Digital Communication

Chapter 1
Conversion of Analog to Digital
Basic Block Diagram of Digital
Communication System:
 Analog to Digital (A/D) converter
• The A/D converter is a device gives a binary code word (digital)
corresponding to the continuous-time (analog) input signal at that time.
• It is a three step process:
1. Sampler
2. Quantizer
3. Encoder
What is Sampling?
• It is a process of converting a continuous time signal into a discreate-time signal
by taking "samples" of the continuous time signal at discrete time instants.
• Continuous time signal: The signal value is defined at every instant of time.
• Discreate-time signal: The signal value is defined at periodic interval of time.

where Ts is sampling period

Pulse Train Signal:
• A repetitive series of pulses, separated in time by a constant interval.
Time-domain representation of Pulse Train

Fourier series representation of pulse train

 How we do Sampling?
• Basic Math:
zX1=z zX0=0
Any number multiplied with 1 results in same number.
Any number multiplied with 0 results in 0.

• Sampling of a signal is done by multiplying the signal with a pulse train.

Sampling Theorem
• Let, be the sampled version of a signal ,

• From a sampled version of a signal , having maximum bandwidth , can

be recovered back, if the sampling rate of the signal is:

• is called Nyquist rate of sampling

• If , then the recover process introduces distortion, called aliasing
distortion(error)
Proof of sampling Theorem
• Consider a continuous time signal s(t).
• The spectrum of s(t) is a band limited to fm Hz i.e. the
spectrum of s(t) is zero for |ω|>ωm.
• Sampling of input signal s(t) can be obtained by multiplying
s(t) with an impulse train δ(t) of period Ts.
• The output of multiplier is a discrete signal called sampled
signal
• Sampled signal

Spectrum of the sample output signal

Anti-aliasing filter
 To avoid aliasing distortion, all signal components having frequency above need to be
removed.

 Hence, the signal passes through a lowpass filter called anti-aliasing filter before the
sampler.
Reconstruction of signal from its samples:
Original signal can be reconstructed from its samples by using a linear low
pass filter.
Pulse Amplitude Modulation (PAM)
• In this, the amplitude of the pulse carrier varies proportional
to the instantaneous amplitude of the message signal.
• It is an outcome of sampling process.
• It is of two types:
1. Natural PAM: the tops of the pulses follow the shape of the
modulating signal.
2. Flat-top PAM: the tops of the pulses remain flat to the
value at of the modulating signal at the start of the pulse.
Pulse Width Modulation (PWM)
• It is a modulation technique that
generates variable-width pulses to
represent the amplitude of
an analog input signal.

• It is a powerful technique for

controlling analog circuits with a
microcontroller's digital outputs.
Pulse Position Modulation (PPM)
• It is a modulation technique that allows variation in the position of the pulses according to the
amplitude of the sampled modulating signal
Quantization
• It is the process of mapping of a discrete-time continuous-valued signal into a
discrete-time, discrete-valued signal.

• The value of each signal sample is represented by a value selected from a finite set
of possible values.

• The sequence of quantized samples at the output of the quantizer.

where ℚ() is quantizer operation.

• The quantization error is defined as the difference between the quantized value
and the actual sample value.
• The values allowed in the digital signal are called the quantization levels.

• The distance Δ between two successive quantization levels is called the

quantization step size or resolution.

• The amplitude of each discreate-time samples x(n) are rounded to its

nearest quantization level

• The instantaneous quantization error cannot exceed half of the quantization

step.

• If L is the number of quantization levels, then

• Theoretically, quantization of
analog signals always results in a
loss of information.

• The quantization error depends

on step size.

• Increasing the number of

quantization levels L results in a
decrease of the quantization step
size.

• Hence, less error with more

number quantization levels. Analysis of Quantization Error
• In the statistical approach, we assume that the quantization error is random
in nature.

• To carry out the analysis, we make the following assumptions about the
statistical properties of quantization error.

1. The error eq(n) is uniformly distributed over the range [−∆/2, ∆/2].
2. The error sequence {eq (n)} is a stationary white noise sequence.
Encoder
• In the Encoding process, each discrete value xq(n) is represented by a b
bits binary sequence.

• For a given L then b is calculated as:

• Each quantization level is assigned with a

b bits binary sequence.
This implies that the SQNR increases approximately 6 dB for every bit added
to the word length that is, for each doubling of the quantization levels.
Ex: A TV signal having a bandwidth 4.2MHz is transmitted using binary PCM
system. Given, the number of quantization level is 512. Determine (i)
Sampling Rate (ii) Code word length (iii) Final bit rate (iv) Transmission
bandwidth (v) Signal to Quantization Noise ratio (SNqR).
Ans:
i. Sampling Rate (fs) = 2 X fm = 2X 4.2 = 8.4 MHz
ii. L = 512 = 2^ 9, So code word length (v) = 9, i.e. each sample is
represented with 9 bits
iii. Final Bit Rate (R) = No of bits per sample X No of samples per
second = v X fs = 9 X 8.4 =75.6 Mbps
iv. Transmission Bandwidth = 75.6 MHz
v. SNqR = 6v = 6 X 9 = 54dB
 Pulse Code Modulation (PCM)
• Pulse-code modulation (PCM) is a method used to digitally represent
the analog signals.

