EXPERIMENT -1

TO STUDY AMPLITUDE SHIFT KEYING USING MATLAB

clc %for clearing the command window

close all %for closing all the window except command window
clear all %for deleting all the variables from the memory
fc=input('Enter the freq of Sine Wave carrier:');
fp=input('Enter the freq of Periodic Binary pulse (Message):');
amp=input('Enter the amplitude (For Carrier & Binary Pulse Message):');
t=0:0.001:1; % For setting the sampling interval
c=amp.*sin(2*pi*fc*t);% For Generating Carrier Sine wave
subplot(3,1,1) %For Plotting The Carrier wave
plot(t,c)
xlabel('Time')
ylabel('Amplitude')
title('Carrier Wave')
m=amp/2.*square(2*pi*fp*t)+(amp/2);%For Generating Square wave message
subplot(3,1,2) %For Plotting The Square Binary Pulse (Message)
plot(t,m)
xlabel('Time')
ylabel('Amplitude')
title('Binary Message Pulses')
w=c.*m; % The Shift Keyed Wave
subplot(3,1,3) %For Plotting The Amplitude Shift Keyed Wave
plot(t,w)
xlabel('Time')
ylabel('Amplitude')
title('Amplitide Shift Keyed Sign

INPUTS GIVEN TO GENERATE ASK MODULATED WAVE:

Enter the freq of Sine Wave carrier:100
Enter the freq of Periodic Binary pulse (Message):10
Enter the amplitude (For Both Carrier & Binary Pulse Message):4

RESULT:
 EXPERIMENT -2

Frequency Shift Keying (FSK) Digital Modulation MATLAB

Simulation, With MATLAB Code
Defining FSK (Frequency Shift Keying):
Frequency-shift keying (FSK) is a frequency modulation scheme in which digital information is
transmitted through carrier frequency. The simplest FSK is binary FSK (BFSK). BFSK uses a pair
of discrete frequencies to transmit binary (0s and 1s) information. If the number of symbols to transmit is
more than 2, so we will just have to use as many carriers as the number of symbols.

The given example of MATLAB Simulation of Frequency Shift Keying (FSK), the user is asked about the
frequency of the two carrier wavea, Message periodic pulse & the Amplitude of the waves (considering for
square message wave & both carrier wave have equal amplitude). The frequency of the carrier wave will
change from f1 to f2 or vice versa whenever a zero is changed to 1 or vice-verso. The frequency will not
change if in 2 successive time period there is no change in message bit value.
The MATLAB below code lets the user to plot 4 graphs, namely 2 graph plot of The Carrier Waves
(Sinusoid), The Binary Message Pulse & The Modulated Wave.

ATLAB Code FOR FSK (Frequency Shift Keying) BFSK in this case:

clc %for clearing the command window

close all %for closing all the window except command window
clear all %for deleting all the variables from the memory
fc1=input('Enter the freq of 1st Sine Wave carrier:');
fc2=input('Enter the freq of 2nd Sine Wave carrier:');
fp=input('Enter the freq of Periodic Binary pulse (Message):');
amp=input('Enter the amplitude (For Both Carrier & Binary Pulse Message):');
amp=amp/2;
t=0:0.001:1; % For setting the sampling interval
c1=amp.*sin(2*pi*fc1*t);% For Generating 1st Carrier Sine wave
c2=amp.*sin(2*pi*fc2*t);% For Generating 2nd Carrier Sine wave
subplot(4,1,1); %For Plotting The Carrier wave
plot(t,c1)
xlabel('Time')
ylabel('Amplitude')
title('Carrier 1 Wave')
subplot(4,1,2) %For Plotting The Carrier wave
plot(t,c2)
xlabel('Time')
ylabel('Amplitude')
title('Carrier 2 Wave')
m=amp.*square(2*pi*fp*t)+amp;%For Generating Square wave message
subplot(4,1,3) %For Plotting The Square Binary Pulse (Message)
plot(t,m)
xlabel('Time')
ylabel('Amplitude')
title('Binary Message Pulses')
for i=0:1000 %here we are generating the modulated wave
if m(i+1)==0
mm(i+1)=c2(i+1);
else
 mm(i+1)=c1(i+1);
end
end
subplot(4,1,4) %For Plotting The Modulated wave
plot(t,mm)
xlabel('Time')
ylabel('Amplitude')
title('Modulated Wave')

