Introduction

Pandit Deendayal Petroleum University School of Technology

20ECE301T Digital Communication 20ECE301T

Teaching Scheme Examination Scheme Teaching

Scheme

L T P C Hrs/Week Theory Practical Total L

Marks
MS ES IA LW LE/Viva

3 0 0 3 3 25 50 25 -- -- 100 3
Week Topic Taught Remarks

1-4 Digital Baseband Communication Techniques Unit 1

5-8 Digital Carrier Communication Systems Unit 2

9-12 Signal Shaping, Transmission and Optimum Reception Unit 3

13-16 Advanced Concepts and Way Forward Unit 4

1. B.P. Lathi, Zhi Ding “Modern Digital and Analog Communication Systems”,
Oxford University Press.
2. P. Chakrabarti,“Analog and Digital Communication”, Dhanpat Rai & Co.
3. Wayne Tomasi “Electronic Communications Systems”, Pearson education
India.
4. Taub and Schilling, “Principles of Communication Systems - Taub &
Schilling”, TMH.
5. Amitabh Bhattacharya,“Digital Communication”, TMH.
6. S. Haykin,“Digital Communication”, John Wiley.
7. John G. Proakis,“Digital Communications”, McGraw Hill Education.
8. Simon Haykins & Michael Moher, Communication Systems, 5th Edition, John
Willey, India Pvt. Ltd, 2010.
 To impart basic knowledge of digital modulation and
demodulation techniques.
 To Understand coding and multiple access principles.
 Understand basic principles of spread spectrum
communication system.
• CO1 - Remember principles of various digital communication methods.
• CO2 - Understand digital modulation, detection, coding, digital transmission and
access techniques.
• CO3 - Apply mathematical concepts to model digital communication system.
• CO4 - Analyze digital communication systems to obtain various parameters like
bandwidth, data rate etc.
• CO5 - Evaluate and compare performance of digital communication systems.
• CO6 - Design digital communication system at block diagram level.
 This is reserved time slot for students by faculty

 Faculty will make sure to remain in cabin during this time slot for the
session
 Monday 04.00 pm -05.00 pm

 Purpose

 Doubt clearing Email:

 One to one interactions [email protected]
Contact No.: 07923275459
Cabin Location
Room No. 208 – Faculty Wing, 2nd Floor, E Block, School of
Technology
• A signal which is a continuous function of time and used to carry the
information is known as an analog signal.
• An analog signal represents a quantity analogous to another quantity,
for example, in case of an analog audio signal, the instantaneous value
of signal voltage represents the pressure of the sound wave.
• Analog signals utilize the properties of medium to convey the
information. All the natural signals are the examples of analog signals.
• However, the analog signals are more susceptible to the electronic
noise and distortion which can degrade the quality of the signal.
• A signal that is discrete function of time, i.e. which is not a
continuous signal, is known as a digital signal.
• The digital signals are represented in the binary form and consist
of different values of voltage at discrete instants of time.
• Basically, a digital signal represents the data and information as a
sequence of separate values at any given time.
• The digital signal can only take on one of a finite number of
values.
Advantages of Digital Communication:
• Ruggedness to channel noise and other interference.
• Flexible implementation of digital hardware systems.
• Conversion of information from a variety of sources into a common format.
• Coding of digital signal to yield extremely low error rate and high fidelity.
• Security of information through encryption.
Disadvantages:
• Increased Transmission Bandwidth.
• Increased system complexity
The Information Capacity (bits per sec): The maximum rate at
which the information can be transmitted across the channel
without error.
The information bearing
 SAMPLE AND HOLD (S/H)

