Name:Rahul Jhajhria

Prn:20230802399
Div:- C

Experiment No.-1

Title: Implementation of frequency response of First order RC Low Pass Filter

Apparatus: Name of Hardware/Software

Diagram: Passive RC LPF

Theory: Block / Circuit diagram explanation

1. ** Input Signal **: The input signal to the filter is typically represented by a voltage

source. This could be any signal that you want to filter, such as an audio waveform or a

sensor output.

2. ** Resistor (R) **: This component is represented by a straight line with a zigzag

symbol. The resistor limits the flow of current in the circuit. In the context of the low-

pass filter, it's essential for determining the cutoff frequency (\( f_c \)). A higher

resistance value will result in a lower cutoff frequency.

3. ** Capacitor (C) **: The capacitor is represented by two parallel lines, usually with

space between them. It stores and releases electrical energy. In this circuit, the capacitor

works in tandem with the resistor to filter out high-frequency components of the input

signal.

4. ** Output Signal **: The filtered output signal is typically taken from the junction

between the resistor and the capacitor. This is where the filtered signal is available for

further processing or use.

5. ** Ground Symbol **: Often denoted by a downward-pointing triangle, it represents the

reference point of the circuit, usually connected to the negative terminal of the power

supply. It's essential for completing the circuit and providing a reference voltage level.

In summary, the input signal passes through the resistor and capacitor in series. The

resistor limits the current, and the capacitor provides impedance that varies with

frequency. Together, they create a filter that attenuates higher frequencies, allowing only

Result and Conclusion:

MATLAB Simulink Model with all selected input /block parameters and Result screenshot:
(Your name should be reflected in the all screenshot)
Write your Name and PRN on first line of code in comment section

Observation Table:

…………
SR.NO FREQUENCY VOUT

1. 100 9.94

2. 200 9.87

3. 300 9.76

4. 400 9.74

5. 500 9.70

6. 600 9.65

7. 720 9.71

8. 750 9.07

9. 800 9.55

10. 900 9.51

11. 1000 9.45

12. 2000 9.04

13. 3000 8.16

14. 4000 7.30

15. 5000 6.70

16. 6000 6.04

17. 7000 5.36

18. 8000 4.93

19. 9000 4.47

20. 10000 4.12

Draw Frequency response from above observation table of Low Pass Filter with Passive
RC components:

Note: Upload single pdf file for all above details

References: https://www.thorlabs.com/newgrouppage9.cfm?objectgroup_id=9817

Experiment No.-2

Title: Write a MTALAB code for Amplitude Modulation and Demodulation

Apparatus: Hardware/Software

Diagram: Amplitude Modulation and demodulation

Modulation:
Demodulation:

Theory: Block / Circuit diagram explanation

According to the standard definition, “The amplitude of the carrier signal varies in
accordance with the instantaneous amplitude of the modulating signal.” Which
means, the amplitude of the carrier signal containing no information varies as per
the amplitude of the signal containing information, at each instant

Demodulation: The process of extracting an original message signal from the

modulated wave is known as detection or demodulation. The circuit, which
demodulates the modulated wave is known as the demodulator.
2. **Carrier Signal Generator**: This block generates the carrier signal, which is
usually a high-frequency sinusoidal waveform. The frequency of the carrier signal
determines the bandwidth and characteristics of the modulated signal.

3. **Modulation Circuit**: This is where the input signal is combined with the carrier
signal. The modulation technique used depends on the application. Common modulation
techniques include Amplitude Modulation (AM), Frequency Modulation (FM), and Phase
Modulation (PM).

4. **Output Signal**: The modulated signal, which contains both the carrier and the
modulated information, is produced at the output of the modulation circuit. This signal is
ready for transmission over the communication channel.

**Demodulator:**

A demodulator, also known as a detector, is the counterpart of a modulator. It extracts the

original information signal from the modulated carrier signal. The block diagram
typically includes the following components:

1. **Modulated Signal Input**: This is the signal received from the communication
channel, which contains both the carrier and the modulated information.

2. **Carrier Signal Regenerator**: In some cases, particularly in coherent detection,

the demodulator needs to recover the carrier signal from the modulated signal. This block
regenerates the carrier signal with the same frequency and phase as the original carrier used
during modulation.

3. **Demodulation Circuit**: This block processes the modulated signal to extract the
original information signal. The demodulation technique employed depends on the
modulation scheme used. For example, in AM demodulation, envelope detection
techniques are used to recover the original signal.

