EC6501Digital Communication
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O/P Signal
Destination
and Output RECEIVER
Transducer
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Channel
Received Signal
User of Source Channel
Demodulator
Information Decoder Decoder
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The Source encoder ( or Source coder) converts the input i.e. symbol
sequence into a binary sequence of 0 s and 1 s by assigning code words to
the symbols in the input sequence. For eg. :If a source set is having
hundred symbols, then the number of bits used to represent each symbol will
be 7 because 27=128 unique combinations are available. The important
parameters of a source encoder are block size, code word lengths, average
data rate and the efficiency of the coder (i.e. actual output data rate
compared to the minimum achievable rate)
At the receiver, the source decoder converts the binary output of the
channel decoder into a symbol sequence. The decoder for a system using
fixed length code words is quite simple, but the decoder for a system using
variable length code words will be very complex.
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The Channel decoder recovers the information bearing bits from the coded
binary stream. Error detection and possible correction is also performed by
the channel decoder.
The important parameters of coder / decoder are: Method of coding,
efficiency, error control capabilities and complexity of the circuit.
MODULATOR:
The Modulator converts the input bit stream into an electrical
waveform suitable for transmission over the communication channel.
Modulator can be effectively used to minimize the effects of channel noise,
to match the frequency spectrum of transmitted signal with channel
characteristics, to provide the capability to multiplex many signals.
DEMODULATOR:
The extraction of the message from the information bearing waveform
produced by the modulation is accomplished by the demodulator. The
output of the demodulator is bit stream. The important parameter is the
method of demodulation.
CHANNEL:
The Channel provides the electrical connection between the source
and destination. The different channels are: Pair of wires, Coaxial cable,
Optical fibre, Radio channel, Satellite channel or combination of any of
these.
The communication channels have only finite Bandwidth, nonideal
frequency response, the signal often suffers amplitude and phase distortion
as it travels over the channel. Also, the signal power decreases due to the
attenuation of the channel. The signal is corrupted by unwanted,
unpredictable electrical signals referred to as noise.
The important parameters of the channel are Signal to Noise power
Ratio (SNR), usable bandwidth, amplitude and phase response and the
statistical properties of noise.
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Optical Fibers: An optical fiber consists of a very fine inner core made of
silica glass, surrounded by a concentric layer called cladding that is also
made of glass. The refractive index of the glass in the core is slightly higher
than refractive index of the glass in the cladding. Hence if a ray of light is
launched into an optical fiber at the right oblique acceptance angle, it is
continually refracted into the core by the cladding. That means the
difference between the refractive indices of the core and cladding helps
guide the propagation of the ray of light inside the core of the fiber from one
end to the other.
Compared to coaxial cables, optical fibers are smaller in size and they
offer higher transmission bandwidths and longer repeater separations.
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Bandwidth:
Bandwidth is simply a measure of frequency range. The range of
frequencies contained in a composite signal is its bandwidth. The bandwidth
is normally a difference between two numbers. For example, if a composite
signal contains frequencies between 1000 and 5000, its bandwidth is 5000 
1000, or 4000. If a range of 2.40 GHz to 2.48 GHz is used by a device, then
the bandwidth would be 0.08 GHz (or more commonly stated as 80MHz).It
is easy to see that the bandwidth we define here is closely related to the
amount of data you can transmit within it  the more room in frequency
space, the more data you can fit in at a given moment. The term bandwidth
is often used for something we should rather call a data rate, as in my Internet
connection has 1 Mbps of bandwidth , meaning it can transmit data
at 1 megabit per second.
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then it is completely described by instantaneous values g(kT s) uniformly spaced in time
with period Ts 1/2W. Ifthe sampling rate, fs is equaltotheNyquist rateor greater (fs
2W), the signal g(t) can be exactly reconstructed.
g(t)
s (t)
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Proof:
Part  I If a signal x(t) does not contain any frequency component beyond W
Hz, then the signal is completely described by its instantaneous uniform
samples with sampling interval (or period ) of Ts < 1/(2W) sec.
