PracticalNo.1 Aim:Tostudy Amplitude Modulation (AM) and its mathematical Practical No.

Practical No. 2 Aim: To study the generation and detection of Double

representation. Sideband Suppressed
Software Used: SCILAB Theory::
Carrier (DSB-SC) modulation using SCILAB.
Amplitude Modulation (AM) is a modulation technique in which the amplitude of a
high-frequency carrier signal is varied in accordance with the instantaneous
Software Used: SCILAB
amplitude of the message signal, while frequency and phase remain constant. Theory: Double Sideband Suppressed Carrier (DSB-SC) is a type of
Let:• Carrier signal: c(t) = Ac cos(2πfct)• Message signal: m(t) = Am cos(2πfmt) amplitude modulation in which the carrier signal is not transmitted. Only
Where:• Ac, Am → amplitudes • fc, fm → frequencies the information-carrying parts of the signal, called sidebands, are
Mathematical Representation: s(t) = Ac[1 + m cos(2πfmt)] cos(2πfct) transmitted. In normal AM, the transmitted signal contains:
Where: m = Am / Ac → modulation index
• Carrier • Upper sideband (USB) Lower sideband (LSB) But in
Expanding: s(t) = Ac cos(2πfct) + (mAc/2) cos 2π(fc + fm)t + (mAc/2) cos 2π(fc - fm)t
Thus, AM contains: • Carrier • Upper Sideband (USB) • Lower Sideband (LSB)
DSB-SC:
Modulation Index: m = Am / Ac • m < 1 → Under modulation • The carrier is removed (suppressed) • Only USB and LSB are
• m = 1 → Perfect modulation • m > 1 → Over modulation (distortion) Carrier at fc transmitted
• Sidebands at: fc + fm (USB) and fc - fm (LSB) • This makes the system more power efficient Mathematical
Bandwidth: BW = 2fm Representation:
Program: // Amplitude Modulation (AM) clc;
Message signal: m(t) = Am cos(2πfmt) Carrier signal: c(t) = Ac
clear; close;
// Parameters Ac = 5; Am = 2; fc = 1000; fm = 100;
cos(2πfct)
t = 0:0.00001:0.01; // Message signal message = Am * cos(2 * %pi * fm * t); // DSB-SC signal: s(t) = m(t) × c(t) = Am Ac cos(2πfmt) cos(2πfct) Using
Carrier signal carrier = Ac * cos(2 * %pi * fc * t); // Modulation index m = Am / Ac; identity: cos A cos B = ½[cos(A+B) + cos(A-B)] ,,s(t) = (Am Ac / 2)[cos
// AM signal AM = Ac * (1 + m * cos(2 * %pi * fm * t)) .* cos(2 * %pi * fc * t); 2π(fc+fm)t + cos 2π(fc-fm)t]
// Plotting subplot(3,1,1); plot(t, message); This shows the signal has two frequencies: (fc + fm) and (fc - fm). These
title("Message Signal") subplot(3,1,2); plot(t, carrier); title("Carrier Signal");
are called sidebands. The carrier frequency fc is absent. Bandwidth:
subplot(3,1,3); plot(t, AM); title("Amplitude Modulated Signal");
Bandwidth of DSB-SC = 2fm (Same as standard AM, but without carrier)
Demodulation (Detection): To recover the original message signal,
multiply the received DSB-SC signal with a carrier signal of same
frequency and phase, then pass through a Low Pass Filter (LPF). This
method is called Coherent Detection.
Conclusion: Program: clc; clear; close; // Parameters fs = 1000; t = 0:1/fs:1; Am
Amplitude Modulation (AM) is a simple and widely used modulation = 1;
technique in which the Ac = 1; fm = 5; fc = 50; // Message Signal m = Am * cos(2 * %pi * fm *
amplitude of the carrier varies according to the message signal. The t);
experiment verifies that the AM // Carrier Signal c = Ac * cos(2 * %pi * fc * t); // DSB-SC Modulated
signal consists of a carrier and two sidebands, and the envelope of the Signal
waveform represents the s = m .* c; // Demodulation (Coherent Detection) demod = s .* c;
message signal. SCILAB simulation helps in clear visualization and
understanding of the modulation
process.