The following INPUTS GIVEN TO GENERATE FSK MODULATED WAVE:

Enter the freq of 1st Sine Wave carrier:10

Enter the freq of 2nd Sine Wave carrier:30
Enter the freq of Periodic Binary pulse (Message):5
Enter the amplitude (For Both Carrier & Binary Pulse Message):4

Result:
 EXPERIMENT -3
Binary Phase Shift Keying (BPSK) Modulation MATLAB
Simulation, With MATLAB Code
So what is PSK (Phase Shift Keying)?
Phase-shift keying (PSK) is a digital modulation technique that projects data by modulating, the phase
of a reference signal (the carrier wave).
Any 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.
In the example MATLAB Simulation of Phase Shift Keying (PSK), the user is asked about the frequency
of the carrier wave, Message periodic pulse & the Amplitude of the waves (considering both square
message wave & carrier wave have equal amplitude). The phase of the carrier wave will change by 180
degree whenever a zero is changed to 1 or vice-verso. The phase will not change if in 2 successive time
period there is no change in message bit value.
The MATLAB code lets the user to plot 3 graphs, namely of The Carrier Wave (Sinusoid), The Binary
Message Pulse & The Phase Shift Keyed Wave.

lc %for clearing the command window

close all %for closing all the window except command window
clear all %for deleting all the variables from the memory
t=0:.001:1; % For setting the sampling interval
fc=input('Enter frequency of Carrier Sine wave: ');
fm=input('Enter Message frequency : ');
amp=input('Enter Carrier & Message Amplitude(Assuming Both Equal):');
c=amp.*sin(2*pi*fc*t);% Generating Carrier Sine
subplot(3,1,1) %For Plotting The Carrier wave
plot(t,c)
xlabel('Time')

ylabel('Amplitude')
title('Carrier')
m=square(2*pi*fm*t);% For Plotting Message signal
subplot(3,1,2)
plot(t,m)
xlabel('time')
ylabel('ampmplitude')
title('Message Signal')% Sine wave multiplied with square wave in order to
generate PSK
x=c.*m;
subplot(3,1,3) % For Plotting PSK (Phase Shift Keyed) signal
plot(t,x)
xlabel('t')
ylabel('y')
title('PSK')

INPUTS GIVEN TO GENERATE ASK MODULATED WAVE:

Enter frequency of Carrier Sine wave: 60
Enter Message frequency : 10
Enter The Carrier & Message Amplitude(Assuming Both Equal): 3
 EXPERIMENT -4

TO STUDY SAMPLING THEOREM

Sampling is defined as, “The process of measuring the instantaneous

values of continuous-time signal in a discrete form.”

Sample is a piece of data taken from the whole data which is continuous in
the time domain.

When a source generates an analog signal and if that has to be digitized,

having 1s and 0s i.e., High or Low, the signal has to be discretized in time.
This discretization of analog signal is called as Sampling.

The following figure indicates a continuous-time signal x (t) and a sampled

signal xs (t). When x (t) is multiplied by a periodic impulse train, the
sampled signal xs (t) is obtained.
Sampling Rate
To discretize the signals, the gap between the samples should be fixed.
That gap can be termed as a sampling period Ts.

SamplingFrequency=1Ts=fsSamplingFrequency=1Ts=fs

Where,

 TsTs is the sampling time

 fsfs is the sampling frequency or the sampling rate
Sampling frequency is the reciprocal of the sampling period. This
sampling frequency, can be simply called as Sampling rate. The sampling
rate denotes the number of samples taken per second, or for a finite set of
values.

For an analog signal to be reconstructed from the digitized signal, the

sampling rate should be highly considered. The rate of sampling should be
such that the data in the message signal should neither be lost nor it should
get over-lapped. Hence, a rate was fixed for this, called as Nyquist rate.

Nyquist Rate
Suppose that a signal is band-limited with no frequency components higher
than W Hertz. That means, W is the highest frequency. For such a signal,
for effective reproduction of the original signal, the sampling rate should be
twice the highest frequency.

Which means,

fS=2WfS=2W

Where,

 fSfS is the sampling rate

 W is the highest frequency
This rate of sampling is called as Nyquist rate.