Part II The signal x(t) can be accurately reconstructed (recovered) from the
set of uniform instantaneous samples by passing the samples sequentially
through an ideal (brickwall) lowpass filter with bandwidth B, where W B
< fs W and fs = 1/(Ts).
1.3
Now, from the theory of Fourier Transform, we know that the F.T of
(t nTs), the
impulse train in time domain, is an impulse train in frequency domain:
If Xs(f) denotes the Fourier transform of the energy signal xs(t), we can
write using Eq. (1.2.4) and the convolution property:
1.5
This equation, when interpreted appropriately, gives an intuitive proof to
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Fig. 1.2.1 indicates that the bandwidth of this instantaneously sampled wave
xs(t) is infinite while the spectrum of x(t) appears in a periodic manner,
centered at discrete frequency values n.fs.
Part I of the sampling theorem is about the condition fs > 2.W i.e. (fs W)
> W and ( fs + W) < W. As seen from Fig. 1.2.1, when this condition is
satisfied, the spectra of xs(t), centered at f = 0 and f = fs do not overlap
and hence, the spectrum of x(t) is present in xs(t) without any distortion.
This implies that xs(t), the appropriately sampled version of x(t), contains all
information about x(t) and thus represents x(t).
The second part suggests a method of recovering x(t) from its sampled
version xs(t) by using an ideal lowpass filter. As indicated by dotted lines in
Fig. 1.2.1, an ideal lowpass filter (with brickwall type response) with a
bandwidth W B < (fs W), when fed with xs(t), will allow the portion
of
Xs(f), centered at f = 0 and will reject all its replicas at f = n fs, for n 0.
This implies that the shape of the continuous time signal xs(t), will be
retained at the output of the ideal filter.
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The signal g(t) can be represented by instantaneous values, g(kTs) if the sampling
rate fs is (2fu/m) where m is an integer defined as
((fu / B) 1 ) < m (fu / B)
If the sample values are represented by impulses, then g(t) can be exactly
reproduced fromit s samplesbyan idealBandPass filter with the response, H(f) defined
as
H(f) = 1 fl <  f  <fu
0 elsewhere
If the sampling rate, fs 2fu, exact reconstruction is possible in which case the signal
fs
4B
3B
2B
0 B 2B 3B 4B 5B fu
Fig 2.3: Relation between Sampling rate, Upper cutoff frequency and Bandwidth.
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Example1 :
Consider a signal g(t) having the Upper Cutoff frequency, fu = 100KHz and the
Lower Cutoff frequency fl = 80KHz.
The ratio of upper cutoff frequency to bandwidth of the signal g(t) is
fu / B = 100K / 20K = 5.
Therefore we can choose m = 5.
Then the sampling rate is fs = 2fu / m = 200K / 5 = 40KHz
Example2 :
Consider a signal g(t) having the Upper Cutoff frequency, fu = 120KHz and the
Lower Cutoff frequency fl = 70KHz.
The ratio of upper cutoff frequency to bandwidth of the signal g(t) is
fu / B = 120K / 50K = 2.4
Therefore we can choose m = 2. ie.. m is an integer less than (fu /B).
Then the sampling rate is fs = 2fu / m = 240K / 2 = 120KHz.
fc, (fc>W). The inphase component, gI(t) is obtained by multiplying g(t) with
cos(2 fct) and then filtering out the high frequency components. Parallelly a quadrature phase
component is obtained by mult iplying g(t) with sin(2 fct) and then filtering out the
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sampler
sin(2 fct)
G(f)
fc 0 fc f
2W>
a) Spectrum of a Band pass signal.
GI(f) / GQ(f)
W 0 W f
b) Spectrum of gI(t) and gQ(t)
Fig 2.5 a) Spectrum of Bandpass signal g(t)
b) Spectrum of inphase and quadrature phase signals
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RECONSTRUCTION:
From the sampled signals gI(nTs) and gQ(nTs), the signals g I(t) and gQ(t) are
obtained. To reconstruct the original band pass signal, multiply the signals g I(t) by
cos(2 fct) and sin(2 fct) respectivelyand thenadd the results.
gI(nTs)
Reconstruction
Filter
+
Cos(2 fct) g(t)

gQ(nTs) Reconstruction
Filter
Sin (2 fct)
In both the natural sampling and flattop sampling methods, the spectrum of the signals
are scaled bytheratio /Ts, where is thepulse duration and Ts is the sampling period.
