Ec 6501-DC QB

EC 6501 - DIGITAL COMMUNCATION V SEM/III YEAR/DEPT.
OF ECE/PPGIT
EC6501 DIGITAL COMMUNICATION LTPC

3003
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
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
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
TEXT BOOK:
1. S. Haykin, “Digital Communications”, John Wiley, 2005
REFERENCES:
1. B. Sklar, “Digital Communication Fundamentals and Applications”, 2nd Edition, Pearson
Education, 2009
2. B.P.Lathi, “Modern Digital and Analog Communication Systems” 3rd Edition, Oxford
University Press 2007.
3. H P Hsu, Schaum Outline Series - “Analog and Digital Communications”, TMH 2006
4. J.G Proakis, “Digital Communication”, 4th Edition, Tata Mc Graw Hill Company, 2001.
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EC 6501 - DIGITAL COMMUNCATION V SEM/III YEAR/DEPT. OF ECE/PPGIT
UNIT I
SAMPLING AND QUANTIZATION
PART A
1. State sampling theorem. [AUC APR/MAY 2011] [AUC APR/MAY 2012]

If a finite –energy signal g(t) contains no frequencies higher than W hertz,it is completely
determined by specifying its co=ordinates at a sequence of points spaced 1/2W seconds apart.·
If a finite energy signal g(t) contains no frequencies higher than W hertz, it may be completely
recovered from its co=ordinates at a sequence of points spaced 1/2W seconds apart.
2. What is aliasing?[AUC APR 2011]

The phenomenon of a high-frequency in the spectrum of the original signal g(t)
seemingly taking on the identity of a lower frequency in the spectrum of the sampled signal g(t)
is called aliasing or fold over.
3. What is meant by PCM?

Pulse code modulation (PCM) is a method of signal coding in which the message signal
is sampled, the amplitude of each sample is rounded off to the nearest one of a finite set of
discrete levels and encoded so that both time and amplitude are represented in discrete form..
4. Define quantizing process.

The conversion of analog sample of the signal into digital form is called quantizing
process.
5. What are the two fold effects of quantizing process.[AUC APR 11]
1. The peak-to-peak range of input sample values subdivided into a finite set of decision levels
or decision thresholds
2. The output is assigned a discrete value selected from a finite set of representation levels are
reconstruction values that are aligned with the treads of the staircase.
6. What is meant by idle channel noise?

Idle channel noise is the coding noise measured at the receiver output with zero transmitter
input.
7. What is meant by prediction error?

The difference between the actual sample of the process at the time of interest and the predictor
output is called a prediction error.
8. Define delta modulation.

Delta modulation is the one-bit version of differential pulse code modulation.
9. Define adaptive delta modulation.

The performance of a delta modulator can be improved significantly by making the step size of
the modulator assume a time- varying form. In particular, during a steep segment of the input
signal the step size is increased.
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10. Name the types of uniform quantizer?

1. Mid tread type quantizer.
2. Mid riser type quantizer.
11. Define quantization error?[AUC APR 14]

Quantization error is the difference between the output and input values of quantizer. Because
of quantization inherent error are introduces in the signal. The error is called Quantization error
Є=xq(nTs)-x(nTs)
xq(nTs)- quantis ed value of the signal
x(nTs)- value of te sample before quantization
12. What you mean by non-uniform quantization? [AU-Apr‟2015]

Step size is not uniform. Non-uniform quantizer is characterized by a step size that
increases as the separation from the origin of the transfer characteristics is increased. Non-
uniform quantization is otherwise called as robust quantization
PART B
1.Draw a neat block diagram of a typical digital communication system and explain the
function of the key signal processing blocks.( 16) [NOV 2010]
• Basic Concepts in Signals
– Sampling
– Quantization
– Coding
• Communication is transferring data reliably from one point to another
– Data could be: voice, video, codes etc…
• It is important to receive the same information that was sent from the transmitter.
• Communication system
– A system that allows transfer of information realiably
• Information Source
– The source of data
• Data could be: human voice, data storage device CD, video etc..
– Data types:
• Discrete: Finite set of outcomes “Digital”
• Continuous : Infinite set of outcomes “Analog”
• Transmitter
– Converts the source data into a suitable form for transmission through signal
processing
Data form depends on the channel
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• Channel:
– The physical medium used to send the signal
– The medium where the signal propagates till arriving to the receiver
– Physical Mediums (Channels):
• Wired : twisted pairs, coaxial cable, fiber optics
• Wireless: Air, vacuum and water
– Each physical channel has a certain limited range of frequencies ,( fmin à fmax ),
that is called the channel bandwidth
– Physical channels have another important limitation which is the NOISE
• Channel:
• Noise is undesired random signal that corrupts the original signal and
degrades it
• Noise sources:
• Electronic equipments in the communication system
• Thermal noise
• Atmospheric electromagnetic noise (Interference with
another signals that are being transmitted at the same
channel)
– Another Limitation of noise is the attenuation
• Weakens the signal strength as it travels over the transmission medium
• Attenuation increases as frequency increases
– One Last important limitation is the delay distortion
• Mainly in the wired transmission
• Delays the transmitted signals à Violates the reliability of the
communication system
• Receiver
– Extracting the message/code in the received signal
• Example
• Speech signal at transmitter is converted into electromagnetic
waves to travel over the channel
• Once the electromagnetic waves are received properly, the
receiver converts it back to a speech form
– Information Sink
• The final stage
• The user
• Information source
– Analog Data: Microphone, speech signal, image, video etc…
– Discrete (Digital) Data: keyboard, binary numbers, hex numbers, etc…
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• Analog to Digital Converter (A/D)

– Sampling:
• Converting continuous time signal to a digital signal
– Quantization:
• Converting the amplitude of the analog signal to a digital value
– Coding:
• Assigning a binary code to each finite amplitude in the analog signal
• Source encoder
– Represent the transmitted data more efficiently and remove redundant
information
• Speech signals frequency and human ear “20 kHz”
– Two types of encoding:
– Lossless data compression (encoding)
• Data can be recovered without any missing information
– Lossy data compression (encoding)
• Smaller size of data
• Data removed in encoding can not be recovered again
• Channel encoder:
– To control the noise and to detect and correct the errors that can occur in the
transmitted data due the noise.
• Modulator:
– Represent the data in a form to make it compatible with the channel
• Carrier signal “high frequency signal”
• Demodulator:
– Removes the carrier signal and reverse the process of the Modulator
• Channel decoder:
– Detects and corrects the errors in the signal gained from the channel
• Source decoder:
– Decompresses the data into it’s original format.
• Digital to Analog Converter:
– Reverses the operation of the A/D
– Needs techniques and knowledge about sampling, quantization, and coding
methods.
• Information Sink
– The User
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2. Derive Geometrical representation of signal.

GEOMETRIC REPRESENTATION OF SIGNALS
• Objective: To represent any set of M energy signals {si(t)} as linear combinations of N
orthogonal basis functions, where N ≤ M
• Real value energy signals s1(t), s2(t),..sM(t), each of duration T sec
• The set of coefficients can be viewed as a N-dimensional vector, denoted by si

• Bears a one-to-one relationship with the transmitted signal si(t)
FIGURE : synthesizer for generating the signal si(t). (b) analyzer for generating the set of signal
vectors {si}.
• The signal vector si concept can be extended to 2D, 3D etc. N-dimensional Euclidian
space
• Provides mathematical basis for the geometric representation of energy signals that is
used in noise analysis
• Allows definition of
– Length of vectors (absolute value)
– Angles between vectors
– Squared value (inner product of si with itself)
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FIGURE : Illustrating the geometric representation of signals for the case when N = 2 and M =
3. (two dimensional space, three signals)
3.Explain the mathematical models of communication channel
PERFORMANCE METRICS
Transmitters modulate analog messages or bits in case of a DCS for transmission over a
channel.
Receivers recreate signals or bits from received signal (mitigate channel effects)
Performance metric for analog systems is fidelity, for digital it is the bit rate and error
probability.
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4. Distinguish between base band and bandpass signaling.