A theorem called, Sampling Theorem, was stated on the theory of this

Nyquist rate.

Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers
the theory of sufficient sample rate in terms of bandwidth for the class of
functions that are bandlimited.

The sampling theorem states that, “a signal can be exactly reproduced if it

is sampled at the rate fs which is greater than twice the maximum
frequency W.”

To understand this sampling theorem, let us consider a band-limited signal,

i.e., a signal whose value is non-zero between some –W and W Hertz.

Such a signal is represented as x(f)=0for|f|>Wx(f)=0for|f|>W

For the continuous-time signal x (t), the band-limited signal in frequency
domain, can be represented as shown in the following figure.
We need a sampling frequency, a frequency at which there should be no
loss of information, even after sampling. For this, we have the Nyquist rate
that the sampling frequency should be two times the maximum frequency.
It is the critical rate of sampling.

If the signal x(t) is sampled above the Nyquist rate, the original signal can
be recovered, and if it is sampled below the Nyquist rate, the signal cannot
be recovered.

The following figure explains a signal, if sampled at a higher rate

than 2w in the frequency domain.
The above figure shows the Fourier transform of a signal xs(t)xs(t). Here,
the information is reproduced without any loss. There is no mixing up and
hence recovery is possible.
The Fourier Transform of the signal xs(t)xs(t) is
Xs(w)=1Ts∑n=−∞∞X(w−nw0)Xs(w)=1Ts∑n=−∞∞X(w−nw0)

Where TsTs = Sampling Period and w0=2πTsw0=2πTs

Let us see what happens if the sampling rate is equal to twice the highest
frequency (2W)

That means,

fs=2Wfs=2W

Where,

 fsfs is the sampling frequency

 W is the highest frequency
The result will be as shown in the above figure. The information is replaced
without any loss. Hence, this is also a good sampling rate.

Now, let us look at the condition,

fs<2Wfs<2W

The resultant pattern will look like the following figure.

We can observe from the above pattern that the over-lapping of information
is done, which leads to mixing up and loss of information. This unwanted
phenomenon of over-lapping is called as Aliasing.

Aliasing
Aliasing can be referred to as “the phenomenon of a high-frequency
component in the spectrum of a signal, taking on the identity of a low-
frequency component in the spectrum of its sampled version.”

The corrective measures taken to reduce the effect of Aliasing are −

 In the transmitter section of PCM, a low pass anti-aliasing filter is

employed, before the sampler, to eliminate the high frequency
components, which are unwanted.

 The signal which is sampled after filtering, is sampled at a rate slightly

higher than the Nyquist rate.

This choice of having the sampling rate higher than Nyquist rate, also helps
in the easier design of the reconstruction filter at the receiver.

Scope of Fourier Transform

It is generally observed that, we seek the help of Fourier series and Fourier
transforms in analyzing the signals and also in proving theorems. It is
because −

 The Fourier Transform is the extension of Fourier series for non-periodic

signals.

 Fourier transform is a powerful mathematical tool which helps to view the

signals in different domains and helps to analyze the signals easily.

 Any signal can be decomposed in terms of sum of sines and cosines using
this Fourier transform.
 EXPERIMENT -5

 MATLAB CODE FOR QPSK MODULATION AND DEMODULATION
 clc;
 clear all;
 close all;
 data=[0 1 0 1 1 1 0 0 1 1]; % information

 %Number_of_bit=1024;
 %data=randint(Number_of_bit,1);

 figure(1)
 stem(data, 'linewidth',3), grid on;
 title(' Information before Transmiting ');
 axis([ 0 11 0 1.5]);

 data_NZR=2*data-1; % Data Represented at NZR form for QPSK modulation
 s_p_data=reshape(data_NZR,2,length(data)/2); % S/P convertion of data


 br=10.^6; %Let us transmission bit rate 1000000
 f=br; % minimum carrier frequency
 T=1/br; % bit duration
 t=T/99:T/99:T; % Time vector for one bit information



