EC6501

DIGITAL COMMUNICATION
 OBJECTIVES:
To know the principles of sampling &
quantization
To study the various waveform coding
schemes
To learn the various baseband transmission
schemes
To understand the various Band pass signaling
schemes
To know the fundamentals of channel coding
 SYLLABUS
UNIT I SAMPLING & QUANTIZATION 9
Low pass sampling – Aliasing- Signal Reconstruction-Quantization - Uniform & non-uniform
quantization - quantization noise - Logarithmic Companding of speech signal- PCM - TDM 56

UNIT II WAVEFORM CODING 9

Prediction filtering and DPCM - Delta Modulation - ADPCM & ADM principles-Linear Predictive
Coding

UNIT III BASEBAND TRANSMISSION 9

Properties of Line codes- Power Spectral Density of Unipolar / Polar RZ & NRZ – Bipolar NRZ -
Manchester- ISI – Nyquist criterion for distortionless transmission – Pulse shaping – Correlative
coding - Mary schemes – Eye pattern - Equalization

UNIT IV DIGITAL MODULATION SCHEME 9

Geometric Representation of signals - Generation, detection, PSD & BER of Coherent BPSK,
BFSK & QPSK - QAM - Carrier Synchronization - structure of Non-coherent Receivers - Principle
of DPSK.

UNIT V ERROR CONTROL CODING 9

Channel coding theorem - Linear Block codes - Hamming codes - Cyclic codes - Convolutional
codes - Vitterbi Decoder

TOTAL: 45 PERIODS
 OUTCOMES
Upon completion of the course, students will be
able to
Design PCM systems
Design and implement base band transmission
schemes
Design and implement band pass signaling
schemes
Analyze the spectral characteristics of band
pass signaling schemes and their noise
performance
Design error control coding schemes
 EC6501
DIGITAL COMMUNICATION

UNIT - 1
INTRODUCTION
 UNIT I
SAMPLING & QUANTIZATION (9)
 Low pass sampling
 Aliasing
 Signal Reconstruction
 Quantization
 Uniform & non-uniform quantization
 Quantization Noise
 Logarithmic Companding of speech signal
 PCM
 TDM
 Digital communication system

Input Low
Signal Source Channel
Pass Sampler Quantizer Multiplexer
Analog/ Encoder Encoder
Filter
Digital
Carrier

Pulse
Line
To Channel Modulator Shaping
Encoder
Filters

De- Receiver
From Channel Detector
Modulator Filter

Carrier Ref.

Signal
Digital-to-Analog Channel De-
at the
Converter Decoder Multiplexer
user end

12
 Key Questions

 How can a continuous wave form be

converted into discrete samples?

 How can discrete samples be converted back

into a continuous form?
 Low Pass Sampling

Sampling (in time) is

 Measure amplitude at regular intervals

 How many times should we sample?

 Nyquist Theorem
For lossless digitization, the sampling rate
should be at least twice the maximum
frequency of the signal to be sampled.

 In mathematical terms:
fs > 2*fm

 where fs is sampling frequency and fm is the

maximum frequency in the signal
 Limited Sampling
 But what if one cannot sample fast
enough?
 Limited Sampling
 Reduce signal frequency to half of
maximum sampling frequency

 low-pass filter removes higher-frequencies

 (e.g.) If max sampling frequency is 22kHz, the it

is a must to low-pass filter a signal down to
11kHz
Aliasing effect

LP filter

Nyquist rate

aliasing
 Three different sampling methods
Practical Sampling Methods are Natural Sampling
and Flat-top Sampling
Natural Sampling
 Pulse-Amplitude Modulation

• Pulse-Amplitude Modulation (PAM)

– The amplitude of regularly spaced pulses are
varied in proportion to the corresponding sample
values of a continuous message signal.
– Two operations involved in the generation of the
PAM signal
• Instantaneous sampling of the message signal m(t)
every Ts seconds,
• Lengthening the duration of each sample, so that it
occupies some finite value T. Fig. 5

28
 Back Next
Fig.5

29
Fig.6 Back Next

30
 Back Next
Fig.7

31
• The advantages offered by digital pulse modulation
– Performance
• Digital pulse modulation permits the use of regenerative repeaters,
when placed along the transmission path at short enough distances,
can practically eliminate the degrading effects of channel noise and
signal distortion.
– Ruggedness
• A digital communication system can be designed to withstand the
effects of channel noise and signal distortion
– Reliability
• Can be made highly reliable by exploiting powerful error-control
coding techniques.
– Security
• Can be made highly secure by exploiting powerful encryption
algorithms
– Efficiency
• Inherently more efficient than analog communication system in the
tradeoff between transmission bandwidth and signal-to-noise ratio
– System integration
• To integrate digitized analog signals with digital computer data
32
 Quantization Process
• Amplitude quantization
– The process of transforming the sample amplitude m(nTs) of a
baseband signal m(t) at time t=nTs into a discrete amplitude
v(nTs) taken from a finite set of possible levels.