BASE BAND SIGNALLING
• Digital transmission is the transmission of electrical pulses. Digital information is binary
in nature in that it has only two possible states 1 or 0. Sequences of bits encode data
(e.g., text characters).
• Digital signals are commonly referred to as baseband signals.
• In order to successfully send and receive a message, both the sender and receiver have
to agree how often the sender can transmit data (data rate).
• Data rate often called bandwidth
BASE BAND TRANSMISSION
• With unipolar signaling techniques, the voltage is always positive or negative (like a dc
current).
• In bipolar signaling, the 1’s and 0’s vary from a plus voltage to a minus voltage (like an
ac current).
• In general, bipolar signaling experiences fewer errors than unipolar signaling because
the signals are more distinct.
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Manchester encoding is a special type of unipolar signaling in which the signal is changed from
a high to low (0) or low to high (1) in the middle of the signal.
• More reliable detection of transition rather than level
– consider perhaps some constant amount of dc noise, transitions still detectable
but dc component could throw off NRZ-L scheme
– Transitions still detectable even if polarity reversed
Manchester encoding is commonly used in local area networks (ethernet, token ring).
Spectrum: frequencies that make up a signal
Bandwidth: range of frequencies passed by the channel with a small amount of
attenuation
Filtering: controlling the channel bandwidth to prevent interference from other signals
BANDPASS SIGNALLING
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5. Classify channels. Explain the mathematical model of any two communication

channels (16)[MAY 2011]
DIGITAL CHANNEL MODELS
• In a digital channel model, the transmitted message is modelled as a digital signal at
a certain protocol layer.
• Underlying protocol layers, such as the physical layer transmission technique, is
replaced by a simplified model.
• The model may reflect channel performance measures such as bit rate, bit errors,
latency/delay, delay jitter, etc. Examples of digital channel models are:
• Binary symmetric channel (BSC), a discrete memoryless channel with a certain bit error
probability
• Binary bursty bit error channel model, a channel "with memory"
• Binary erasure channel (BEC), a discrete channel with a certain bit error detection
(erasure) probability
• Packet erasure channel, where packets are lost with a certain packet loss probability or
packet error rate
• Arbitrarily varying channel (AVC), where the behavior and state of the channel can
change randomly
ANALOG CHANNEL MODELS
• In an analog channel model, the transmitted message is modelled as an analog
signal.
• The model can be a linear or non-linear, time-continuous or time-discrete (sampled),
memoryless or dynamic (resulting in burst errors), time-invariant or time-variant (also
resulting in burst errors), baseband, passband (RF signal model), real-valued or
complex-valued signal model.
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• The model may reflect the following channel impairments:

• Noise model, for example
o Additive white Gaussian noise (AWGN) channel, a linear continuous memoryless
model
o Phase noise model
• Interference model, for example cross-talk (co-channel interference) and intersymbol
interference (ISI)
• Distortion model, for example a non-linear channel model causing intermodulation
distortion (IMD)
• Frequency response model, including attenuation and phase-shift
• Group delay model
• Modelling of underlying physical layer transmission techniques, for example a complex-
valued equivalent baseband model of modulation and frequency response
• Radio frequency propagation model, for example
o Log-distance path loss model
o Fading model, for example Rayleigh fading, Ricean fading, log-normal shadow
fading and frequency selective (dispersive) fading
o Doppler shift model, which combined with fading results in a time-variant system
Types of communications channels
• Digital (discrete) or analog (continuous) channel
• Baseband and passband channel
• Transmission medium, for example a fibre channel
• Multiplexed channel
• Computer network virtual channel
• Simplex communication, duplex communication or half duplex communication channel
• Return channel
• Uplink or downlink (upstream or downstream channel)
• Broadcast channel, unicast channel or multicast channel
CHANNEL PERFORMANCE MEASURES
These are examples of commonly used channel capacity and performance measures:
• Spectral bandwidth in Hertz
• Symbol rate in baud, pulses/s or symbols/s
• Digital bandwidth bit/s measures: gross bit rate (signalling rate), net bit rate (information
rate), channel capacity, and maximum throughput
• Channel utilization
• Link spectral efficiency
• Signal-to-noise ratio measures: signal-to-interference ratio, Eb/No, carrier-to-interference
ratio in decibel
• Bit-error rate (BER), packet-error rate (PER)
• Latency in seconds: propagation time, transmission time
• Delay jitter
6. Explain how PWM and PPM signals are generated.(16 ) [MAY 2011]
PULSE WIDTH MODULATION
Output signal alternates between on and off within specified period
Controls power received by a device
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The voltage seen by the load is directly proportional to the source voltage
TYPES OF PULSE WIDTH
Pulse center fixed, edges modulated
Leading edge fixed, tailing edge modulated
Tailing edge fixed, leading edge modulated
Pulse Width constant, period modulated
DIGITAL METHODS OF GENERATING PWM

Digital: Counter used to handle transition
Delta : used to find the PWM at a certain limit
Delta Sigma: used to find the PWM but has advantage of reducing optimization noise
ANALOG GENERATION OF PWM

PULSE POSITION MODULATION
Also sometimes known as pulse-phase modulation
The amplitude and width of the pulse is kept constant in the system
The position of each pulse, in relation to the position of a recurrent reference pulse, is
varied by each instantaneous sampled value of the modulating wave
Often used in optical communication, such as fiber optics, in which there is little
multipath way interference
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Used exclusively for transferring digital signals and cannot be used with analog systems
Used for transferring simple data and is not effective at transferring files
ADVANTAGES OF PULSE MODULATION
It has the advantage over pulse amplitude modulation (PAM) in that it has a higher noise
immunity
It requires constant transmitter power since the pulses are of constant amplitude and
duration
DISADVANTAGES OF PULSE POSITION MODULATION
Signal and noise separation is very easy
Depending on transmitter-receiver synchronization
Highly sensitive to multipath way interference
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UNIT II
WAVEFORM CODING
PART A
1.What is the disadvantage of uniform quantization over the non-uniform quantization?

SNR decreases with decrease in input power level at the uniform quantizer but non-
uniform quantization maintains a constant SNR for wide range of input power levels. This type
of quantization is called as robust quantization.
2. What do you mean by companding? Define compander.[AUC ND 12]

The signal is compressed at the transmitter and expanded at the receiver. This is called
as companding. The combination of a compressor and expander is called a compander.
3. What is PAM?
PAM is the pulse amplitude modulation. In pulse amplitude modulation, the amplitude of
a carrier consisting of a periodic train of rectangular pulses is varied in proportion to sample
values of a message signal.
4. What is the need for speech coding at low bit rates?
The use of PCM at the standard rate of 64 Kbps demands a high channel bandwidth for
its transmission ,so for certain applications, bandwidth is at premium, in which case there is a
definite need for speech coding at low bit rates, while maintaining acceptable fidelity or quality of
reproduction.
5. Define ADPCM.
It means adaptive differential pulse code modulation, a combination of adaptive
quantization and adaptive prediction. Adaptive quantization refers to a quantizer that operates
with a time varying step size. The autocorrelation function and power spectral density of speech
signals are time varying functions of the respective variables.
6. What is meant by forward and backward estimation?[AUC ND 15]

AQF: Adaptive quantization with forward estimation. Unquantized samples of the input
signal are used to derive the forward estimates.
AQB: Adaptive quantization with backward estimation. Samples of the quantizer output are
used to derive the backward estimates.
APF: Adaptive prediction with forward estimation, in which unquantized samples of the input
signal are used to derive the forward estimates of the predictor coefficients.
APB: Adaptive prediction with backward estimation, in which Samples of the quantizer
output and the prediction error are used to derive estimates of the predictor coefficients.
7. What are the limitations of forward estimation with backward estimation?

o Side information
o Buffering
o Delay
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8. How are the predictor coefficients determined?