 % XXXXXXXXXXXXXXXXXXXXXXX QPSK modulatio
XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
 y=[];
 y_in=[];
 y_qd=[];
 for(i=1:length(data)/2)
 y1=s_p_data(1,i)*cos(2*pi*f*t); % inphase component
 y2=s_p_data(2,i)*sin(2*pi*f*t) ;% Quadrature component
 y_in=[y_in y1]; % inphase signal vector
 y_qd=[y_qd y2]; %quadrature signal vector
 y=[y y1+y2]; % modulated signal vector
 end
 Tx_sig=y; % transmitting signal after modulation
 tt=T/99:T/99:(T*length(data))/2;

 figure(2)

 subplot(3,1,1);
 plot(tt,y_in,'linewidth',3), grid on;
 title(' wave form for inphase component in QPSK modulation ');
 xlabel('time(sec)');
 ylabel(' amplitude(volt0');

 subplot(3,1,2);
 plot(tt,y_qd,'linewidth',3), grid on;
 title(' wave form for Quadrature component in QPSK modulation ');
 xlabel('time(sec)');
 ylabel(' amplitude(volt0');


 subplot(3,1,3);
 plot(tt,Tx_sig,'r','linewidth',3), grid on;
 title('QPSK modulated signal (sum of inphase and Quadrature phase
signal)');
 xlabel('time(sec)');
 ylabel(' amplitude(volt0');




 % XXXXXXXXXXXXXXXXXXXXXXXXXXXX QPSK demodulation
XXXXXXXXXXXXXXXXXXXXXXXXXX
 Rx_data=[];
 Rx_sig=Tx_sig; % Received signal
 for(i=1:1:length(data)/2)

 %%XXXXXX inphase coherent dector XXXXXXX
 Z_in=Rx_sig((i-1)*length(t)+1:i*length(t)).*cos(2*pi*f*t);
 % above line indicat multiplication of received & inphase carred
signal

 Z_in_intg=(trapz(t,Z_in))*(2/T);% integration using trapizodial
rull
 if(Z_in_intg>0) % Decession Maker
 Rx_in_data=1;
 else
 Rx_in_data=0;
 end

 %%XXXXXX Quadrature coherent dector XXXXXX
 Z_qd=Rx_sig((i-1)*length(t)+1:i*length(t)).*sin(2*pi*f*t);
 %above line indicat multiplication ofreceived & Quadphase carred
signal

 Z_qd_intg=(trapz(t,Z_qd))*(2/T);%integration using trapizodial rull
 if (Z_qd_intg>0)% Decession Maker
 Rx_qd_data=1;
 else
 Rx_qd_data=0;
 end


 Rx_data=[Rx_data Rx_in_data Rx_qd_data]; % Received Data
vector
 end


 figure(3)
 stem(Rx_data,'linewidth',3)
 title('Information after Receiveing ');
 axis([ 0 11 0 1.5]), grid on;

 EXPERIMENT -6
Quadrature Amplitude Modulation – QAM with Matlab Code
First, we will explain what QAM modulation is. Quadrature amplitude modulation is a
method that combines two amplitude modulated signals into a single channel. QAM is wide
used with pulse amplitude modulation (PAM) in wireless applications.
In this Matlab tutorial, we will simulate 16-QAM modulation using Matlab, and then we will
compare theoretical and simulated BER.

M = 16; % M determine the size of signal constellation

k = log2(M); % k determine the number of bits per symbol

n = 3e4; % n determine the number of bits to process

nSyms = n/k; % nSyms represent the number of symbols

hMod = modem.qammod(M); % We will create a 16-QAM modulator

hMod.InputType = 'Bit'; % Then accept bits as inputs

hMod.SymbolOrder = 'Gray'; % accept bits as inputs

hDemod = modem.qamdemod(hMod); %

x = randi([0 1],n,1); % Random binary data stream

tx = modulate(hMod,x);

EbNo = 0:10; % In dB

SNR = EbNo + 10*log10(k);

rx = zeros(nSyms,length(SNR));

bit_error_rate = zeros(length(SNR),1);

for i=1:length(SNR)

rx(:,i) = awgn(tx,SNR(i),'measured');

end
rx_demod = demodulate(hDemod,rx);

for i=1:length(SNR)

[~,bit_error_rate(i)] = biterr(x,rx_demod(:,i));

end

theoryBer = 3/(2*k)*erfc(sqrt(0.1*k*(10.^(EbNo/10))));

figure;

semilogy(EbNo,theoryBer,'-',EbNo, bit_error_rate, '^-');

grid on;

legend('Theory BER', 'simulation');

xlabel('Eb/No, dB');

ylabel('Bit Error Rate');

title('Bit error probability curve for 16-QAM modulation');

