC) for encoding alphanumerics

7 7 8

Page 2
2
Message and Symbol Message and Symbol: Example
n  Textual message comprises a sequence of
alphanumeric characters.
n  Example: Hello, how are you.
n  Textual message is converted into a sequence
of bits, i.e. bit stream or baseband signal.
n  Symbols are formed by a group of k bits from a
finite symbol set of M=2k such symbols.
n  A system using a symbol set size of M is
referred to as an M-ary system.
9 10

CSE4214 Digital Communications Periodic Sampling

n  Typically, discrete-time signals are formed by periodically
sampling a continuous-time signal : x(n)=xa(nTs)
The sampling interval Ts is the sampling period, and fs=1/
Ts is the sampling frequency in samples per second.
Formatting Analog Information n  The sampling process:

xa(t)
fs=1/Ts xs(t) Convert impulses
x(n)
into samples

Sa(t)

12

Page 3
3
 xa (t )
Illustration of Ideal Sampling
0 Ts 2Ts 3Ts 4Ts 5Ts 6Ts 7Ts 8Ts 9Ts 10Ts 11Ts
∞
sa (t ) = ∑ δ (t − nTs )
n=−∞

0 Ts 2Ts 3Ts 4Ts 5Ts 6Ts 7Ts 8Ts 9Ts 10Ts 11Ts
∞ ∞
xs (t ) = xa (t ) sa (t ) = ∑ xa (t )δ (t − nTs ) = ∑ xa (nTs )δ (t − nTs )
n=−∞ n=−∞

0 Ts 2Ts 3Ts 4Ts 5Ts 6Ts 7Ts 8Ts 9Ts 10Ts 11Ts 13 14

1 ∞ 2π
X s (ω ) = ∑ X a (ω − nω s ) , ω s =
Fourier Transform of a CT Sampled Signal Ts n=−∞ Ts

n  Fourier transform pair: n  The Fourier transform of the continuous-time
∞
sampled signal X s (ω ) is a periodic function of ω
X (ω ) = ∫ x(t )e − jωt dt consisting of a superposition of shifted replicas
of X a (ω ), scaled by 1/Ts .
−∞

1 ∞
x(t ) = ∫ X (ω )e dω
jωt

2π −∞ For ω s <
> 2ω B X a (ω )
n  Fourier transform of sampled signal : The overlap of the Fourier transform of each of the
terms of the sampled ω
−ω Bsignal
0 ωisB called aliasing
1 ∞ 2π X s (X
ωs)(ω )
X s (ω ) = ∑ X a (ω − nω s ) , ω s = n=-2 n=-2n=-1 n=-1 1/Tss n=0
1/T n=0 n=1 n=1 n=2 n=2
Ts n=−∞ Ts
ω
ω
−ω s −ω s 00 ωs ωs
15 16
ω s − ω sB − ω B

Page 4
4
Sampling Theorem : Example

n  A bandlimited continuous-time signal, with

highest frequency(bandwidth) B Hz, can be
uniquely recovered from its samples provided
that the sampling rate Fs ≥ 2B samples per
second.
n  The frequency Fs = 2B is called the Nyquist
sampling frequency.
n  If the signal is sampled at less than the Nyquist
rate, then the aliasing occurs.
17 18

Natural Sampling Illustration of Natural Sampling

n  Replace impulse train in ideal sampling with a
pulse train p(t) (also know as the gating
waveform).
n  The pulse train

where
n  The pulse train can be implemented by an on/off
switch.
19 20

Page 5
5
Analog-to-Digital Conversion Anti-aliasing Filter
n  The role of anti-aliasing filter is to cut off the
n  Components : anti-aliasing filter, sample and frequency components that is higher than the half of
sampling frequency.
hold, analog-to-digital converter (quantization). n  Ideally, the anti-aliasing filter should have a lowpass
frequency response,

Anti- Sample Analog ⎧1, Ω < ΩT / 2

H a ( jΩ) = ⎨
Aliasing and to Ω ≥ ΩT / 2
Filter Hold Digital
⎩ 0,
Such a “brickwall” filter can’t be realized using
practical analog circuit, hence, must be
Block Diagram of an ADC approximated.
21 22

Anti-aliasing Filter’
’s Effect on Signal Band Anti-Aliasing Filter Design
n  Requirement :
X a ( jΩ ) 1.  Approximate linear phase in passband
H a ( jΩ ) 2.  Passband edge > highest frequency in signal
1 3.  Stopband edge < half of sampling frequency
Spectrum
n  Four types of analog filter
of aliased
component 1.  Butterworth filter : good passband, slow roll-off
of input 2.  Chebyshev filter : good roll-off and linear phase
3.  Elliptic filter : fast roll-off, non-linear phase
4.  Bessel filter : close to linear phase, wide transitionband
1/A
Ω n  Design can be done in Matlab
0 Ωp ΩT Ω0 ΩT
2 = ΩT − Ω p 23 24

Page 6
6
Frequency Response of 4 Types of Filter Sample and Hold
n  Sample and hold is the most popular sampling
method.
n  Involves two operations:
n  Sample and hold