For the adaptation of the predictor coefficients the least mean square (LMS) algorithm is
used.
9. Define adaptive subband coding?[AUC APR 15]

It is a frequency domain coder, in which the speech signal is divided in to number of
subbands and each one is coded separately. It uses non masking phenomenon in perception
for a better speech quality.
10. What are formant frequencies?

In the context of speech production the formant frequencies are the resonant
frequencies of the vocal tract tube. The formants depend on the shape and dimensions of the
vocal tract.
11. What is the bit rate in ASBC?

Nfs= (MN) (fs/M)
Nfs->bit rate
M->number of subbands of equal bandwidths
N->average number of bits
fs/M->sampling rate for each subband
12. Define Adaptive filter?

It is a nonlinear estimator that provides an estimate of some desired response without
requiring knowledge of correlation functions, where the filter coefficients are data dependent.
PART-B
1. With neat diagram and necessary equations, explain the concept of sampling and
reconstruction of signals (MAY/JUNE 2009)
Analog signal is sampled every TS secs.
Ts is referred to as the sampling interval.
fs = 1/Ts is called the sampling rate or sampling frequency.
There are 3 sampling methods:
Ideal - an impulse at each sampling instant
Natural - a pulse of short width with varying amplitude
Flattop - sample and hold, like natural but with single amplitude value
The process is referred to as pulse amplitude modulation PAM and the outcome is a
signal with analog (non integer) values
According to the Nyquist theorem, the sampling rate must be at least 2 times the highest
frequency contained in the signal.
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Sampling: obtain samples of x(t) at uniformly spaced time intervals

Quantization: map each sample into an approximation value of finite precisi
precision
Pulse Code Modulation: telephone speech
CD audio
Compression: to lower bit rate further, apply additional compression method
Differential coding: cellular telephone speech
Subband coding: MP3 audio
A signal that varies faster needs to be sampled more frequently.
SAMPLING THEOREM
2. Explain the principle of Delta modulation and derive and expression for thermal noise
in delta modulation.
Next form of pulse modulation
Transmits information only to indicate whether the analog signal that is being encoded
goes up or goes down
The Encoder Outputs are highs or lows that “instruct” whether to go up or down,
respectively
DM takes advantage of the fact that voice signals do not change abruptly
DM sends only
ly the difference between pulses, if the pulse at time tn+1 is higher in
amplitude value than the pulse at time tn, then a single bit, say a “1”, is used to indicate
the positive value.
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If the pulse is lower in value, resulting in a negative value, a “0” is used.

This scheme works well for small changes in signal values between samples.
If changes in amplitude are in large large, this will errors.result
The analog signal is quantized by a one-bit ADC (a comparator implemented as a

comparator)
The comparator output is converted back to an analog signal with a 1-bit DAC, and
subtracted from the input after passing through an integrator
The shape of the analog signal is transmitted as follows: a "1" indicates that a positive
excursion has occurred since the last sample, and a "0" indicates that a negative
excursion has occurred since the last sample.
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3. Discuss the principle of Adaptive Delta modulation in detail [MAY /JUNE 2009]
• Adaptive delta modulation (ADM) or continuously variable slope delta modulation
(CVSD) is a modification of DM in which the step size is not fixed.
• Rather, when several consecutive bits have the same direction value, the encoder and
decoder assume that slope overload is occurring, and the step size becomes
progressively larger.
• Otherwise, the step size becomes gradually smaller over time.
• ADM reduces slope error,at the expense of increasing quantizing error.
• This error can be reduced by using a low pass fil ADM provides robust performance in
the presence of bit errors meaning error detection and correction are not typically used
in an ADM radio design, this allows for a reduction in host processor workload (allowing
a low-cost processor to be used).
• To minimize slope overload noise while holding the granular noise at a reasonable level.
• One way to improve is to use adaptive DM, where the step size is not required to be
constant.
• The step size of the DAC is automatically varied, depending on the amplitude
characteristics of the analog input signal.
• With ADPCM, after predetermined number of consecutives 1s or 0s, the step size is
automatically increased - reduce slope overload noise.
• When alternative sequence 0s and 1s is occurring, DAC will reduce the step size -
reduce granular noise.
• Adaptive delta modulation

– A better performance can be achieved if the value of δ is not fixed.
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– The value of δ changes according to the amplitude of the analog signal.

• Quantization Error
– DM is not perfect.
– Quantization error is always introduced in the process.
– Much less than that for PCM.
TRANSMISSION MODES
• The transmission of binary data across a link can be accomplished in either parallel or
serial mode.
• In parallel mode, multiple bits are sent with each clock tick. In serial mode, 1 bit is sent
with each clock tick.
• There is only one way to send parallel data, there are three subclasses of serial
transmission: asynchronous, synchronous, and isochronous.
4. Explain uniform and non uniform quantization

Sampling results in a series of pulses of varying amplitude values ranging between two
limits: a min and a max.
The amplitude values are infinite between the two limits.
We need to map the infinite amplitude values onto a finite set of known values.
This is achieved by dividing the distance between min and max into L zones, each of
height ∆.
∆ = (max - min)/L
in)/L
The midpoint of each zone is assigned a value from 0 to L
L-1 (resulting in L values)
Each sample falling in a zone is then approximated to the value of the midpoint.
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QUANTIZATION ERROR
When a signal is quantized, we introduce an error - the coded signal is an approximation
of the actual amplitude value.
The difference between actual and coded value (midpoint) is referred to as the
quantization error.
The more zones, the smaller ∆ which results in smaller errors.
BUT, the more zones the more bits required to encode the samples -> higher bit rate
The bit rate of a PCM signal can be calculated form the number of bits per sample x the
sampling rate
Bit rate = nb x fs
The bandwidth required to transmit this signal depends on the type of line encoding
used. Refer to previous section for discussion and formulas.
A digitized signal will always need more bandwidth than the original analog signal. Price
we pay for robustness and other features of digital transmission
• Digital representations of analog signals are in the form of bits. These bits are taken
from an analog-to-digital converter, processed and then put to a digital-to-analog
converter.
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• Negative values can also be represented digitally. There are two common formats: sign
magnitude and two’s complement.
• In sign magnitude format, the most significant bit is a sign bit:1 is negative, 0 is
positive.
• In two’s complement format, positive numbers are like normal positive numbers.
Negative numbers are wrapped backwards: -1 is 111, -2 is 110, etc.
5. Give the block diagram of differential pulse Code Modulation Scheme and explain the
principles in detail
• Sampling at higher then Nyquist rate creates correlation between samples (good and
bad)
• Difference between samples has small variance – smaller than the variance of the
signal itself
• Encoded signal contains redundant information
• Can be used to a positive end – remove redundancy before encoding to get a more
efficient signal to be transmitted
• Sampling at higher then Nyquist rate creates correlation between samples (good and
bad)
• Difference between samples has small variance – smaller than the variance of the
signal itself
• Encoded signal contains redundant information
• Can be used to a positive end – remove redundancy before encoding to get a more
efficient signal to be transmitted
To recover an analog signal from a digitized signal we follow the following steps:
We use a hold circuit that holds the amplitude value of a pulse till the next pulse
arrives.
We pass this signal through a low pass filter with a cutoff frequency that is equal
to the highest frequency in the pre-sampled signal.
The higher the value of L, the less distorted a signal is recovered.
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• Predicted value – achieved by linear prediction filter whose input is the quantized version
of the input sample m[n].
• The difference e[n] is the prediction error (what we expect and what actually happens)
• By encoding the quantizer output we actually create a variation of PCM called differential
PCM (DPCM).
• Decoder – constructs the quantized error signal
• Quantized version of the input is recovered by using the same prediction filter as at the
tx
• If there is no channel noise – encoded input to the decoder is identical to the transmitter
output
• Then the receiver output will be equal to mq[n] (differs from m[n] by q[n] caused by
quantizing the prediction error e[n])
6. Explain a spectral waveform encoding process

Encoding of Waveforms to Compress Information
Data, Speech, Image
Encoding of Speech Signals – Vocoders
Makes use of special properties of speech
Periodicity
Distinction between voiced and unvoiced sounds
Image Encoding
Makes use of suitable transforms
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Uses special techniques

Transmits only the difference between image frames
Combines speech and image coding for video
Observe Original Signal
Amplitude of a train of pulses is modulated: Pulse Amplitude ∝ Signal Amplitude
Width of a train of pulses is modulated: Pulse Width ∝ Signal Amplitude
Position of a train of pulses is modulated: Pulse Position ∝ Signal Amplitude
LINEAR PREDICTION CODING

Consider a finite-duration impulse response (FIR) discrete-time filter which consists of
three blocks :
1. Set of p ( p: prediction order) unit-delay elements (z-1)
2. Set of multipliers with coefficients w1,w2,…wp
3. Set of adders ( ∑ )
REDUCE THE SAMPLING RATE
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UNIT III
BASEBAND TRANSMISSION
PART-A
1. What is linear code?
A code is linear if the sum of any two code vectors produces another code vector.
2. What is code rate?[AUC 12]

Code rate is the ratio of message bits (k) and the encoder output bits (n). It is defined by
r (i.e) r= k/N
3. Define code efficiency.