 EXPERIMENT -7

Pulse Code Modulation And Demodulation

Pulse code modulation is a method that is used to convert an analog signal into a
digital signal, so that modified analog signal can be transmitted through the
digital communication network. PCM is in binary form ,so there will be only
two possible states high and low(0 and 1). We can also get back our analog
signal by demodulation. The Pulse Code Modulation process is done in three
steps Sampling, Quantization, and Coding. There are two specific types of
pulse code modulations such as differential pulse code modulation(DPCM)
and adaptive differential pulse code modulation(ADPCM)
In sampling we are using PAM sampler that is Pulse Amplitude Modulation
Sampler which converts continuous amplitude signal into Discrete-time-
continuous signal (PAM pulses).Basic block diagram of PCM is given below
for better understanding.

What is a Pulse Code Modulation?

To get a pulse code modulated waveform from an analog waveform at the
transmitter end (source) of a communications circuit, the amplitude of the
analog signal samples at regular time intervals. The sampling rate or number
of samples per second is several times the maximum frequency. The
message signal converted into binary form will be usually in the number of
levels which is always to a power of 2. This process is called quantization.
Basic Elements of PCM System

At the receiver end, a pulse code demodulator decodes the binary signal back
into pulses with same quantum levels as those in the modulator. By further
processes we can restore the original analog waveform.

Pulse Code Modulation Theory

This above block diagram describes the whole process of PCM. The source of
continuous time message signal is passed through a low pass filter and then
sampling, Quantization, Encoding will be done. We will see in detail step by
step.
Sampling
Sampling is a process of measuring the amplitude of a continuous-time signal
at discrete instants, converts the continuous signal into a discrete signal. For
example, conversion of a sound wave to a sequence of samples. The Sample
is a value or set of values at a point in time or it can be spaced. Sampler
extract samples of a continuous signal, it is a subsystem ideal sampler
produces samples which are equivalent to the instantaneous value of the
continuous signal at the specified various points. The Sampling process
generates flat- top Pulse Amplitude Modulated (PAM) signal.

Analog and Sampled Signal

Sampling frequency, Fs is the number of average samples per second also

known as Sampling rate. According to the Nyquist Theorem sampling rate
should be at least 2 times the upper cutoff frequency. Sampling frequency,
Fs>=2*fmax to avoid Aliasing Effect. If the sampling frequency is very higher
than the Nyquist rate it become Oversampling, theoretically a bandwidth
limited signal can be reconstructed if sampled at above the Nyquist rate. If the
sampling frequency is less than the Nyquist rate it will become
Undersampling.

Basically two types of techniques are used for the sampling process. Those
are 1. Natural Sampling and 2. Flat- top Sampling.
Quantization
In quantization, an analog sample with an amplitude that converted into a
digital sample with an amplitude that takes one of a specific defined set of
quantization values. Quantization is done by dividing the range of possible
values of the analog samples into some different levels, and assigning the
center value of each level to any sample in quantization interval. Quantization
approximates the analog sample values with the nearest quantization values.
So almost all the quantized samples will differ from the original samples by a
small amount. That amount is called as quantization error. The result of this
quantization error is we will hear hissing noise when play a random signal.
Converting analog samples into binary numbers that is 0 and 1.

In most of the cases we will use uniform quantizers. Uniform quantization is

applicable when the sample values are in a finite range (Fmin, Fmax). The
total data range is divided into 2n levels, let it be L intervals. They will have an
equal length Q. Q is known as Quantization interval or quantization step size.
In uniform quantization there will be no quantization error.
 Uniformly Quantized Signal

As we know,
L=2n, then Step size Q = (Fmax – Fmin) / L

Interval i is mapped to the middle value. We will store or send only index value
of quantized value.

An Index value of quantized value Qi (F) = [F – Fmin / Q]

Quantized value Q (F) = Qi (F) Q + Q / 2 + Fmin

But there are some problems raised in uniform quantization those are

 Only optimal for uniformly distributed signal.