I k : {mk  m  mk 1}, k  1,2,..., L (17) Fig. 9

– Representation level (or Reconstruction level)

• The amplitudes vk , k=1,2,3,……,L
– Quantum (or step-size)
• The spacing between two adjacent representation levels

v  g (m ) (18) Fig. 10

33
 Back Next
Fig.9

34
 Fig.10 Back Next

Two types of quantization are

a) Mid-tread
b) Mid-rise 35
 Linear Quantization
• Applicable when the signal is in a
finite range (fmin, fmax)
• The entire data range is divided
into L equal intervals of length Q
(known as quantization interval or
quantization step-size)
• Q=(fmax-fmin)/L Interval i is
mapped to the middle value of this
interval
• We store/send only the index of
quantized value min
Signal Range is Symmetric
Quantization Noise
 Non-Uniform Quantization
 Many signals such as speech have a nonuniform distribution.
– The amplitude is more likely to be close to zero than to be at higher levels.
 Nonuniform quantizers have unequally spaced levels
– The spacing can be chosen to optimize the SNR for a particular type of signal.

Output sample
XQ 6

2 Example: Nonuniform 3 bit quantizer

-8 -6 -4 -2 2 4 6 8

-2
Input sample
X
-4

-6

39
 Non-Linear Quantization
• The quantizing intervals are not of equal size
• Small quantizing intervals are allocated to small
signal values (samples) and large quantization
intervals to large samples so that the signal-to-
quantization distortion ratio is nearly independent of
the signal level
• S/N ratios for weak signals are much better but are
slightly less for the stronger signals
• “Companding” is used to quantize signals
Function representation
Uniform and Non-uniform Quantization
 Companding
• Formed from the words compressing and
expanding.
• A PCM compression technique where analogue
signal values are rounded on a non-linear scale.
• The data is compressed before sent and then
expanded at the receiving end using the same
non-linear scale.
• Companding reduces the noise and crosstalk
levels at the receiver.
 u-LAW and A-LAW definitions
• A-law and u-law are companding schemes used in
telephone networks to get more dynamics to the
8 bit samples that is available with linear coding.
• Typically 12..14 bit samples (linear scale) sampled
at 8 kHz sample are companded to 8 bit
(logarithmic scale) for transmission over 64 kbit/s
data channel.
• In the receiving end the data is then converted
back to linear scale (12..14 bit) and played back.
converted back
– Compressor
• A particular form of compression law : μ-law

log(1   m )
v (5.23)
log(1   )

d m log(1   )
 (1   m ) (5.24)
dv 
• μ-law is neither strictly linear nor strictly logarithmic

• A-law :

 Am 1
1  log A , 0  m  A

v  (5.25)
1  log( A m ) , 1  m  1
 1  log A A
1  log A 1
, 0  m 
d m  A A
 (5.26)
dv  1 Fig. 5.11
(1  log A) m , A  m  1
45
Fig.11 Back Next

46
 Example: -law Companding
1

0.5

x[n]=speech /song/ 0

-0.5

-1
0 1000 2000 3000 4000 5000 6000 7000 8000 9000 10000

0.5

y[n]=C(x[n]) 0

-0.5
Companded Signal -1
0 1000 2000 3000 4000 5000 6000 7000 8000 9000 10000

0.5
Close View of the Signal
Segment of x[n] 0

-0.5

-1
2200 2300 2400 2500 2600 2700 2800 2900 3000

Segment of y[n]
0.5

Companded Signal -0.5

-1
2200 2300 2400 2500 2600 2700 2800 2900 3000

Eeng 360 47
 A-law and -law Companding
• These two are standard companding methods.
• u-Law is used in North America and Japan
• A-Law is used elsewhere to compress digital telephone signals

Eeng 360 48
Quantization - why do we need such classification ?! - (3)

Comparison – Uniform Vs. Non-Uniform Usage

 Speech signals doesn’t require high quantization resolution for

high amplitudes (50% Vs. 15%).
 wasteful to use uniform quantizer ?
The goal is decrease the SQNR, more levels for low amplitudes, less levels for
high ones.
 Maybe use a Non-uniform quantizer ?