Since this ratio is very small, the signal power at the output of the reconstruction filter is
correspondingly small. To overcome this problem a sampleandhold circuit is used .
SW
AMPLIFIER
Input Output
g(t) u(t)
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The SampleandHold circuit consists of an amplifier of unity gain and low output
impedance, a switch and a capacitor; it is assumed that the load impedance is large. The
switch is timed to close only for the small duration of each sampling pulse, during which
time the capacitor charges up to a voltage level equal to that of the input sample. When
the switch is open , the capacitor retains the voltage level until the next closure of the
switch. Thus the sampleandhold circuit produces an output waveform that represents a
staircase interpolation of the original analog signal.
Natural Sampling:
In this method of sampling, an electronic switch is used to periodically shift
between the two contacts at a rate of fs = (1/Ts ) Hz, staying on the input contact for C
seconds and on the grounded contact for the remainder of each sampling period.
The output xs(t) of the sampler consists of segments of x(t) and hence x s(t) can be
considered as the product of x(t) and sampling function s(t).
xs(t) = x(t) . s(t)
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n 0
1 X(f)
f
W 0 +W
Xs(f)
C0
C2 C1 C1 C2
f
2fs fs W 0 +W fs 2fs
Sampled Signal Spectrum (fs > 2W)
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The signal xs(t) has the spectrum which consists of message spectrum and repetition of
message spectrum periodically in the frequency domain with a period of fs. But the
message term is scaled by Co . Since the spectrum is not distorted it is possible to
reconstruct x(t) from the sampled waveform x s(t).
Quantization:
The process of transforming Sampled amplitude values of a message signal into a
discrete amplitude value is referred to as Quantization.
A quantizer is memory less in that the quantizer output is determined only by the value of
a corresponding input sample, independently of earlier analog samples applied to the
input.
Analog Signal
Discrete Samples
( Quantized )
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Types of Quantizers:
1. Uniform Quantizer
2. Non Uniform Quantizer
Uniform Quantizer:
In Uniform type, the quantization levels are uniformly spaced, whereas in non
uniform type the spacing between the levels will be unequal and mostly the relation is
logarithmic.
In the stair case like graph, the origin lies the middle of the tread portion in Mid Tread
type where as the origin lies in the middle of the rise portion in the MidRise type.
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Output
7/2
5/2
3/2
/2
2 3 4 Input
Output
/2 3/2 Input
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The Quantizationprocess introduces anerror defined asthe difference betweenthe input signal,
x(t) and theoutput signal, yt). This error is called the QuantizationNoise.
Let bethe step size ofa quantizer and L bethe totalnumber ofquantization levels.
Quantization levels are 0, ., 2 ., 3 . . . .. ..
The Quantization error, Q is a random variable and will have its sample values bounded
by [( /2) < q < ( /2)]. If is small, the quantization error can be assumed to a
uniformly distributed random variable.
which is a staircase function that befits the type of mid tread or mid riser quantizer of
interest.
UNIFORM
COMPRESSOR QUANTIZER EXPANDER
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1. Higher average signal to quantization noise power ratio than the uniform quantizer
when the signal pdf is non uniform which is the case in many practical situation.
2. RMS value of the quantizer noise power of a non uniform quantizer is
substantially proportional to the sampled value and hence the effect of the
quantizer noise is reduced.
Encoding:
Encoding is the final step in what we call analogtodigital (A/D)
conversion.Sometimes, encoding is considered as a separate stage that
follows the A/D stage. The encoding stage takes a bit stream of 1 s and
0 s
and converts it into voltages appropriate for transmission on a physical
channel. Encoding schemes are usually divided into: Source encoding,
Channel encoding and Line encoding .