 Real audio signals are more concentrated near zeros.
 The Human ear is more sensitive to quantization errors at small values.
The solution for this problem is using Non- uniform quantization. In this
Process quantization interval is smaller near zero.
Coding
The encoder encodes the quantized samples. Each quantized sample is
encoded into an 8-bit code word by using A-law in the encoding process.
 Bit 1 is the most significant bit (MSB), it represents the polarity of the sample. “1”
represents positive polarity and “0” represents negative polarity.
 Bit 2,3 and 4 will defines the location of sample value. These three bits together
form linear curve for low level negative or positive samples.
 Bit 5,6,7 and 8 are the least significant bits (LSB) it represents one of the segments
quantized value. Each segment is divided into 16 quantum levels.
PCM is two types Differential Pulse Code Modulation (DPCM) and Adaptive
Differential Pulse Code Modulation (ADPCM).

In DPCM only the difference between a sample and the previous value is
encoded. The difference will be much smaller than the total sample value so
we need some bits for getting same accuracy as in ordinary PCM. So that the
required bit rate will also reduce. For example, in 5 bit code 1 bit is for polarity
and the remaining 4 bits for 16 quantum levels.

ADPCM is achieved by adapting the quantizing levels to analog signal

characteristics. We can estimate the values with preceding sample values.
Error estimation is done as same as in DPCM. In 32Kbps ADPCM method
difference between predicted value and sample value is coded with 4 bits, so
that we’ll get 15 quantum levels. In this method data rate is half of the
conventional PCM.

Pulse Code Demodulation

Pulse Code Demodulation will be doing the same modulation process in
reverse. Demodulation starts with decoding process, during transmission the
PCM signal will effected by the noise interference. So, before the PCM signal
sends into the PCM demodulator, we have to recover the signal into the
original level for that we are using a comparator. The PCM signal is a series
pulse wave signal, but for demodulation we need wave to be parallel.
By using a serial to parallel converter the series pulse wave signal will be
converted into a parallel digital signal. After that the signal will pass through n-
bits decoder, it should be a Digital to Analog converter. Decoder recovers the
original quantization values of the digital signal. This quantization value also
includes a lot of high frequency harmonics with original audio signals. For
avoiding unnecessary signals we utilize a low-pass filter at the final part.

Pulse Code Modulation Advantages

 Analog signal can be transmitted over a high- speed digital communication system.
 Probability of occurring error will reduce by the use of appropriate coding
methods.
 PCM is used in Telkom system, digital audio recording, digitized video special
effects, digital video, voice mail.
 PCM is also used in Radio control units as transmitter and also receiver for remote
controlled cars, boats, planes.
 The PCM signal is more resistant to interference than normal signal.
 EXPERIMENT -8
Differential Phase Shift Keying
In Differential Phase Shift Keying (DPSK) the phase of the modulated
signal is shifted relative to the previous signal element. No reference signal
is considered here. The signal phase follows the high or low state of the
previous element. This DPSK technique doesn’t need a reference oscillator.

The following figure represents the model waveform of DPSK.

It is seen from the above figure that, if the data bit is Low i.e., 0, then the
phase of the signal is not reversed, but continued as it was. If the data is a
High i.e., 1, then the phase of the signal is reversed, as with NRZI, invert
on 1 (a form of differential encoding).

If we observe the above waveform, we can say that the High state
represents an M in the modulating signal and the Low state represents
a W in the modulating signal.

DPSK Modulator
DPSK is a technique of BPSK, in which there is no reference phase signal.
Here, the transmitted signal itself can be used as a reference signal.
Following is the diagram of DPSK Modulator.
DPSK encodes two distinct signals, i.e., the carrier and the modulating
signal with 180° phase shift each. The serial data input is given to the
XNOR gate and the output is again fed back to the other input through 1-
bit delay. The output of the XNOR gate along with the carrier signal is
given to the balance modulator, to produce the DPSK modulated signa

DPSK Demodulator
In DPSK demodulator, the phase of the reversed bit is compared with the
phase of the previous bit. Following is the block diagram of DPSK
demodulator.

From the above figure, it is evident that the balance modulator is given
the DPSK signal along with 1-bit delay input. That signal is made to
confine to lower frequencies with the help of LPF. Then it is passed to a
shaper circuit, which is a comparator or a Schmitt trigger circuit, to
recover the original binary data as the output.