Technical Presentation  Page 49

Concepts

Quantization
More About Non-Uniform Quantizers (Companding)
 Uniform quantizer = use more levels when you need it.
 The human ear follows a logarithmic process in which high amplitude sound doesn’t
require the same resolution as low amplitude sounds.
 One way to achieve non-uniform quantization is to use what is called as “Companding”
 Companding = “Compression + Expanding”

Compressor Uniform
Expander
Function Quantization
Function

(-1)

Technical Presentation  Page 50

 Pulse-Code Modulation
• PCM (Pulse-Code Modulation)
– A message signal is represented by a sequence of coded pulses, which is
accomplished by representing the signal in discrete form in both time and
amplitude
– The basic operation
• Transmitter : sampling, quantization, encoding
• Receiver : regeneration, decoding, reconstruction

• Operation in the Transmitter

1. Sampling
1. The incoming message signal is sampled with a train of rectangular pulses
2. The reduction of the continuously varying message signal to a limited
number of discrete values per second
2. Nonuniform Quantization
1. The step size increases as the separation from the origin of the input-
output amplitude characteristic is increased, the large end-step of the
quantizer can take care of possible excursions of the voice signal into the
large amplitude ranges that occur relatively infrequently.
51
Fig.11 Back Next

52
3. Encoding
1.To translate the discrete set of sample vales to a
more appropriate form of signal Fig. 11
2.A binary code
 The maximum advantage over the effects of noise in a
transmission medium is obtained by using a binary
code, because a binary symbol withstands a relatively
high level of noise.
 The binary code is easy to generate and regenerate
Table. 2

53
• Regeneration Along the Transmission Path
– The ability to control the effects of distortion and noise produced by
transmitting a PCM signal over a channel
– Equalizer
• Shapes the received pulses so as to compensate for the effects of
amplitude and phase distortions produced by the transmission
– Timing circuitry
• Provides a periodic pulse train, derived from the received pulses
• Renewed sampling of the equalized pulses
– Decision-making device Fig. 13
• The sample so extracted is compared o a predetermined threshold
– ideally, except for delay, the regenerated signal is exactly the same as the
information-bearing signal
1. The unavoidable presence of channel noise and interference causes
the repeater to make wrong decisions occasionally, thereby
introducing bit errors into the regenerated signal
2. If the spacing between received pulses deviates from its assigned
value, a jitter is introduced into the regenerated pulse position,
thereby causing distortion.

54
 Back Next
Fig.13

55
• Operations in the Receivers

1. Decoding and expanding

1.Decoding : regenerating a pulse whose amplitude is
the linear sum of all the pulses in the code word
2.Expander : a subsystem in the receiver with a
characteristic complementary to the compressor
1. The combination of a compressor and an expander is a
compander

2. Reconstruction
1.Recover the message signal : passing the expander
output through a low-pass reconstruction filter
56
Categories of multiplexing
Time Division Multiplexing (TDM)

TDM is a technique used for

transmitting several message signals
over a single communication channel
by dividing the time frame into slots,
one slot for each message signal
 Time Division Multiplexing
• Entire spectrum is allocated for a channel (user) for a limited time.
• The user must not transmit until its
next turn.
k1 k2 k3 k4 k5 k6
• Used in 2nd generation
c
Frequency
f

t
• Advantages: Time
– Only one carrier in the medium at any given time
– High throughput even for many users
– Common TX component design, only one power amplifier
– Flexible allocation of resources (multiple time slots).
 Time Division Multiplexing
• Disadvantages
– Synchronization
– Requires terminal to support a much higher data
rate than the user information rate therefore
possible problems with intersymbol-
interference.

• Application: GSM
 GSM handsets transmit data at a rate of 270 kbit/s in
a 200 kHz channel using GMSK modulation.
 Each frequency channel is assigned 8 users, each
having a basic data rate of around 13 kbit/s
 Time Division Multiplexing
At the Transmitter
Simultaneous transmission of several signals on a time-sharing basis.
 Each signal occupies its own distinct time slot, using all frequencies, for
the duration of the transmission.
 Slots may be permanently assigned on demand.

At the Receiver
 Decommutator (sampler) has to be synchronized with the incoming
waveform  Frame Synchronization
 Low pass filter
 ISI – poor channel filtering
 Feedthrough of one channel's signal into another channel -- Crosstalk

Applications of TDM: Digital Telephony, Data communications, Satellite

Access, Cellular radio.
61
 Time Division Multiplexing

Conceptual diagram of multiplexing-demultiplexing.

PAM TDM System

62
TDM-PAM: Transmitter
TDM-PAM : Receiver
 Samples of Signal -1

g1(t)

time
0 Ts 2Ts
Samples of signal - 2

g2(t)

Ts Ts
 Multiplexing of TWO signals

0 Ts 2Ts
 TDM-PAM for 4 signals.

4
4
4

1 1 1
2 2 2
3 3 3
Time
 Problem
Two low-pass signals of equal bandwidth
are sampled and time division
multiplexed using PAM. The TDM signal is
passed through a Low-pass filter & then
transmitted over a channel with a
bandwidth of 10KHz.
Continued….
 Problem (continued…)

a) What is maximum Sampling rate for each

Channel?
b) What is the maximum frequency content
allowable for each signal?
 Problem: Solution

Channel Bandwidth = 10 KHz.

Number of samples that can be transmitted
through the channel = 20K
Maximum Sampling rate for each channel =
10K Samples/sec.
Maximum Frequency for each Signal = 5KHz
End of Unit-1