• It is the standard form of digital audio in computers, compact discs, digital

telephony and other digital audio applications.

• In a PCM stream,
i. the amplitude of the analog signal is sampled regularly at uniform intervals,

ii. each sample is quantized to the nearest value quantized level,

iii. And, each quantized samples are encoded with a bit stream.
PCM communication system
Case study: Telephony System
 Maximum frequency of baseband (Voice) signal fm= 4kHz

 Nyquist rate of sampling fs = 2Xfm = 8k samples/ sec. i.e. 8000 samples generated per second

 8 bit quantizer is used, i.e. 256 quantization levels

 Each samples is represented with 8 bits

 Data rate (Rb) is the number of bits generated per second i.e. 8000 X 8 =64000 =64kbps

 Duration of 1 bit = 1/64000 = 15.625µs

 Bandwidth of the signal = 64kHz

Multiplexing
• It is the process of sharing a single communication channel (or medium) for transmission of
multiple signals.
1. A multiplexer (MUX) combines multiple signal into a single data stream.
2. A de-multiplexer (DeMux) separate a single data stream into multiple signals.

It is of two types:
1. Frequency Division Multiplexing
(FDM)
2. Time Division Multiplexing (TDM)
 Total bandwidth available in a
communication medium is
divided into a series of non-
overlapping frequency bands,
each of which is used to carry a
separate signal.

Frequency Division Multiplexing

• When Time Division Multiplexing (TDM) technique is used, multiple signals share the same
channel by taking turn transmitting.
• Data is broken up into frames and assigned to time slots.
• This technique is primarily used for digital data.
• Each signal uses the entire bandwidth of the channel when transmitting.
• On the receiving end, the demultiplexing process requires synchronization of the frames.
This is often accomplished through a sync pulse.

Time Division Multiplexing (TDM)

 It allows 24 simultaneous calls over single
T1 carrier system channel using TDM
• 1 frame: Represents one sample per each
call.
= 24 samples = 24 X 8 bits = 192 bits
• 1 synchronization bit per frame
representing start of the frame.
• Total bits / frame = 192 + 1 = 193
Number of frames per second = 8000

Number of bits per second = Data Rate of T1 carrier system

= 8000 X 193 = 1544000 = 1.544Mbps
Digital Hierarchy
Non-uniform Quantization
• It is a process of quantization with non-uniform step size.

• Need of non-uniform quantizer:

In uniform quantizer, the step size is uniform through out the dynamic range of the signal.

Hence penalty due to quantization is more for samples with low amplitude and less for
samples with high amplitude

It results non-uniform signal to quantization noise ratio (SNqR)

The non-uniform quantizer is implemented using Companding technique to ensure

unform SNqR for all range of sample values.
 Companding
• In non-uniform quantization, the step size varies according to the signal level.

• If the signal level is low then step size will be small. So, the step size will be low for
weak signal.

• Thus the quantization noise will also be low.

• Companding is a technique of achieving non-uniform quantization.

• It is a word formed by the combination of words compression and expanding.

• Companding is done in order to improve SNqR of weak signals.

 Companding Model

• Companding model consists of

1. Compressor (Tx side)
2. Uniform quantizer (Tx side)
3. Expander (Rx side)
• At the transmitting end the signal is compressed and uniform quantized
• At the receiving end the compressed signal is expanded in order to have
the original signal.
Compressor
At the transmitting end, the signal is first
provided to the compressor.

The compressor provides

1. high gain to low input signal and
2. low gain to high input signal.

The output of compressor is provided to

uniform quantizer resulting non-uniform
quantization.
 Expander
• It performs reverse operation of
compandor

The expander provides

1. low gain to low input signal and
2. high gain to high input signal.

• This is done in order to achieve the

originally transmitted signal at the output.
Companding standards
used in industry:

1. A-Law Companding
2. µ-Law Companding
 A-Law Companding
European countries practice A-Law Companding
Compression Equation

Expansion Equation

A = 87.6 is the compression parameter defined by Consultative Committee for

International Telephony And Telegraphy (CCITT) G.711
µ-Law Companding
North America and Japan practices µ-law Companding
technique .
Compression Equation

Expansion Equation

µ = 255 is the compression parameter defined by Consultative Committee for

International Telephony And Telegraphy (CCITT) G.711