 Reconstruction:
 SAMPLE AND HOLD (S/H)
 By using flat-top samples to generate a PAM signal we have introduced amplitude
distortion as well as a delay of τ/2. This is referred to as the aperture effect.
 However, Sampling theorem still holds.
 Reconstruction can be done by passing signal through ideal LPF and then through
equalizer filter which is inverse of filter used at sampling to remove the aperture
effect distortion.
 For a duty cycle τ/Ts ≤ 0.1, the amplitude distortion is less than 0.5%. In this case,
equalization may not be necessary in practical applications.
 Sample and Hold (S/H) is practically used in PCM: Input signal is continuously
sampled and each sample value is held until next processing unit (quantizer in
PCM) needs it’s value.
 It also helps in reduction of distortion at reconstruction end.
 Speech
- Telephone quality speech has a bandwidth of 4 kHz
(actually 300 to 3300Hz)
- Most digital telephone systems are sampled at 8000
samples/sec
 Audio:
- The highest frequency the human ear can hear is
approximately 20 kHz
- CD quality audio are sampled at rate of 44,000
samples/sec
 Video
- The human eye requires samples at a rate of at least 20
frames/sec to achieve smooth motion
  PWM:
 Generation: Generated by
comparing analog signal with
high frequency saw-tooth
signal.
 Detection: Spectra of PWM
signal contains baseband
spectra; easily demodulated
by using LPF.
 Applications: Digital Speed
control of DC Motor
• PPM:
• Generation: Passing PWM signal to Mono stable Multivibrator to trigger it to
generate a fixed duration short pulse on –ve edge of input pulse.
• Detection: Spectra of PPM signal does not contain baseband spectra. Needs
synchronization between transmitter and receiver. Detected by using product
detector (phase detector).
• Applications: Telemetry and position control systems
 SQNR Improvement:
 Overall SQNR of DPCM scheme = Pg/Pq = (Pg/Pd)(Pd/Pq) = Gp (SQNR)d;
where Gp= Pg/Pd (>1) and (SQNR)d is the SQNR for difference(error) signal
d(n). With ordinary PCM the achievable SQNR is (SQNR)d only as it is the
signal which is quantized. Here Gp is the processing gain (SQNR improvement
factor) due to differential quantization scheme in DPCM.
 Accordingly, the objective in implementing the DPCM should be to design the
prediction filter so as to minimize the prediction-error power.
 To achieve same SQNR, the DPCM needs less number of quantization levels
(less no of bits/sample) compared to PCM.
 For voice signals: 3 to 4 bits/sample in DPCM gives same SQNR performance
as 8 bits/sample in PCM.
 For TV signals: DPCM offers a saving of 18Mbps over PCM.
 The step size Δ of the quantizer in DPCM is
made variable here: updated automatically
depending on whether prediction error signal
d(n) is large or small.
 The prediction filter can also be made
adaptive by making the co-efficients of
prediction filter time varying at every sample
time using some adaptive algorithms (for e.g.
LMS – Least Mean Square).
 This will reduce the prediction error and
allows to match with varying power and
spectrum of input signal.
 Receiver uses the same adaptive algorithm
strategy.
 ADM Transmitter
 Practical Solution to
problems in DM: Use time
varying step size ∆:
Adaptive Delta Modulation
(ADM).
 Use Adaptive Algorithm in
feedback path: Logic to
control step-size ∆
according to input signal
derivative.
ADM Receiver
 Practically used Algo.:
Song Algorithm and Space-
Shuttle Algorithm.
 Same Algo. at Tx and Rx.
 DM: Difference between successive samples is quantized into 2 levels (1bit) : Quantizing
derivative of input signal and transmit derivative of a signal.
 So, the receiver of DM needs Integrator (Accumulator).

 The integrator or accumulator is undesired in receiver as it will accumulate the channel

noise received.
 To avoid integrator at receiver we must place it at suitable place in transmitter as shown in
figure so that overall system remains same as delta modulation. This modified scheme is
called sigma – delta modulation (SDM).
 The receiver is simply a low pass filter, as accumulator is already at transmitter.

 Two accumulators/integrators can be replaced by single accumulator/integrator as

shown in figure.
 SDM Simplifies the receiver design and improves noise performance.

 Also, it allows quantization of DC signal and reduces granular noise.