It is the ratio of message bits in a block to the transmitted bits for that block by the
encoder i.e Message bits in a block. Transmitted bits for the block
4. What is hamming distance?[AUC APR 15]

The hamming distance between two code vectors is equal to the number of elements in
which they differ. For example let the two code vectors be X=(101) and Y= (110) . These two
code vectors differ in second and third bits.
5. What is meant by systematic & non-systematic code?

In a systematic block code, message bit appear first and then check bits. In the non-
systematic code, message and check bits cannot be identified in the code vector.
6. How syndrome is calculated in Hamming codes and cyclic codes ?
In Hamming codes the syndrome is calculated as , S = YH T. Here Y is the received and HT is
the transpose of parity check matrix. In cyclic code, the syndrome vector polynomial is given as,
S (P) = remainder ( y (p)/ G (P) )
Y(P) is received vector polynomial and G (p) is generator polynomial.
7.. What is BCH Code ?[AUC 11]

BCH codes are most extensive and powerful error correcting cyclic code. The decoding
of BCH coder is comparatively simpler. For any positive integer ‘m’ and ‘t’ , there exists a BCH
code with following parameters : Block length n = 2 m-1
No. of parity check bits : n-k mt . Minimum distance : dmin 2t +1
8. What are the conditions to satisfy the hamming code.

1) No. of Check bits q 3
2) Block length n = 2q –1
3) No of message bits K = n-q
4) Minimum distance dmin =3
9. Define code word & block length.

The encoded block of ‘n’ bits is called code word. The no. of bits ‘n’ after coding is called block
length.
10. Give the parameters of RS codes .

Reed Solomon codes. These are non binary BCH codes. Block length = n =2m -1 symbols
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Message size : k symbols . Parity check size : n-k= 2t symbols . Minimum distance , dmin =2t
+1 symbols.
11. Why RS codes are called maximum distance separable codes ?

( n,k) Linear block code for which the minimum distance equals n – k + 1 is
called maximum distance separable codes. For RS code minimum distance equals n – k + 1 so
it is called as maximum distance separable codes.
12. What are Golay codes ?[AUC 14]

Golay code is the (23, 12) cyclic code whose generating polynomial is, G(p)
=P11+P9+p7+P6+p5+p+1 . This code has a minimum distance of dmin=7. This code can
correct upto 3 errors. It is perfect code.
PART B
1. Explain linear block coding
Linearity:
where m is a k-bit information sequence
c is an n-bit codeword.
⊕ is a bit-by-bit mod-2 addition without carry

Linear code: The sum of any two codewords is a codeword.
Observation: The all-zero sequence is a codeword in every linear block code.
Def: The weight of a codeword ci , denoted by w(ci), is the number of of nonzero

elements in the codeword.
Def: The minimum weight of a code, wmin, is the smallest weight of the nonzero
codewords in the code.
Theorem: In any linear code, dmin = wmin
The code C is called a k-dimensional subspace. G is called a generator matrix of the code.
Here G is a k ×n matrix of rank k of elements from GF(2), gi is the i-th row vector of G. The rows
of G are linearly independent since G is assumed to have rank k.
LINEAR SYSTEMATIC BLOCK CODE:
An (n, k) linear systematic code is completely specified by a k × n generator matrix of the

following form.
where Ik is the k × k identity matrix.
The number of codeworde is 2k since there are 2k distinct messages.
The set of vectors {gi} are linearly independent since we must have a set of unique
codewords.
Linearly independent vectors mean that no vector gi can be expressed as a linear

combination of the other vectors.
25
HAMMING WEIGHT:
The minimum hamming distance of a linear block code is equal to the minimum hamming
weight of the nonzero code vectors. Since each gi єC ,we must have W h(gi) ≥ dmin this a
necessary condition but not sufficient.
GENERATOR MATRIX
All 2k codewords can be generated from a set of k linearly independent codewords.
The simplest choice of this set is the k codewords corresponding to the information
sequences that have a single nonzero element.
Illustration: The generating set for the (7,4) code:
1000 ===> 1101000
0100 ===> 0110100
0010 ===> 1110010
0001 ===> 1010001
GENERATOR MATRIX:
PARITY CHECK MATRIX:
HAMMING CODES:
Hamming codes constitute a class of single-error correcting codes defined as: n = 2r-1, k
= n-r, r > 2
The minimum distance of the code dmin = 3
Hamming codes are perfect codes.
DECODING:
26
2. Explain convolution coding
VITERBI DECODING APPLICATIONS:
• Decoding trellis-coded modulation in modems
• Most common FEC technique used in space communications (r = ½, K = 7)
• Usually implemented as serial concatenated block and convolutional coding – first Reed-
Solomon, then convolutional
• Turbo codes are a new parallel-concatenated convolutional coding technique
Output Symbols, if
Current Input = Input = 1:

State 0:
00 00 11
01 11 00
10 10 01
11 01 10
27
28
3. Explain the classification of line codes
• Binary data can be transmitted using a number of different types of pulses.
• The choice of a particular pair of pulses to represent the symbols 1 and 0 is

called Line Coding and the choice is generally made on the grounds of one or
more of the following considerations:
– Presence or absence of a DC level.
– Power Spectral Density- particularly its value at 0 Hz.
– Bandwidth.
– BER performance (this particular aspect is not covered in this lecture).
– Transparency (i.e. the property that any arbitrary symbol, or bit, pattern can be
transmitted and received).
– Ease of clock signal recovery for symbol synchronisation.
– Presence or absence of inherent error detection properties.
UNIPOLAR SIGNALLING
Unipolar signalling (also called on-off keying, OOK) is the type of line coding in which one binary
symbol (representing a 0 for example) is represented by the absence of a pulse (i.e. a SPACE)
and the other binary symbol (denoting a 1) is represented by the presence of a pulse (i.e. a
MARK).
There are two common variations of unipolar signalling: Non-Return to Zero (NRZ) and Return
to Zero (RZ).
UNIPOLAR NON-RETURN TO ZERO (NRZ):
In unipolar NRZ the duration of the MARK pulse (Ƭ ) is equal to the duration (To) of the symbol
slot.
Unipolar Non-Return to Zero (NRZ):
In unipolar NRZ the duration of the MARK pulse (Ƭ ) is equal to the duration (To) of the symbol
slot. (put figure here).
Advantages:
– Simplicity in implementation.
– Doesn’t require a lot of bandwidth for transmission.
Disadvantages:
– Presence of DC level (indicated by spectral line at 0 Hz).
– Contains low frequency components. Causes “Signal Droop” (explained later).
– Does not have any error correction capability.
– Does not posses any clocking component for ease of synchronisation.
Is not Transparent. Long string of zeros causes loss of synchronisation.
29
RETURN TO ZERO (NRZ):