Source coding:
Source coding (sometimes called entropy encoding) refers to the process of
compressing data. This is typically done by replacing long binary codes
(named codewords) that occur frequently by shorter ones, and those that
occur less frequently by longer codes. For example, a 4bit sequence 0110
occurring frequently can be mapped into the shorter 2bit 01 sequence,
while another 4bit sequence 1011 occurring less frequently can be
mapped to the longer 7bit sequence 0011011 . This makes sure that
shorter sequences occur more often in the bit stream.
In information theory, Shannon's noiseless coding theorem places an upper
and a lower bound on the expected compression ratio. Examples of source
codes currently in use are: Shannon codes, Huffman codes, runlength
coding, arithmetic coding, LempelZiv coding, MPEG video coding, etc.
Channel coding:
A channel code is a broadly used term mostly referring to error
correcting codes. Such codes are used to protect data sent over the channel
from corruption even in the presence of noise. In other words, channel codes
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can improve the signaltonoise ratio (SNR) of the received signal. The most
obvious example of such codes is the simple parity bit system.
The theory behind designing and analyzing channel codes is called
Shannons noisy channel coding theorem. It puts an upper limit on the
amount of information you can send in a noisy channel using a perfect
channel code. This is given by the following equation:
where C is the upper bound on the capacity of the channel (bit/s), B is the
bandwidth of the channel (Hz) and SNR is the SignaltoNoise ratio
(unitless). Examples of channel codes currently inuse include: Hamming
codes, ReedSolomon codes, convolutional codes (usually decoded by an
iterative Viterbi decoder), Turbo codes, etc.
The transmitter and receiver of the DPCM scheme is shown in the below
fig.respectively.
Transmitter: Let x(t) be the signal to be sampled and x(nTs) be it s samples. In this
scheme the input to the quantizer is a signal
where x^(nTs) is the prediction for unquantized sample x(nTs). This predicted value is
produced by using a predictor whose input, consists of a quantized versions of the input
signal x(nTs). The signal e(nTs) is called the prediction error.
By encoding the quantizer output, in this method, we obtain a modified version of the
PCM called differential pulse code modulation (DPCM).
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u(nTs) = x^(nTs) + v(nTs)  (3.33)
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The receiver consists of a decoder to reconstruct the quantized error signal. The quantized
version of the original input is reconstructed from the decoder output using the same
predictor as used in the transmitter. In the absence of noise the encoded signal at the
receiver input is identical to the encoded signal at the transmitter output. Correspondingly
the receive output is equal to u(nTs), which differs from the input x(nts) only by the
quantizing error q(nTs).
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Sampled Input
x(nTs) e(nTs) v(nTs) Output
Quantizer
+
^
x(nTs)
Predictor
u(nTs)
x^(nTs Predictor
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DM provides a staircase approximation to the over sampled version of an input base band
signal. The difference between the input and the approximation is quantized into only two
levels, namely, corresponding to posit ive and negativedifferences, respectively, Thus,
if the approximation falls below the signal at any sampling epoch, it is increased by .
Provided that the signal does not change too rapidly from sample to sample, we find that
the stair case approximatio nremains within ofthe input signal. The symbol denotes
the absolute value of the two representation levels of the onebit quantizer used in the
DM.
Output
0 Input
Let the input signal be x(t) and the staircase approximation to it is u(t).
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Sampled Input
x(nTs) e(nTs) b(nTs) Output
One  Bit
+ Quantizer
^
x(nTs)
Delay
Ts u(nTs)
Block diagram for Transmitter of a DM system
In the receiver the stair case approximation u(t) is reconstructed by passing the incoming
sequence of positive and negative pulses through an accumulator in a manner similar to
that used in the transmitter. The outof band quantization noise in the high frequency
staircase waveform u(t) is rejected by passing it through a lowpass filter with a band
width equal to the original signal bandwidth.
Delay
Ts
u(nTsTs)
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Disadvantage of DM:
There are several types of ADM, depending on the type of scheme used for adjusting the
step size. In this ADM, a discrete set of values is provided for the step size.