 Also, as signal is integrated (low pass filtered or smoothed) before quantization, slope
overloading becomes less likely.
The L-quantization level quantizer generates digital signal which is an L-ary digital signal.
It can be converted to binary digital signal by using binary encoding circuit (for e.g. successive approximation ADC).
Select the number of levels L = integer power of 2 (so that binary encoding can be done easily) = 2n
n=Number of bits used to represent a quantization level using binary code.
After binary encoding each bit (0 or 1) can be assigned a pulse (waveform)  Line coding. (Illustrated in figure for L =
16).
SQNR with binary encoding = 3*22nPg /Am2 =3*4nPg /Am2
SQNR in dB = 10 log10(3)+10 log10(Pg /Am2)+10 log10(4)n = 4.77+10log10(Pg /Am2) + 6.02n
For any signal g(t) it can be generalized as SQNR in dB ≈ α + 6n; where α is constant depending on nature of g(t).
Additionof 1 bit in quantization improves SQNR by 6 dB: But it will also Increases Transmission rate, BW requirement
and memory requirement.
n= Number of bits used in binary encoding; now specifies resolution of PCM or ADC.
In some applications like reading data from sensors 8-bit resolution is sufficient; but in audio-video applications we need
10,12 or 16 bits of resolution.
30
 Special Case: Let g(t) = Am cos(ωmt): A single tone sinusoid
 Pg = Am2/2
 SQNR = (3/2)(4)n
 SQNR in dB = 10 log10(3/2) + 10 log10(4)n
 SQNR in dB = 1.8 + 6n
 W = BW of baseband (analog) signals

 Fs = Sampling frequency (≥2W): samples/sec.

 n = No. of bits used to encode a quantized sample: bits/sample

 Bit rate (Rb) = nFs bits per second (bps)

 Sampling Theory: We can transmit a maximum of two pieces of information per second (2 bps)
per unit channel BW (1Hz) assuming noiseless channel.
 Transmission BW required to transmit PCM signal: BT = nFs/2 Hz

 BTmin = nW Hz (Fsmin=2W)

 e.g. Telephone Speech Signal:W = 3.4KHz, Fs =8KHz, n = 8 bits, bit rate = 64 kbps, BT =32KHz.

 Recall: SQNR = (3Pg/ Am2)(4)n

 SQNR = (3Pg/ Am2) (4)2BT/Fs

 SQNR increases exponentially with increase in transmission bandwidth: PCM allows exchange of
SQNR with required transmission bandwidth BT
 Recall: SQNR = α + 6n dB

 Increasing n by 1; improves SQNR by 6dB at the cost of increase in BT by Fs/2 (min. W): A
Good Deal!!!!!
 A raised cosine pulse with excess bandwidth 𝑓𝑥 = 0.8 MHz is used to
transmit binary data with bit duration 𝑇𝑏 = 0.5 𝜇𝑠.
a) Determine the rate at which the binary data can be transmitted
by this pulse via the Nyquist criterion?
b) What is the roll-off factor?

1
Data Rate, 𝑅𝑏 = = 2 𝑀𝑏𝑝𝑠
𝑇𝑏

𝑓𝑥 0.8×106
Roll-off factor, 𝑟 = = = 0.8
0.5𝑅𝑏 0.5×2×106

35
 Minimum Shift Keying (MSK): BW conserving CPFSK scheme: A very special
case of CPFSK.
 It is Digital NBFM (Narrow band FM  modulation index less than 1).
 With this properties the transmission BW is reduced to 1.5Rb.
 The binary pulses are rectangular in MSK.
 If the rectangular pulses are passed through a special filter called Gaussian
filter and then passed through MSK modulator GMSK.
 GMSK = Gaussian filter + MSK
• The spectral efficiency of MSK is further enhanced by GMSK.
 Gaussian filter has unit impulse response and frequency response given by

 Gaussian function has same nature in time domain and frequency domain
 α is a parameter of Gaussian function which controls the BW of a filter (B).
 GMSK
 Designed based on the product of
the filter bandwidth (B) and the
symbol period (Ts). Ts =Tb when
number of symbols in modulated
signal = 2.
 BTs = ∞ corresponds to MSK.
 GSM (Global System for Mobile)
uses BTs = 0.3, which defines the
bandwidth of the Gaussian filter.
 The smaller the value of BTs,
however, the higher the error
rates.
 Sacrifices the irreducible error
rate in exchange for extremely
good spectral efficiency and
constant envelop properties.
 QPSK Demodulator:

 Analysis: output of upper LPF = I(t)/2

 output of lower LPF = Q(t)/2
 Increase in M will improve BW requirement further but detection becomes more difficult and error
in detection increases as the successive phase differences become narrower.
MODULATION BITS PER SYMBOL MIN. BW
BPSK 1 2Rb
QPSK 2 Rb
8PSK 3 2Rb/3
8QAM 3 2Rb/3
16QAM 4 Rb/2
32QAM 5 2Rb/5
64QAM 6 Rb/3