4. **Output Signal**: The recovered information signal is obtained at the output of the
demodulation circuit. This signal is an approximation of the original input signal, ready for
further processing or utilization.

These are the basic components you'll find in the block diagrams of modulators and
demodulators. The specific implementation and complexity can vary depending on
factors such as the modulation scheme, the communication medium, and the desired
performance characteristics.
Result and Conclusion:

Write a MATLAB code with all the value of m (Modulation Index) and observe the output
waveform. Paste the screenshot of output waveform with code. (Your name should be
reflected in the whole screenshot)

Note: Upload single pdf file for all above details

Experiment No.-3

Title: Write a MATLAB code for DSB-SC Modulation and Demodulation

Apparatus: Hardware/Software

Diagram: DSB-SC Modulation and demodulation

Modulation:

DeModulation:

Theory: Block / Circuit diagram explanation

1. ** Modulation :**

** Message Signal (m(t)) :** This is the original signal that we want to transmit. It could

be any analog signal, such as voice, music, or data.

** Carrier Signal (c(t)) :** This is a high-frequency signal generated by an oscillator. The

frequency of the carrier is much higher than the frequency of the message signal. In DSB-

SC modulation, the carrier is suppressed.

** Multiplier :** The message signal is multiplied by the carrier signal. This process

modulates the message onto the carrier, resulting in a DSB-SC signal.

**Output:** The output of the multiplier is a DSB-SC signal, which contains two
sidebands, each carrying a copy of the message signal, with the carrier frequency
removed.

2. **Demodulation:**

**Received DSB-SC Signal (r(t)):** This is the signal that has been transmitted over
the communication channel and received by the demodulator.

**Local Carrier Signal (c(t)):** A local carrier signal at the same frequency
as the carrier used in modulation is generated.

**Multiplier:** The received DSB-SC signal is multiplied by the local carrier

signal. This process effectively shifts the frequency components of the
received signal to baseband.

**Low-pass Filter:** Since DSB-SC contains only two sidebands, a low-pass

filter is used to filter out one of the sidebands. This leaves only the desired
message signal. **Output:** The output of the low-pass filter is the
demodulated message signal, which is an approximation of the original
message signal.

In summary, DSB-SC modulation involves suppressing the carrier signal and

transmitting two sidebands containing the message signal. Demodulation of
DSB-SC involves multiplying the received signal by a local carrier to shift the
frequency components back to baseband and then filtering out one of the
sidebands to extract the original message signal.

Result and Conclusion:

Implement a MATLAB Code and observe the output waveform. Paste the screenshot of
output waveform with Simulink Model. (Your name should be reflected in the whole
screenshot)

Note: Upload single pdf file for all above details

Experiment No.-4

Title: Write a MATLAB code for SSB-SC Modulation and Demodulation

Apparatus: Hardware/Software

Diagram: SSB-SC Modulation and demodulation

Modulation:
DeModulation:

Theory: Block / Circuit diagram explanation

** SSB-SC Modulation :**

1. ** Input Signal **: This is the baseband signal that carries the information to be

transmitted. It could be audio, video, or data.

2. ** Carrier Signal Generator **: A carrier signal at the desired frequency is generated.
However, in SSB-SC modulation, the carrier is suppressed, so it's not usually shown in

the block diagram.

3. ** Modulation Circuit **: In SSB-SC modulation, the input signal is modulated with a

carrier using a method such as frequency shifting or phasing. This process results in only

one sideband being transmitted, either the upper or lower sideband, while the
carrier and the other sideband are suppressed.

4. **Output Signal**: The modulated signal, containing only one sideband

and no carrier, is produced at the output of the modulation circuit. This signal
is ready for transmission over the communication channel.

**SSB-SC Demodulation:**

1. **Modulated Signal Input**: This is the received signal from the

communication channel, which contains only one sideband and no carrier.

2. **Carrier Signal Regeneration (Optional)**: In some cases, the

demodulator may require a regenerated carrier signal to aid in demodulation.
This block generates a carrier signal with the same frequency as the carrier used
during modulation.

3. **Demodulation Circuit**: The demodulation circuit extracts the original

baseband signal from the modulated signal. This process involves reversing the
modulation technique applied during modulation to recover the original
information signal.

4. **Output Signal**: The recovered baseband signal, which is an

approximation of the original input signal, is obtained at the output of the
demodulation circuit. This signal can then be further processed or utilized as
needed.