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PCM and ADPCM are both timedomain coders in that the speech signal is
processed in the timedomain as a single full band signal. Adaptive subband coding
is a frequency domain coder, in which the speech signal is divided into a number of
subbands and each one is encoded separately. The coder is capable of digitizing
speech at a rate of 16 kb/s with a quality comparable to that of 64 kb/s PCM. To
accomplish this performance, it exploits the quasiperiodic nature of voiced speech
and a characteristic of the hearing mechanism known as noise masking.
Periodicity of voiced speech manifests itself in the fact that people speak with a
characteristic pitch frequency. This periodicity permits pitch prediction, and
therefore a further reduction in the level of the prediction error that requires
quantization, compared to differential pulse code modulation without pitch
prediction. The number of bits per sample that needs to be transmitted is thereby
greatly reduced, without a serious degradation in speech quality.
Applications
1. Hierarchy of Digital Multiplexers
2. Light wave Transmission Link
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Unit3
Baseband Transmission
Baseband Signal  The information signal is called the baseband signal. The
bandwidth is always a positive quantity so the bandwidth of this signal is fm.
Block Codes:
Block codes operate on a block of bits.Block codes are referred to as (n, k)
codes. A block of k information bits are coded to become a block of n bits.
But before we go any further with the details, let s look at an important
concept in coding called Hamming distance.
Let s say that we want to code the 10 integers, 0 to 9 by a digital sequence.
Sixteen unique sequences can be obtained from four bit words. We assign
the first ten of these, one to each integer. Each integer is now identified by
its own unique sequence of bits .
Hamming Weight: The Hamming weight of this code scheme is the largest
number of 1 s in a valid codeword. This number is 3 among the 10
codewords we have chosen. (the ones in the while space)
Concept of Hamming Distance:
In continuous variables, we measure distance by Euclidean concepts such as
lengths, angles and vectors.In the binary world, distances are measured
between two binary words by something called the Hamming distance. The
Hamming distance is the number of disagreements between two binary
sequences of the same size. The Hamming distance between sequences 001
and 101 is = 1 The Hamming distance between sequences 0011001 and
1010100 is = 4.
Hamming distance and weight are very important and useful concepts in
coding. The knowledge of Hamming distance is used to determine the
capability of a code to detect and correct errors. Although the Hamming
weight of our chosen code set is 3, the minimum Hamming distance is 1.
We can generalize this to say that the maximum number of error bits that
can be detected is
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t = dmin 1
Where dmin is Hamming distance of the codewords. For a code with
dmin = 3, we can both detect 1 and 2 bit errors. So we want to have a code
set with as large a Hamming distance as possible since this directly effects
our ability to detect errors.
The number of errors that we can correct is given by
ReedSolomon Codes:
Reed Solomon (RS) codes form an important subclass of the family of
Bose ChaudhuriHocquenghem (BCH) codes and are very powerful linear
nonbinary block codes capable of correcting multiple random as well as
burst errors. They have an important feature that the generator polynomial
and the code symbols are derived from the same finite field. This enables to
reduce the complexity and also the number of computations involved in their
implementation. A large number of RS codes are available with different
code rates.
An RS code is described by a generator polynomial g(x) and other usual
important code parameters such as the number of message symbols per
block (k), number of code symbols per block (n), maximum number o f
erroneous symbols (t) that can surely be corrected per block of received
symbols and the designed minimum symbol Hamming distance (d). A
paritycheck polynomial h (X) of order k also plays a role in designing the
code. The symbol x, used in polynomials is an indeterminate which usually
implies unit amount of delay.
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Convolutional codes:
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Fig 3.2 State diagram representation for the encoder in Fig. 6.35.1
State diagram
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35
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H(7,4)
Generator matrix G: first 4by4 identical matrix
Transmission vector x
Received vector r
and error vector e
Parity check matrix H
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Error Correction:
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CLASSIFICATION OF LINECODES
Line coding:
Line coding refers to the process of representing the bit stream
(1 s and 0 s) in the form of voltage or current variations optimally tuned for
the specific properties of the physical channel being used.