25 26

Sample and Hold Circuit Analog-to-Digital Converter

n  Samples the analog signal at uniform intervals n  Converts an analog signal into a binary
and holds the sampled value after each coded digital signal.
sampling operation for sufficient time for n  Types of A/D converter
accurate conversion by the A/D converter. 1.  Integrating converter
2.  Successive approximation converter
Hold
3.  Flash converter
Input Sample
+ Output 4.  Folding A/D converter
5.  Pipelined A/D converter
-
Clock
27 28

Page 7
7
 CSE4214 Digital Communications A/D Conversion
n  Uniform quantizer
n  Peak signal power
to average
quantization noise
power is:
Quantization
⎛ S ⎞ 2
⎜ ⎟ ≤ 3L
N
⎝ ⎠ q
n  SNR increases as a
function of the
number of
quantization level
squared.
30

Examples of Sampling Pulse Code Modulation (PCM)

n  In pulse modulation, some parameter of a pulse train is
varied in accordance with the sample values of a
message signal.
n  Pulse-amplitude modulation (PAM)
n  Amplitudes of regularly spaced pulses are varied.

Original analog waveform

n  Pulse-width modulation (PWM)
Natural-sampled data
n  Widths of the individual pulses are varied.
n  Pulse-position modulation (PPM)
n  Position of a pulse relative to its original of occurrence is
varied.
n  Pulse modulation techniques are still analog
modulation. For digital communications of an analog
Quantized samples Sample and Hold 31 32
source, quantization of sampled values is needed.

Page 8
8
 PCM PCM - Example
n  A PCM signal is obtained from the quantized PAM
signal by encoding each quantized sample to a digital
codeword
n  In binary PCM each quantized sample is digitally
encoded into an R-bit binary codeword.
n  Binary digits of a PCM signal can be transmitted using
many efficient modulation schemes.

33 34

PCM Waveform Example Uniform Quantization (1)

§  For most voice
communications, very
PCM sequence low speech volumes
predominate.
§  Large amplitudes are
very rare while low
amplitudes are more
often
Pulse representation

Pules waveform 35 36

Page 9
9
 Uniform Quantization (2) Nonuniform Quantization (1)
§  Nonuniform quantizers are used for speech signals, which
—  Using a uniform quantizer for speech signals provides coarse
provide coarse quantization at high amplitudes and fine
quantization at low amplitudes
quantization at low amplitudes.
§  Nonuniform quantization is achieved by the process of
companding followed by uniform quantization.

37 38

Nonuniform Quantization (2) CSE4214 Digital Communications

§  Two commonly used companders are:

µ − law compander A − law compander
⎧ A(| x | / xmax ) | x| 1
⎪⎪ y max sgn( x) 0< ≤
log e [1 + µ(| x | / xmax )] 1 + log e A xmax A
y = ymax sgn( x) y = ⎨
log e [1 + µ] 1 + log e [A(| x | / xmax )] 1 | x|
⎪ ymax sgn( x) < ≤1
1 + log e A A xmax
⎪⎩
Baseband Transmission

39

Page 10
10
PCM Waveform Types PCM Coding (1)
n  Nonreturn-to-zero (NRZ) 1 = 1 voltage level,
0=another voltage level
n  NRZ is most commonly used PCM waveform
n  NRZ-L (L for level) 1 (mark)=change in level,
0 (space) = no change in level
n  NRZ-M (M for mark)
n  NRZ-S (S for space) A complement of NRZ-M

n  Return-to-zero (RZ) 1=half-bit-wide pulse,

0=no pulse
n  Unipolar-RZ, bipolar-RZ, RZ-AMI(alternate mark
inversion) 1 and 0 by opposite-level pulses

n  Phase encoded

1=equal amplitude alternating pulse
n  Multilevel binary 0=no pulse
41 42

PCM Coding (2) Bits per PCM Word and Bits per Symbol
n  PCM word size
n  How many bits shall we assign to each analog
sample?
e ≤ pVpp e: quantization error,
q Vpp Vpp peak-to-peak voltage
emax = = q: quantization level
2 2L
Vpp 1
≤ pVpp → 2 l = L ≤
2L 2p
$ 1 '
l ≥ log 2 & )
43 %2p( 44

Page 11
11
M-ary Pulse-Modulation Activity 1
n  Multilevel signaling - a group of k-bit is The information in an analog waveform, with maximum frequency
fm=3kHz, is to be transmitted over an M-ary PAM system, where
transmitted by M=2k level pulse. the number of pulse levels is M=16. The quantization error is
specified not to exceed (+/-)1% of the peak-to-peak analog signal.
(a)  What is the minimum number of bits/samples, or PCM word size
that should be used in digitizing the analog waveform?
(b)  What is the minimum required sampling rate, and what is the

resulting bit transmission rate?

(c)  What is the PAM pulse or symbol transmit rate?

(d)  If the transmission bandwidth equals 12 kHz, determine the

bandwidth efficiency for this system.

45 46

Page 12
12