UNIPOLAR NON-RETURN
• When Unipolar NRZ signals are transmitted over links with either transformer or
capacitor coupled (AC) repeaters, the DC level is removed converting them into a polar
format.
• The continuous part of the PSD is also non-zero

non zero at 0 Hz (i.e. contains low frequency
components). This means that AC coupling will result in distortion of the transmitted
pulse shapes.
• AC coupled transmission lines typically behave like high high-pass RC filters rs and the
distortion takes the form of an exponential decay of the signal amplitude after each
transition. This effect is referred to as “Signal Droop” and is illustrated in figure below.
RETURN TO ZERO (RZ):
In unipolar RZ the duration of the MARK pulse puls (Ƭ ) is less than the duration (To) of the symbol
slot. Typically RZ pulses fill only the first half of the time slot, returning to zero for the second
half.
POLAR SIGNALLING
In polar signalling a binary 1 is represented by a pulse g1(t) and a binary 0 by the opposite (or
antipodal) pulse g0(t) = -g1(t). Polar signalling also has NRZ and RZ forms.
30
4. Explain bipolar signaling in line coding
• Bipolar Signalling is also called “alternate mark inversion” (AMI) uses three voltage
levels (+V, 0, -V) to represent two binary symbols. Zeros, as in unipolar, are represented
by the absence of a pulse and ones (or marks) are represented by alternating voltage
levels of +V and –V.
• Alternating the mark level voltage ensures that the bipolar spectrum has a null at DC
And that signal droop on AC coupled lines is avoided.
• The alternating mark voltage also gives bipolar signalling a single error detection
capability.
• Like the Unipolar and Polar cases, Bipolar also has NRZ and RZ variations.
31
HDBN SIGNALING
• HDBn is an enhancement of Bipolar Signalling. It overcomes the transparency problem

encountered in Bipolar signalling.
• In HDBn systems when the number of continuous zeros exceeds n they are replaced by
a special code.
• The code recommended by the ITU-T for European PCM systems is HDB-3 (i.e. n=3).
• In HDB-3 a string of 4 consecutive zeros are replaced by either 000V or B00V.
Where, ‘B’ conforms to the Alternate Mark Inversion Rule. ‘V’ is a violation of the Alternate Mark
Inversion Rule
ADVANTAGES:
– No DC component.
– Occupies less bandwidth than unipolar and polar RZ schemes.
– Does not suffer from signal droop (suitable for transmission over AC coupled
lines).
– Possesses single error detection capability.
– Clock can be extracted by rectifying (a copy of) the received signal.
DISADVANTAGES:
– Is not Transparent.
5. Explain Manchester signalling in digital communication
MANCHESTER SIGNALLING:
– In Manchester encoding , the duration of the bit is divided into two halves. The voltage
– remains at one level during the first half and moves to the other level during the
– second half.
– A ‘One’ is +ve in 1st half and -ve in 2nd half.
– A ‘Zero’ is -ve in 1st half and +ve in 2nd half.

32
• The transition at the centre of every bit interval is used for synchronization at the
receiver.
• Manchester encoding is called self-synchronizing. Synchronization at the receiving end

can be achieved by locking on to the the transitions, which indicate the middle of the
bits.
• It is worth highlighting that the traditional synchronization technique used for unipolar,
polar and bipolar schemes, which employs a narrow BPF to extract the clock signal
cannot be used for synchronization in Manchester encoding.
• The PSD of Manchester encoding does not include a spectral line/ impulse at symbol
rate (1/To). Even rectification does not help.
Advantages:
– No DC component.
– Does not suffer from signal droop (suitable for transmission over AC coupled
lines).
– Easy to synchronise with.
– Is Transparent.
Disadvantages:
– Because of the greater number of transitions it occupies a significantly large

bandwidth.
– Does not have error detection capability.
33
UNIT IV
DIGITAL MODULATION SCHEME
PART-A
1 .Define data Signalling Rate.[AUC 11]

Data signalling rate is defined as the rate measured in terms bits per second(b/s) at which data
are transmitted. Data signaling rate Rb=I/Tb Where Tb=bit duration.
2. Define modulation rate.[AUC 13]

It is defined as the rate at which signal level is changed depending . On the nature of the
format used to represent the digital data.It is measured in Bauds or symbols per second.
3. State NRZ unipolar format

In this format binary 0 is represent by no pulse and binary 1 is Represented by the positive
pulse.
4. State NRZ polar format.

Binary 1 is represented by a positive pulse and binary 0 is represented by a Negative pulse.
5. State NRZ bipolar format.

Binary 0 is reporesented by no pulse and binary one is represented by the alternative
positive and negative pulse.
6. State Manchester format.

Binary 0 Æ The first half bit duration negative pulse and the second half Bit duration
positive pulse. Binary 1Æ first half bit duration positive pulse and the second half . Bit duration
negative pulse.
7. What is an eye pattern?[AUC 14]

Eye Pattern is used to study the effect of intersymbol interference.
8. What Are the Properties of matched filter.

1) The signal to noise ratio of the matched filter depends only upon the the ratio of the signal
energy to the psd of white noise at the filter input
2)The output signal of a matched filter is proportional to a shifted version of the auto_correlation
function of the input signal to which the filter is matched.
9. Why do we go for Gram-Schmidt Orthogonalization procedure?

Consider a message signal m. The task of transforming an incoming message
mi=1,2,…..M, into a modulated wave si(t) may be divided into separate discrete time &
continuous time operations.
10. What is matched filter receiver?

A filter whose impulse response is a time reversed & delayed version of some signal Æ j (t) then
it is said to be matched to Æj (t) correspondingly, the optimum receiver based on the detector is
referred to as the matched filter receiver.
34
11. What is maximum likelihood detector.[AUC

detector. 13]
Maximum likelihood detector computes the metric for each transmitted message
compares them and then decides in favor of maximum. The device for implementing the
decision rule i.e; set ^m = mi if In [ fx(x/mk)] is maximum for k=i is called maximum –likelihood
likelihood
detector and the decision rule is called maximum likelihood.
12. Define antipodal signals.

signals
A pair of sinusoidal signals that differ only in a phase shift of 180 degrees are referre
referred
d to as
antipodal signals.
PART B
1.Write briefly about eye pattern for data transmission
• In telecommunication,
telecommunication an eye pattern,, also known as an eye diagram,, is an
oscilloscope display in which a digital data signal from a receiver is repetitively sampled
and applied to the vertical input, while the data rate is used to trigger the horizontal
sweep.
• It is so called because, for several types of coding, the pattern looks like a series of
eyes between a pair of rails.
• It is an experimental tool for the evaluation of the combined effects of channel noise and
intersymbol interference on the performance of a baseband pulse
pulse-transmission system.
• It is the synchronised superposition of all possible realisations of the signal of interest
viewed within a particular signalling interval.
• Several system performance measures can be derived by analyzing the display.
• If the signals are too long, too short, poorly synchronized with the system clock, too
high, too
o low, too noisy,, or too slow to change, or have too much undershoot or
overshoot, this can be observed from the eye diagram.
• An open eye pattern corresponds to minimal signal distortion
distortion.
• Distortion of the signal waveform due to intersymbol interference and noise appears as
closure of the eye pattern.
There are many measurements that can be obtained from an Eye Diagram
Amplitude Measurements
• Eye Amplitude
• Eye Crossing Amplitude
• Eye Crossing Percentage
35
Time Measurements
• Deterministic Jitter
• Eye Crossing Time
• Eye Delay
• Eye Fall Time
• Eye Rise Time
• Eye Width
• Horizontal Eye Opening
• Peak-to-Peak Jitter
• Random Jitter
• RMS Jitter
2. What is known as ISI? Discuss the cause for ISI. Also explain Nyquist criteria for
distortionless Transmission
Intersymbol Interference
ISI on Eye Patterns
Combatting ISI
Nyquist’s First Method for zero ISI
Raised Cosine-Rolloff Pulse Shape
Nyquist Filter
• In telecommunication, intersymbol interference (ISI) is a form of distortion of a signal
in which one symbol interferes with subsequent symbols.
• This is an unwanted phenomenon as the previous symbols have similar effect as noise,
thus making the communication less reliable.
• ISI is usually caused by multipath propagation or the inherent non-linear frequency
response of a channel causing successive symbols to "blur" together.
• The presence of ISI in the system introduces errors in the decision device at the receiver
output.
• In the design of the transmitting and receiving filters, the objective is to minimize the
effects of ISI, and thereby deliver the digital data to its destination with the smallest error
rate possible.
• Intersymbol interference (ISI) occurs when a pulse spreads out in such a way that it
interferes with adjacent pulses at the sample instant.
36
• Example: assume polar NRZ line code. The channel outputs are shown as spreaded
(width Tb becomes 2Tb) pulses shown (Spreading due to bandlimited channel
characteristics).
• Three strategies for eliminating ISI:
o Use a line code that is absolutely bandlimited.
Would require Sinc pulse shape.
Can’t actually do this (but can approximate).
o Use a line code that is zero during adjacent sample instants.
It’s okay for pulses to overlap somewhat, as long as there is no overlap at
the sample instants.
Can come up with pulse shapes that don’t overlap during adjacent
sample instants.
3. Write a note on Base band M-ary PAM transmission