Practical No. 3
Aim: To analyze the modulation index and power relations of an AM
signal using SCILAB. Software Used: SCILAB
Theory: Amplitude Modulation (AM) is a technique in which the
amplitude of a high-frequency carrier signal is varied in proportion to the
instantaneous amplitude of the message signal, while the frequency and
phase of the carrier remain constant. Mathematical Representation:
The AM signal can be expressed as: s(t) = (Ac + Am sin(2πfmt))
sin(2πfct) Where: Ac = Carrier amplitude, Am = Message amplitude, fc
= Carrier frequency, fm = Message frequency Modulation Index (m):
m = Am / Ac • m < 1 → Undermodulation (signal is clear but weaker)
• m = 1 → 100% modulation (maximum without distortion)
• m > 1 → Overmodulation (causes distortion in the envelope)
It can also be determined from the envelope: m = (Emax - Emin) / (Emax
+ Emin) Frequency Components of AM: • Carrier at frequency fc •
Upper Sideband (USB) at fc + fm • Lower Sideband (LSB) at fc - fm
Power Relations in AM: • Carrier Power: Pc = Ac² / 2 • Total Power: Pt
= Pc(1 + m²/2) • Sideband Power: Ps = Pt - Pc = Pc × m²/2
Efficiency of AM: Transmission efficiency: η = Ps / Pt = m² / (2 + m²)
Maximum efficiency ≈ 33.33% at m = 1
Program: clc; clear; close; // Input parameters Ac = 10; // Carrier
amplitude Am = 5; // Message amplitude fc = 1000; // Carrier frequency
(Hz) fm = 100; // Message frequency (Hz) t = 0:0.00001:0.01; // Time
vector // Modulation index m = Am / Ac; disp("Modulation Index (m) =
"); disp(m); // Message signal message = Am * sin(2 * %pi * fm * t);
// Carrier signal carrier = Ac * sin(2 * %pi * fc * t);

Output:
Practical No. 4 Practical No. 5
Aim: To analyze Narrowband and Wideband FM and their differences. Aim: To study the Sampling Theorem and different sampling techniques
Software Used: SCILAB using
Theory: Frequency Modulation (FM) is a method of encoding SCILAB. Software Used: SCILAB Theory: Sampling is the process of
information on a carrier wave by varying its instantaneous frequency. converting a continuous-time signal into a discrete-time signal by taking
Depending on the modulation index (β), FM is categorized into two its values at uniform time intervals. According to the Nyquist-Shannon
types: Narrowband FM (NBFM) and Wideband FM (WBFM). Sampling Theorem, a band-
Mathematical Foundation: The general expression for an FM signal is: limited signal can be completely reconstructed from its samples if it is
s(t) = Ac cos[2πfct + β sin(2πfmt)] The modulation index β is defined as: sampled at a frequency greater than or equal to twice its highest
β = Δf / fm frequency component.
Where Δf = frequency deviation and fm = modulating frequency Nyquist Rate: Fs ≥ 2Fm Where Fs = Sampling frequency, Fm =
Narrowband FM (NBFM): Maximum frequency of message signal Sampling Techniques: 1. Ideal
Occurs when β < 1. In NBFM, the frequency deviation is very small. Sampling: A theoretical method using an impulse train with infinitely
• Bandwidth: Approximately 2fm (similar to AM) • Spectrum: Carrier + narrow pulses. Because pulses have zero width and infinite amplitude, it
only two sidebands (USB and LSB) • Applications: Police wireless, cannot be physically implemented. 2. Natural Sampling: A practical
ambulances, walkie-talkies method where the signal is multiplied by rectangular pulses of finite
Wideband FM (WBFM): Occurs when β > 1. In WBFM, the frequency width. The top of
deviation is large. each pulse follows the shape of the original signal. 3. Flat-Top
• Bandwidth: BW ≈ 2(Δf + fm) = 2fm(1 + β) [Carson's Rule] Sampling:
• Spectrum: Carrier + infinite sidebands (amplitudes by Bessel functions) A practical technique where the amplitude of each sample is held
• Applications: FM radio (88-108 MHz), TV sound transmission constant for the pulse duration. This introduces slight distortion known
Program: clear; clc; close; // Parameters fs = 10000; // Sampling as the aperture effect.