The selection of a proper line code can help in so many ways: One
possibility is to aid in clock recovery at the receiver. A clock signal is
recovered by observing transitions in the received bit sequence, and if
enough transitions exist, a good recovery of the clock is guaranteed, and the
signal is said to be selfclocking.
Another advantage is to get rid of DC shifts. The DC component in a line
code is called the bias or the DC coefficient. Unfortunately, most long
distance communication channels cannot transport a DC component. This is
why most line codes try to eliminate the DC component before being
transmitted on the channel.Such codes are called DC balanced, zeroDC,
zerobias, or DC equalized.Some common types of line encoding in
commonuse nowadays are unipolar, polar, bipolar, Manchester, MLT3 and
Duobinary encoding. These codes are explained here:
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The rms value of polar signals is bigger than unipolar signals, which means
that polar signals have more power than unipolar signals, and hence have
better SNR at the receiver. Actually, polar NRZ signals have more power
compared to polar RZ signals.
The drawback of polar NRZ, however, is that it lacks clock information
especially when a long sequence of 0 s or 1 s is transmitted.
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This is the code used on compact discs (CD), USB ports, and on fiberbased
Fast Ethernet at 100Mbit/s .
Manchester encoding:
In Manchester code each bit of data is signified by at least one transition.
Manchester encoding is therefore considered to be selfclocking, which
means that accurate clock recovery from a data stream is possible. In
addition, the DC component of the encoded signal is zero. Although
transitions allow the signal to be selfclocking, it carries significant overhead
as there is a need for essentially twice the bandwidth of a simple NRZ or
NRZI encoding
Unipolar most of signal power is centered around origin and there is waste of
power due to DC component that is present.
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Polar format most of signal power is centered around origin and they are simple
to implement.
Bipolar format does not have DC component and does not demand more
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Unit 4
Analog signal: If the magnitudes of a real signal s(t) over its range of
definition, T1 t T2, are real numbers (there are infinite such values)
within a finite range, say, Smin S(t) Smax, the signal is analog. A digital
signal s(t), on the contrary, can assume only any of a finite number of
values. Usually, a digital signal implies a discretetime, discreteamplitude
signal.
i.e. the energy of the signal is finite,the signal is called an energy signal.
However, the same signal may have large power.The voltage generated by
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Power signal: A power signal, on the contrary, will have a finite power but
may have finite or infinite energy. Mathematically,
While electrical signals, derived from physical processes are mostly energy
signals,several mathematical functions, usually deterministic, represent
power signals.
Periodic signal: If s(t) = s(t + T), for entire range of t over which the signal
s(t) is defined and T is a constant, s(t) is said to be periodic or repetitive. T
indicates the period of the signal and 1/T is its frequency of repetition.
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BASEBAND:
RECEIVING FILTER:
Correlative receiver
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Observation
Vector x
For an AWGN channel and for the case when the transmitted signals are equally likely,
the optimum receiver consists of two subsystems
1) .Receiver consists of a bank of M productintegrator or correlators
1(t) , 2(t) . M(t) orthonormal function
The bank of correlator operate on the received signal x(t) to produce observation vector x
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MATCHED FILTER
Science each of t he orthonormal basic functions are 1(t) , 2(t) . M(t) is assumed
to be zero outside the interval 0<t<T. we can design a linear filter with impulse response
hj(t), with the received signal x(t) the fitter output is given by the convolution integral
yj(t) = xj
where xj is the j th correlator output produced by the received signal x(t).
A filter whose impulse response is timereversed and delayed version of the input signal
is said to be matched to j (t) . correspondingly , the optimum receiver based on this is
referred as the matched filter receiver.
For a matched filter operating in real time to be physically realizable, it must be causal.
For causal system
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MATCHED FILTER
The impulse response of the matched filter is timereversed and delayed version of the
input signal
PROPERTY 2
PROPERTY 3
The output Signal to Noise Ratio of a Matched filter depends only on the ratio of the
signal energy to the power spectral density of the white noise at the filter input.