• Pulse-amplitude modulation (PAM), is a form of signal modulation where the message
information is encoded in the amplitude of a series of signal pulses.
• It is an analog pulse modulation scheme in which the amplitudes of a train of carrier
pulses are varied according to the sample value of the message signal.
• Demodulation is performed by detecting the amplitude level of the carrier at every
symbol period.
• Types
There are two types of pulse amplitude modulation:
1. Single polarity PAM: In this a suitable fixed DC bias is added to the signal to ensure that
all the pulses are positive.
2. Double polarity PAM: In this the pulses are both positive and negative.
37
Pulse-amplitude modulation is widely used in modulating signal transmission of digital data, with
non-baseband applications having been largely replaced by pulse-code modulation, and, more
recently, by pulse-position modulation.
In particular, all telephone modems faster than 300 bit/s use quadrature amplitude modulation
(QAM). (QAM uses a two-dimensional constellation).
• Such a signal may be available directly (usually not because of the waste involved in
sending a signal with no information content)
• Usually, the sample clock has to be derived directly from the received signal.
• The ability to extract a symbol timing clock usually depends upon the presence of
transitions or zero crossings in the received signal.
• Line coding aims to raise the number of such occurrences to help the extraction process.
• Unfortunately, simple line coding schemes often do not give rise to transitions when long
runs of constant symbols are received.
• Some line coding schemes give rise to a spectral component at the symbol rate
• A BPF or PLL can be used to extract this component directly
• Sometimes the received data has to be non-linearly processed eg, squaring, to yield a
component of the correct frequency.
4. Discuss on Adaptive equalization

• Need For Equalization:
– Overcome ISI degradation
• Need For Adaptive Equalization:
– Changing Channel in Time
– => Objective:
Find the Inverse of the Channel Response to reflect a ‘delta channel to the Rx
38
• Least Square Method:

– Unbiased estimator
– Exhibits minimum variance (optimal)
– No probabilistic assumptions (only signal model)
– Presented by Guass (1795) in studies of planetary motions)
Linear equalization (reminder):

• Tap delayed equalization
• Output is linear combination of the equalizer input
EQUALISATION
39
5. Discuss on Base band binary PAM system.
• PAM is a general signalling technique whereby pulse amplitude is used to convey the
message
• For example, the PAM pulses could be the sampled amplitude values of an analogue
signal
• We are interested in digital PAM, where the pulse amplitudes are constrained to chosen
from a specific alphabet at the transmitter
• In binary PAM, each symbol ak takes only two values, say {A1 and A2}
• In a multilevel, i.e., M-ary system, symbols may take M values {A1, A2 ,... AM}
• Signalling period, T
• To generate the PAM output signal, we may choose to represent the input to the
transmit filter hT(t) as a train of weighted impulse functions
40
6. Derive the expression for probability of error for matched filter.

A matched filter is a filter used in communications to “match” a particular transit
waveform.
It passes all the signal frequency components while suppressing any frequency
components where there is only noise and allows to pass the maximum amount of signal
power.
The purpose of the matched filter is to maximize the signal to noise ratio at the sampling
point of a bit stream and to minimize the probability of undetected errors received from a
signal.
To achieve the maximum SNR, we want to allow through all the signal frequency
components, but to emphasize more on signal frequency components that are large and
so contribute more to improving the overall SNR.
A basic problem that often arises in the study of communication systems is that of
detecting a pulse transmitted over a channel that is corrupted by channel noise (i.e.
AWGN)
Let us consider a received model, involving a linear time-invariant (LTI) filter of impulse
response h(t).
Goal of the linear receiver

To optimize the design of the filter so as to minimize the effects of noise at the
filter output and improve the detection of the pulse signal.
Signal to noise ratio is:
We sampled at t = T because that gives you the max power of the filtered signal.
Examine go(t):
Fourier transform
41
42
UNIT V
ERROR CONTROL CODING
PART A
1. Define spread spectrum communication.

In spread spectrum communication the transmitted data sequence occupies much more
band width than the minimum required band width. Special code is used to aspread the
bandwidth of the message signal. This special code is known only to authorized receiver.
2. What is pseudo noise sequence?[AUC 11]

Pseudo noise sequence is a noise like high frequency signal. The sequence is
not completely random, but it is generated by well defined logic. Hence it is called
pseudo noise sequence.
3. What is direct sequence spread spectrum modulation
In direct sequence spread spectrum modulation, the pseudo noise sequence is
directly modulated with data sequence. Thus pseudo noise sequence acts as high
frequency carrier and data sequence acts as low frequency modulating signal. The pseudo
noise sequence and data sequence are applied to a product modulator.
4.What is frequency hap spread spectrum modulation?
In frequency hop spread spectrum, the data is transmitted in different frequency slots.
These frequency slots are selected with the help of pseudoCnoise sequence. Selection of
frequency slots is called frequency hopping.
5.What is processing gain? [AUC 12]

processing gain is given as,
Processing gain (PG) = Bandwidth of spreaded signal / Bandwidth of unspreaded signal
6. State four applications of spread spectrum.

I) Spread spectrum has the ability to resist the effect of intentional jamming
ii) Spread spectrum is used in mobile communications. This is because the spread
spectrum signal has the ability to resist the effects of multipath fading.
iii) Spread spectrum communication are used in distance measurement.
iv) Spread spectrum communications are secure. This secrecy capability of spread
spectrum is used in military as well as in many commercial applications
7. When is the PN sequence called as maximal length sequence?

When the PN sequence has the length of 2m 1, it is called maximal length sequence.
8.What are the application of spread spectrum modulation.

Application
(i) Multipath access capability.
(ii) Multipath protection in mobile communication
(iii) Low probability intercept.
(iv) Interference rejection.
(v) To provide antijam capability.
43
9.Define frequency hopping.

The frequency of the carrier is changed (hopped) according to bits of PN sequence.
Types: I) Slow frequency hopping
II) Fast frequency hopping
10.What are the Disadvantages of DS-SS system.