frequency t = 0:1/fs:0.1; // Time vector fc = 500; // Carrier frequency fm Program: clc; clear; close; // Input parameters fm = 2; // Message
= 50; // Modulating frequency signal frequency fs = 20; // Sampling frequency (Fs > 2Fm) t =
Ac = 1; // Carrier amplitude Am = 1; // Modulating amplitude 0:0.001:1; // Continuous time
// Modulation Indices beta_nb = 0.2; // Small beta for NBFM beta_wb = ts = 0:1/fs:1; // Sampling time // Message Signal x = sin(2 * %pi * fm * t);
5.0; // Large beta for WBFM // Generate Signals mt = Am * xs = sin(2 * %pi * fm * ts); // Ideal Sampling ideal_sampled = zeros(t);
cos(2*%pi*fm*t); nb_fm = Ac * cos(2*%pi*fc*t + for i = 1:length(ts) [val, idx] = min(abs(t - ts(i))); ideal_sampled(idx)
beta_nb*sin(2*%pi*fm*t)); wb_fm = Ac * cos(2*%pi*fc*t + = xs(i);
beta_wb*sin(2*%pi*fm*t)); end // Natural Sampling pulse_train = zeros(t); pulse_width = 0.01;
// FFT N = length(t); freq = (0:N-1)*(fs/N); NB_fft = abs(fft(nb_fm)); for i = 1:length(ts) pulse_train(t >= ts(i) & t <= (ts(i) + pulse_width)) =
WB_fft = abs(fft(wb_fm)); // Plotting plot(freq(1:N/10), WB_fft(1:N/10)); 1;
title("WBFM Spectrum"); xlabel("Frequency (Hz)"); ylabel("Magnitude"); end natural_sampled = x .* pulse_train; // Flat-Top Sampling
xlabel("Time"); ylabel("Amplitude"); legend(["Original Signal"; "Flat Top
Sampled Signal"]);

Practical No. 6
Aim: To generate a Pulse Amplitude Modulated (PAM) signal using
SCILAB and observe its waveform characteristics in the time domain.
Software Used: SCILAB Theory:
Pulse Amplitude Modulation (PAM) is an analog pulse modulation
technique where the amplitude of the carrier pulses is varied in
accordance with the instantaneous amplitude of the message signal.
Mathematical Expression: If m(t) = message signal and p(t) = periodic
pulse train, then: s(t) = m(t) × p(t)
Where p(t) is a rectangular pulse train. The multiplication effectively
samples the message at
intervals defined by the pulse train.
Program: clf(); fc = 100; // Carrier pulse frequency
fm = 10; // Message signal frequency
fs = 1000; // Sampling frequency t = 0:1/fs:0.2;
// Step 1: Message Signal m = sin(2 * %pi * fm * t);
// Step 2: Carrier Pulse Train c = 0.5 * (1 + squarewave(2 * %pi * fc * t,
20)); // Step 3: PAM Signal
pam = m .* c;
// Step 4: Plotting
subplot(3,1,1);
plot(t, m);
xtitle('Message Signal', 'Time (s)', 'Amplitude');
subplot(3,1,2);
plot(t, c);
xtitle('Carrier Pulse Train', 'Time (s)', 'Amplitude');
subplot(3,1,3);
plot(t, pam);
xtitle('PAM Signal', 'Time (s)', 'Amplitude');
Practical No. 7 Practical No. 8 Aim: To analyze and implement Quantization and Delta
Aim: To study and implement Pulse Code Modulation (PCM) and to Modulation techniques. Software Used: SCILAB
observe the processes of sampling, quantization, and encoding of an Theory: 1. Quantization:
Analog signal into a Digital signal. Quantization is the process of converting a continuous-range analog
Software Used: SCILAB signal into discrete levels. It is a
Theory: lossy process. The difference between actual and quantized value is
Pulse Code Modulation (PCM) is a technique used to convert an analog Quantization Error. More bits
signal into a digital signal. mean more levels (L = 2^n) and higher resolution.
PCM represents the continuous-time message signal as a sequence of 2. Delta Modulation (DM): DM is a 1-bit ADC technique that
binary coded pulses. The approximates a signal by comparing current sample with previous
binary form permits only 2 states: 0 and 1. one. It sends '1' if signal is increasing and '0' if decreasing.
Three Main Steps in PCM: Two Major Distortions in DM: • Slope Overload: Occurs when signal
1. Sampling: The analog signal is sampled at regular intervals. changes too fast for step size to keep up. • Granular Noise: Occurs
According to Nyquist theorem, fs ≥ 2fm to avoid when signal is flat, causing staircase to hunt around the signal.