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PROPERTY 4
The Matched Filtering operation may be separated into two matching conditions; namely
spectral phase matching that produces the desired output peak at time T, and the spectral
amplitude matching that gives this peak value its optimum signal to noise density ratio.
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Fig: Illustrating the effect of noise perturbation on location of the received signal point
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uncontrolled. This problem of ISI exists strongly in Telephone channels like coaxial
cables and optical fibers.
In this chapter main objective is to study the effect of ISI, when digital data is
transmitted through band limited channel and solution to overcome the degradation of
waveform by properly shaping pulse.
1 0 11
Tb
Transmitted Waveform Pulse Dispersion
The effect of sequence of pulses transmitted through channel is shown in fig. The
Spreading of pulse is greater than symbol duration, as a result adjacent pulses interfere.
i.e. pulses get completely smeared, tail of smeared pulse enter into adjacent symbol
intervals making it difficult to decide actual transmitted pulse.
First let us have look at different formats of transmitting digital data.In base band
transmission best way is to map digits or symbols into pulse waveform. This waveform
is generally termed as Line codes.
EYE PATTERN
The quality of digital transmission systems are evaluated using the bit error rate.
Degradation of quality occurs in each process modulation, transmission, and detection.
The eye pattern is experimental method that contains all the information concerning the
degradation of quality. Therefore, careful analysis of the eye pattern is important in
analyzing the degradation mechanism.
Eye patterns can be observed using an oscilloscope. The received wave is applied
to the vertical deflection plates of an oscilloscope and the sawtooth wave at a rate
equal to transmitted symbol rate is applied to the horizontal deflection plates,
resulting display is eye pattern as it resembles human eye.
The interior region of eye pattern is called eye opening
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The width of the eye opening defines the time interval over which the received
wave can be sampled without error from ISI
The height of the eye opening at a specified sampling time is a measure of the
margin over channel noise.
The sensitivity of the system to timing error is determined by the rate of closure of the
eye as the sampling time is varied.
Any non linear transmission distortion would reveal itself in an asymmetric or squinted
eye. When the effected of ISI is excessive, traces from the upper portion of the eye
pattern cross traces from lower portion with the result that the eye is completely closed.
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EQUALISING FILTER
Adaptive equalization
Adaptive equalization It consists of tapped delay line filter with set of delay elements,
set of adjustable multipliers connected to the delay line taps and a summer for adding
multiplier outputs.
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Y(nt)= Ci x(nTiT)
Ci is weight of the ith tap Total number of taps are M .Tap spacing is equal to symbol
duration T of transmitted signal
In a conventional FIR filter the tap weights are constant and particular designed response
is obtained. In the adaptive equaliser the Ci's are variable and are adjusted by an
algorithm
Mechanism of adaptation
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Training mode
This training sequence has maximal length PN Sequence, because it has large average
power and large SNR, resulting response sequence (Impulse) is observed by measuring
the filter outputs at the sampling instants.
The difference between resulting response y(nT) and desired response d(nT)is error
signal which is used to estimate the direction in which the coefficients of filter are to be
optimized using algorithms
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The BASK system has one dimensional signal space with two messages (N=1, M=2)
Region E2 Region E1
Message
Point 2 Eb
(t)
0 Eb
Message
2
Point 1
Signal Space representation of BASK signal
wave [in unipolar form] and sinusoidal carrier are applied to a product modulator. The
desired BASK wave is obtained at the modulator output.
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In demodulator, the received noisy BASK signal x(t) is apply to correlator with
coherent reference signal. The correlator output x is compared withthreshold .
Ifx> the receiver decides in favourofsymbol1. Ifx<
thereceiver decides in favourofsymbol0.
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FSK transmitter
fig b
FSK receiver
symbol 1 and 0 volts for symbol 0 . When we have symbol 1 the upper channel
is
switched on with oscillator frequency f1, for symbol 0 , because of inverter the lower
channel is switched on with oscillator frequency f2 . These two frequencies are combined
using an adder circuit and then transmitted. The transmitted signal is nothing but
required BFSK signal.