1. It required wideband channel with small phase distortion.
2. It has long acquisition time.
3. The pseudo –noise generator should generate sequence at high rates.
4. This system is distance relative.
11. What are the Advantages of FH-SS System

1. These systems bandwidth (spreads) are very large
2. They can be programmed to avoid some portions of
the spectrum.
3. They have relatively short acquisition time.
4. The distance effect is less
12.What are the Application of Direct Sequence Spread Spectrum Signals

Antijamming with the help of direct sequence spread spectrum signals. Low delectability
signal transmission or low probability intercept. Code division multiple access with direct
sequence SS ( SSMA)
44
PART B
1.Discuss QPSK Signalling
• Phase-shift keying (PSK) is a digital modulation scheme that conveys data by changing,
or modulating, the phase of a reference signal (the carrier wave).
• Digital modulation scheme uses a finite number of distinct signals to represent digital
data.
• PSK uses a finite number of phases, each assigned a unique pattern of binary digits.
Usually, each phase encodes an equal number of bits. Each pattern of bits forms the
symbol that is represented by the particular phase.
• The demodulator, which is designed specifically for the symbol-set used by the
modulator, determines the phase of the received signal and maps it back to the symbol it
represents, thus recovering the original data.
• The receiver to be able to compare the phase of the received signal to a reference signal
— such a system is termed coherent (and referred to as CPSK).
• Alternatively, instead of operating with respect to a constant reference wave, the
broadcast can operate with respect to itself.
• Changes in phase of a single broadcast waveform can be considered the significant
items.
• In this system, the demodulator determines the changes in the phase of the received
signal rather than the phase (relative to a reference wave) itself.
• The scheme depends on the difference between successive phases, it is termed
differential phase-shift keying (DPSK).
• DPSK can be significantly simpler to implement than ordinary PSK since there is no
need for the demodulator to have a copy of the reference signal to determine the exact
phase of the received signal (it is a non-coherent scheme).
There are three major classes of digital modulation techniques used for transmission of digitally
represented data:
• Amplitude-shift keying (ASK)
• Frequency-shift keying (FSK)
• Phase-shift keying (PSK)
All convey data by changing some aspect of a base signal, the carrier wave (usually a sinusoid),
in response to a data signal.
In the case of PSK, the phase is changed to represent the data signal. There are two
fundamental ways of utilizing the phase of a signal in this way:
• By viewing the phase itself as conveying the information, in which case the demodulator
must have a reference signal to compare the received signal's phase against; or
• By viewing the change in the phase as conveying information — differential schemes,
some of which do not need a reference carrier (to a certain extent).
45
• A convenient way to represent PSK schemes is on a constellation diagram.

• This shows the points in the complex plane where, in this context, the real and imaginary
axis are termed the in-phase and quadrature axes respectively due to their 90°
separation.
• A representation on perpendicular axes lends itself to straightforward implementation.
• The amplitude of each point along the in-phase axis is used to modulate a cosine (or
sine) wave and the amplitude along the quadrature axis to modulate a sine (or cosine)
wave.
• In PSK, the constellation points chosen are usually positioned with uniform angular
spacing around a circle.
• It gives maximum phase-separation between adjacent points and thus the best
immunity to corruption.
• They are positioned on a circle so that they can all be transmitted with the same energy.
• In this way, the moduli of the complex numbers they represent will be the same and thus
so will the amplitudes needed for the cosine and sine waves.
• Two common examples are "binary phase-shift keying" (BPSK) which uses two phases,
and "quadrature phase-shift keying" (QPSK) which uses four phases, although any
number of phases may be used.
• The data to be conveyed are usually binary, the PSK scheme is usually designed with
the number of constellation points being a power of 2
2 .Explain the detection of binary FSK signal with block diagram.

• Frequency-shift keying (FSK) is a frequency modulation scheme in which digital
information is transmitted through discrete frequency changes of a carrier wave.
• The simplest FSK is binary FSK (BFSK). BFSK uses a pair of discrete frequencies to
transmit binary (0s and 1s) information.
• With this scheme, the "1" is called the mark frequency and the "0" is called the space
frequency.
• The time domain of an FSK modulated carrier is illustrated in the figures to the right.
OTHER FORMS OF FSK
MINIMUM-SHIFT KEYING
• Minimum frequency-shift keying or minimum-shift keying (MSK) is a particular spectrally
efficient form of coherent FSK.
• In MSK, the difference between the higher and lower frequency is identical to half the bit
rate.
46
• Consequently, the waveforms that represent a 0 and a 1 bit differ by exactly half a
carrier period.
• The maximum frequency deviation is δ = 0.25 fm, where fm is the maximum modulating
frequency.
• As a result, the modulation index m is 0.5.
• This is the smallest FSK modulation index that can be chosen such that the waveforms
for 0 and 1 are orthogonal.
• A variant of MSK called GMSK is used in the GSM mobile phone standard.
AUDIO FSK
• Audio frequency-shift keying (AFSK) is a modulation technique by which digital data is
represented by changes in the frequency (pitch) of an audio tone, yielding an encoded
signal suitable for transmission via radio or telephone.
• Normally, the transmitted audio alternates between two tones: one, the "mark",
represents a binary one; the other, the "space", represents a binary zero.
• AFSK differs from regular frequency-shift keying in performing the modulation at
baseband frequencies.
• In radio applications, the AFSK-modulated signal normally is being used to modulate an
RF carrier (using a conventional technique, such as AM or FM) for transmission.
• AFSK is not always used for high-speed data communications, since it is far less
efficient in both power and bandwidth than most other modulation modes.
• In addition to its simplicity, however, AFSK has the advantage that encoded signals will
pass through AC-coupled links, including most equipment originally designed to carry
music or speech.
• AFSK is used in the U.S. based Emergency Alert System to notify stations of the type of
emergency, locations affected, and the time of issue without actually hearing the text of
the alert.
47
3.Explain binary PSK signal with geometrical representation.
data. PSK uses a finite number of phases, each assigned a unique pattern of binary
digits.
48
• Usually, each phase encodes an equal number of bits.

• Each pattern of bits forms the symbol that is represented by the particular phase.
• The demodulator,
demodulator, which is designed specifically for the symbol
symbol-set
set used by the
modulator, determines the phase of the received signal and maps it back to the symbol it
represents, thus recovering the original data.
• PSK requires the receiver to be able to compare the phase of the received signal to a
reference signal — such a system is termed coherent (and referred to as CPSK).
• Alternatively, instead
instead of operating with respect to a constant reference wave, the
broadcast can operate with respect to itself.
• Changes in phase of a single broadcast waveform can be considered the significant
items
• In this system, the demodulator determines the changes in the phase of the received
signal rather than the phase (relative to a reference wave) itself.
• This scheme depends on the difference between successive phases, it is termed
differential phase-shift
phase keying (DPSK).
• DPSK can be significantly simpler to implement
implement than ordinary PSK since there is no
need for the demodulator to have a copy of the reference signal to determine the exact
phase of the received signal (it is a non-coherent
non coherent scheme).
• In exchange, it produces more erroneous demodulation.
CONSTELLATION
TION DIAGRAM EXAMPLE FOR BPSK.
• BPSK (also sometimes called PRK, phase reversal keying, or 2PSK) is the simplest form
of phase shift keying (PSK).
• It uses two phases which are separated by 180° and so can also be termed 2
2-PSK.
• It does not particularly matter
matter exactly where the constellation points are positioned, and
in this figure they are shown on the real axis, at 0° and 180°.
49
• This modulation is the most robust of all the PSKs since it takes the highest level of
noise or distortion to make the demodulator reach an incorrect decision.
• It is, however, only able to modulate at 1 bit/symbol (as seen in the figure) and so is
unsuitable for high data-rate
data applications.
• In the presence of an arbitrary phase-shift
shift introduced by the communications channel,,
the demodulator is unable to tell which constellation point is which.
• As a result, the data is often differentially encoded prior to modulation.
• BPSK is functionally equivalent to 2-QAM modulation.
4. Draw the block diagrams of MSK transmitter and receiver and explain the functions of
each block. Draw the constellation diagram. Derive probability of error.
• In digital modulation,
modulation minimum-shift keying (MSK
MSK) is a type of continuous-phase
phase
frequency-shift
shift keying that was developed in the late 1950s and 1960s.
• Similar to OQPSK,
OQPSK, MSK is encoded with bits alternating between quadrature
components, with the Q component delayed by half the symbol period.
• Instead of square pulses as OQPSK uses, MSK encodes each bit as a half sinusoid.
• This results in a constant-modulus
constant signal, which reduces problems caused by non
non-linear
linear
distortion.
• In addition to being viewed as related to OQPSK, MSK can also be viewed as a
continuous phase frequency shift keyed (CPFSK)
( ) signal w
with
ith a frequency separation of
one-half
half the bit rate.
• In MSK the difference between the higher and lower frequency is identical to half the bit
rate.
• The waveforms used to represent a 0 and a 1 bit differ by exactly half a carrier period.
• The maximum frequency
frequ deviation is = 0.25 fm where fm is the maximum modulating
frequency.
• The modulation index m is 0.5.
• The smallest FSK modulation index that can be chosen such that the waveforms for 0
and 1 are orthogonal.
orthogonal
• A variant of MSK called GMSK is used in the GSM mobile phone standard.
GAUSSIAN MINIMUM SHIFT KEYING
• In digital communication,
communication Gaussian minimum shift keying or GMSK is a continuous--
phase frequency-shift
frequency keying modulation scheme.
• It is similar to standard minimum-shift
minimum shift keying (MSK); however the digital data stream is
first shaped with a Gaussian filter before being applied to a frequency modulator.
50
• The advantage of reducing sideband power, which in turn reduces out-of-band