aliasing. Program: clf(); clc;
2. Quantization: The process of approximating continuous amplitude // 1. Input Signal
values with a finite set of discrete levels. Each t = 0:0.01:1;
level represents a specific digital code. The number of levels determines f = 2; x = 2 * sin(2*%pi*f*t); // 2. Quantization
resolution (e.g., 8-bit = 256 bits = 3; L = 2^bits;
levels). vmax = 2; vmin = -2;
3. Encoding: Representing quantized samples using binary codes. Each step_size = (vmax - vmin) / (L-1);
quantized sample is converted to a binary xq = round(x/step_size) * step_size;
number. // 3. Delta Modulation
Program: // PCM Modulation in SCILAB clc; clear; ts = 0:0.05:1;
close; // Step 1: Generate Analog Signal xs = 2 * sin(2*%pi*f*ts);
t = 0:0.001:1; delta = 0.35; xdm(1) = 0;
fm = 5; for i = 1:length(xs)-1
x = sin(2*%pi*fm*t); if xs(i) > xdm(i)
// Step 2: Sampling xdm(i+1) = xdm(i) + delta;
fs = 20; else
ts = 0:1/fs:1; xdm(i+1) = xdm(i) - delta;
xs = sin(2*%pi*fm*ts); end
// Step 3: Quantization end
L = 8; // 4. Plotting
xmin = min(xs); subplot(2,1,1);
plot(t, x, 'g', t, xq, 'r--');

Practical No. 9
Aim: To study different digital modulation techniques such as BPSK,
QPSK and QAM.
Software Used: SCILAB
Theory: Digital modulation is a technique used to transmit digital data
over communication channels by
modifying a carrier signal. It improves noise immunity and bandwidth
efficiency.
1. Binary Phase Shift Keying (BPSK):
In BPSK, the phase of carrier is shifted by 180° depending on binary
input.
• Binary '0' → one phase
• Binary '1' → phase shifted by 180°
• Simple and robust, transmits 1 bit per symbol
2. Quadrature Phase Shift Keying (QPSK):
QPSK uses four phase shifts (0°, 90°, 180°, 270°).
• Each symbol represents 2 bits
• More efficient than BPSK
• Requires more complex receiver
3. Quadrature Amplitude Modulation (QAM):
QAM combines amplitude and phase variations.
• Transmits multiple bits per symbol (e.g., 16-QAM = 4 bits/symbol)
• High data rate
• More sensitive to noise
Program:
clc;
clear;
close;
// Generate random binary data
data = round(rand(1,10));
// Time axis Conclusion:
t = 0:0.01:1; Different digital modulation techniques were studied:
Conclusion: Quantization accuracy improves with increase in bits. In
fc = 5; • BPSK is simple but less efficient
Delta Modulation, step size (delta) and sampling frequency must be
Output: • QPSK improves bandwidth efficiency
carefully chosen to avoid Slope Overloadand Granular Noise. The
• BPSK: Shows phase reversal (180° shift) for binary changes. • QAM provides higher data rates but is more noise sensitive
experiment demonstrates how continuous signals are represented in
• QPSK: Shows four distinct phase shifts representing 2 bits per symbol.
digital format.
• QAM: Shows variations in both amplitude and phase.
Practical No. 10 Aim: To analyze the performance of various digital
modulation schemes using SCILAB. Software Used: SCILAB
Theory: Digital modulation is used to transmit binary data by varying a
carrier signal. The performance is measured using Bit Error Rate (BER),
which decreases as Signal-to-Noise Ratio (SNR) increases. BPSK: The
phase of carrier changes according to input bit. Binary '1' = positive, '0' =
negative. Simple with low error but lower data rate. QPSK: Four phases
represent data, allowing 2 bits per transmission. Higher data rate but
slightly higher error. Performance Metrics: • Higher SNR → Lower BER
• Lower SNR → Higher BER Program:
clc;
clear;
close;
// Number of bits
N = 1000;
// Generate random binary data
data = round(rand(1, N));
// BPSK Modulation
bpsk_tx = 2 * data - 1;
// QPSK Modulation
data_qpsk = matrix(data, 2, N/2);
qpsk_tx = data_qpsk(1, :) + %i * data_qpsk(2, :);
// SNR range
snr_db = 0:2:10;
ber_bpsk = zeros(snr_db);
ber_qpsk = zeros(snr_db);
for k = 1:length(snr_db)
snr = snr_db(k); Conclusion:
// BPSK • BPSK provides better error performance than QPSK.
noise = rand(1, N, "normal"); • QPSK offers higher data rate but slightly higher BER.
sig_pow = mean(bpsk_tx.^2); • Increasing SNR improves overall system performance.
noise_pow = sig_pow / (10^(snr/10));
noise = sqrt(noise_pow) * noise;
rx_bpsk = bpsk_tx + noise;
rx_data = rx_bpsk > 0;
errors = sum(data <> rx_data);