The detector consists of two correlators. The incoming noisy BFSK signal x(t) is
common to both correlator. The Coherent reference signal 1(t) & 2(t) are supplied to
upper and lower correlators respectively.
The correlator outputs are then subtracted one from the other and resulting a
randomvector l (l=x1  x2). Theoutput l iscompared withthreshold ofzero volts.
If l > 0, the receiver decides in favour of symbol 1.
l < 0, the receiver decides in favour of symbol 0.
FSK Bandwidth:
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Applications
On voicegrade lines, used up to 1200bps
Used for highfrequency (3 to 30 MHz) radio transmission
used at higher frequencies on LANs that use coaxial cable.
Therefore Binary FSK system has 2 dimensional signal space with two messages
S1(t) and S2(t), [N=2 , m=2] they are represented,
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Non Return to
Product
Zero Level
Encoder Modulator
Binary Binary PSK Signal
Data Sequence
2
1 (t) Cos2 f c t
Tb
x(t) Tb
x1 Decision Choose 1 if x1>0
dt
0
Device
Choose 0 if x1<0
Correlator
1 (t) Threshold =0
In a Coherent binary PSK system the pair of signals S 1(t) and S2(t) are used to
represent binarysymbol 1 and 0 respectively.
To generate a binary PSK signal we have to represent the input binary sequence in polar
formwit h symbol 1 and 0 represented byconstant amplitude levels
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Region R2 Region R1
 Eb Eb
0
Message Point 2 Message Point 1
S2(t) Decision S1(t)
Boundary
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QPSK Receiver:
The QPSK receiver consists of a pair of correlators with a common input and supplied
with a locally generated pair of coherent reference signals 1(t) & 2(t)as shown in
fig(b).The correlator outputs x1 and x2 produced in response to the received signal x(t) are
each compared with a threshold value of zero.
The inphase channel output :
If x1 > 0 a decision is made in favour of symbol 1
x1 < 0 a decision is made in favour of symbol 0
Similarly quadrature channel output:
If x2 >0 a decision is made in favour of symbol 1 and
x2 <0 a decision is made in favour of symbol 0
Finally these two binary sequences at the in phase and quadrature channel outputs are
combined in a multiplexer (Parallel to Serial) to reproduce the original binary sequence.
Probability of error:
A QPSK system is in fact equivalent to two coherent binary PSK systems
working in parallel and using carriers that are inphase and quadrature.
The inphase channel output x1 and the Qchannel output x2 may be viewed as
the individual outputs of the two coherent binary PSK systems. Thus the two binary
PSK
systems may be characterized as follows.
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2
The bit errors in the Ichannel and Qchannel of the QPSK system are statistically
independent . The Ichannel makes a decision on one of the two bits constituting a
symbol (di bit) of the QPSK signal and the Qchannel takes care of the other bit.
As an example of QAM, 12 different phases are combined with two different amplitudes.
Since only 4 phase angles have 2 different amplitudes, there are a total of 16
combinations.
With 16 signal combinations, each baud equals 4 bits of information (2 ^ 4 = 16).
Combine ASK and PSK such that each signal corresponds to multiple bits.
More phases than amplitudes. Minimum bandwidth requirement same as ASK or PSK
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i)Channel coding
ii)Block Coding
7.Define Bandwidth
Bandwidth is simply a measure of frequency range. The range of
frequencies contained in a composite signal is its bandwidth. The bandwidth
is normally a difference between two numbers.
8.Define Power signal
A power signal, will have a finite power but may have finite or infinite
energy. Mathematically it is expressed as,
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Matched filter is used for detection of signal in base band and pass band
transmission. A filter whose impulse response is a time reversed & delayed version of
some signal ,then it is said to be matched to correspondingly, the optimum receiver
based on the detector is referred to as the matched filter receiver.
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vector si. The set of message points corresponding to the set of transmitted signals {si(t))}
i=1..M is called a signal constellation.
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