interference between signal carriers in adjacent frequency channels.
• The Gaussian filter increases the modulation memory in the system and causes
intersymbol interference, making it more difficult to differentiate between different
transmitted data values and requiring more complex channel equalization algorithms
such as an adaptive equalizer at the receiver.
• GMSK has high spectral efficiency, but it needs a higher power level than QPSK, for
instance, in order to reliably transmit the same amount of data.
• GMSK is most notably used in the Global System for Mobile Communications (GSM)
and the Automatic Identification System (AIS) for maritime navigation.
5. Discuss briefly about Minimum Shift Keying for a CPFSK signal. (MAY/JUNE 2009)
• In digital modulation, minimum-shift keying (MSK) is a type of continuous-phase
frequency-shift keying that was developed in the late 1950s and 1960s.
• Similar to OQPSK, MSK is encoded with bits alternating between quadrature
components, with the Q component delayed by half the symbol period.
• Instead of square pulses as OQPSK uses, MSK encodes each bit as a half sinusoid.
• In digital communication, Gaussian minimum shift keying or GMSK is a continuous-
phase frequency-shift keying modulation scheme.
• It is similar to standard minimum-shift keying (MSK); however the digital data stream is
first shaped with a Gaussian filter before being applied to a frequency modulator.
• The advantage of reducing sideband power, which in turn reduces out-of-band

interference between signal carriers in adjacent frequency channels.
• The Gaussian filter increases the modulation memory in the system and causes
intersymbol interference, making it more difficult to differentiate between different
transmitted data values and requiring more complex channel equalization algorithms
such as an adaptive equalizer at the receiver.
51
• GMSK has high spectral efficiency, but it needs a higher power level than QPSK, for
instance, in order to reliably transmit the same amount of data.
• GMSK is most notably used in the Global System for Mobile Communications (GSM)
and the Automatic Identification System (AIS) for maritime navigation.
6. Discuss Coherent PSK and DPSK.
data. PSK uses a finite number of phases, each assigned a unique pattern of binary
digits.
• Usually, each phase encodes an equal number of bits.
• Each pattern of bits forms the symbol that is represented by the particular phase.
• PSK requires the receiver to be able to compare the phase of the received signal to a
reference signal — such a system is termed coherent (and referred to as CPSK).
DIFFERENTIAL PHASE SHIFT KEYING

• An Optical DPSK demodulator is a device that provides a method for converting an
optical differential phase-shift keying (DPSK) signal to an intensity-keyed signal at the
receiving end in fiber-optic communication networks.
52
• It is also known as delay line interferometer (DLI), or simply called DPSK demodulator..
FIGURE : Working principle of optical DPSK demodulation: (a) Incoming DPSK signal with
uniform intensity, (b) 1-bit
1 bit delay of the incoming DPSK signal with un
uniform
iform intensity, and (c)
Demodulated intensity signal after interference between (a)/(a') and (b)/(b').
• The DPSK decoding method is achieved by comparing the phase of two sequential bits.
• An incoming DPSK optical signal is first split into two beams with equal intensities, in
which one beam is delayed in space by an optical path difference that introduces a time
delay corresponding to one bit.
• The two beams in the two paths are then coherently recombined to interfere each other
constructively or destructively.
• The interference intensity is measured and becomes the intensity
intensity-keyed signal.
• A typical optical system for such a purpose is Mach--Zehnder interferometer or Michelson
interferometer,, forming an
an optical DPSK Demodulator.
• Delay time depends on the data rate.
• For instance, in a 40 Gbit/s system, one bit corresponds to 25 picoseconds, and light
travels 5 mm in a fiber optics or 7.5 mm in free space within that period.
• Thus the optical path difference between the two beams is 5 mm or 7.5 mm depending
epending
on the type of interferometer used.
• DQPSK is the four-level
four version of DPSK.
• As a result, DQPSK allows processing of 40 Gbit/s data
data-rate in a 50 GHz channel
spacing system. A demodulator for optical DQPSK signals can be constructed using two
matched DPSK demodulators with phase off-set

off at .
53
54

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EC 6501 - DIGITAL COMMUNCATION V SEM/III YEAR/DEPT.

EC6501 DIGITAL COMMUNICATION LTPC

UNIT I SAMPLING & QUANTIZATION 9

Low pass sampling – Aliasing- Signal Reconstruction-Quantization - Uniform & non-uniform

UNIT II WAVEFORM CODING 9

UNIT III BASEBAND TRANSMISSION 9

UNIT IV DIGITAL MODULATION SCHEME 9

SAMPLING AND QUANTIZATION

1. State sampling theorem. [AUC APR/MAY 2011] [AUC APR/MAY 2012]

2. What is aliasing?[AUC APR 2011]

3. What is meant by PCM?

4. Define quantizing process.

6. What is meant by idle channel noise?

7. What is meant by prediction error?

8. Define delta modulation.

9. Define adaptive delta modulation.

10. Name the types of uniform quantizer?

11. Define quantization error?[AUC APR 14]

12. What you mean by non-uniform quantization? [AU-Apr‟2015]

• Analog to Digital Converter (A/D)

2. Derive Geometrical representation of signal.

• The set of coefficients can be viewed as a N-dimensional vector, denoted by si

3.Explain the mathematical models of communication channel

4. Distinguish between base band and bandpass signaling.

5. Classify channels. Explain the mathematical model of any two communication

• The model may reflect the following channel impairments:

DIGITAL METHODS OF GENERATING PWM

ANALOG GENERATION OF PWM

1.What is the disadvantage of uniform quantization over the non-uniform quantization?

2. What do you mean by companding? Define compander.[AUC ND 12]

4. What is the need for speech coding at low bit rates?

6. What is meant by forward and backward estimation?[AUC ND 15]

7. What are the limitations of forward estimation with backward estimation?

8. How are the predictor coefficients determined?

9. Define adaptive subband coding?[AUC APR 15]

10. What are formant frequencies?

11. What is the bit rate in ASBC?

12. Define Adaptive filter?

Sampling: obtain samples of x(t) at uniformly spaced time intervals

If the pulse is lower in value, resulting in a negative value, a “0” is used.

The analog signal is quantized by a one-bit ADC (a comparator implemented as a

• Adaptive delta modulation

– The value of δ changes according to the amplitude of the analog signal.

4. Explain uniform and non uniform quantization

6. Explain a spectral waveform encoding process

Uses special techniques

LINEAR PREDICTION CODING

REDUCE THE SAMPLING RATE

2. What is code rate?[AUC 12]

3. Define code efficiency.

4. What is hamming distance?[AUC APR 15]

5. What is meant by systematic & non-systematic code?

7.. What is BCH Code ?[AUC 11]

8. What are the conditions to satisfy the hamming code.

9. Define code word & block length.

10. Give the parameters of RS codes .

11. Why RS codes are called maximum distance separable codes ?

12. What are Golay codes ?[AUC 14]

1. Explain linear block coding