SSB-SC modulation and demodulation are commonly used in radio

communication, particularly in scenarios where bandwidth efficiency is crucial, as
it allows for the transmission of information using a smaller bandwidth compared
to traditional AM modulation

Result and Conclusion:

Write a MATLAB code for the generation of SSB using phase shift method, use the Hilbert
transform technique and observe the output waveform. Paste the screenshot of output
waveform with code. (Your name should be reflected in the whole screenshot)

Note: Upload single pdf file for all above details

Experiment No.-5

Title: Write a MATLAB code for Frequency Modulation and Demodulation

Apparatus: Hardware/Software

waveform: Frequency Modulation and demodulation

modulation:
THEORY :

Certainly! Here's an explanation of a basic circuit diagram for FM modulation and

demodulation:

FM Modulation Circuit:

1. **Input Signal Source (Modulating Signal)**: This is where the audio

signal or the signal containing the information to be transmitted is introduced
into the circuit. It could be a microphone for audio signals or any other source
for data signals.

2. **Voltage-Controlled Oscillator (VCO)**: The VCO generates the

carrier signal whose frequency will vary based on the amplitude of the input
signal. The output frequency of the VCO is directly proportional to the
amplitude of the modulating signal.
3. **Summing Amplifier (or Mixer)**: The modulating signal from the
input source and the carrier signal from the VCO are combined in the summing
amplifier. This results in frequency modulation of the carrier signal according
to the amplitude variations of the modulating signal.

4. **Output**: The output of the FM modulation circuit is the frequency-

modulated carrier signal, which carries the information encoded in the
variations of its frequency.

FM Demodulation Circuit:

1. **Input (FM Signal)**: The frequency-modulated signal received from

the transmitter or communication channel is fed into the demodulator circuit.

2. **Frequency Discriminator (or Frequency-to-Voltage Converter)**:

This component converts the frequency variations of the FM signal into
corresponding voltage variations. One common implementation of this is
through a frequency-to-voltage converter circuit.

3. **Low-Pass Filter (LPF)**: The output of the frequency discriminator

contains both the original modulating signal and high-frequency noise. The
low-pass filter is used to remove the high-frequency noise and extract the
original modulating signal.

4. **Output**: The output of the FM demodulation circuit is the

demodulated signal, which is the original modulating signal that was encoded
onto the carrier wave during modulation. This signal can then be amplified or
processed further for various applications.

Both the FM modulation and demodulation circuits are essential components of FM

communication systems, enabling the transmission and reception of information with

Result and conclusion:

Write a MATLAB code for the generation of Frequency Modulation and demodulation
observe the output waveform. Paste the screenshot of output waveform with code. (Your
name should be reflected in the whole screenshot)
Experiment No.-6

Title: Write a MATLAB code for PAM generation and Reconstruction

Apparatus: Hardware/Software
Theory: Block / Circuit diagram explanation

Pam:

1.Pulse amplitude modulation is the basic form of pulse modulation. In this

modulation, the signal is sampled at regular intervals and each sample is made
proportional to the amplitude of the modulating signal. Before we study in detail
PAM lets us know the concepts of modulation.

2. **Pulse Generator**: The digital input is fed into a pulse generator,

which converts each digital bit into a corresponding pulse. These pulses
typically have a constant width but varying amplitudes.

3. **Analog Modulator**: The pulse generator output is then fed into an

analog modulator. This component is responsible for converting the digital
pulse stream into an analog waveform. In PAM, this is done by varying the
amplitude of the pulses according to the digital input.

4. **Analog Output**: The output of the analog modulator is an analog

waveform. The amplitude of this waveform varies based on the amplitude of
the pulses generated by the modulator, which in turn is determined by the
digital input.

Here's a simplified summary of the process: the digital input is converted into
pulses, and these pulses are used to modulate the amplitude of an analog signal,
effectively encoding the digital information into an analog waveform. This
analog waveform can then be transmitted over a communication channel, such as
a wired or wireless medium.

Result and Conclusion:

Write a MATLAB code for the generation of Pulse Amplitude Modulation and observe the
output waveform. Paste the screenshot of output waveform with code. (Your name should be
reflected in the whole screenshot)

Experiment No.-7

Title: Write a MATLAB code for generation of Sampling Theorem

Apparatus: Hardware/Software

Theory: Block / Circuit diagram explanation

A continuous signal or an analog signal can be represented in the digital version in

the form of samples. Here, these samples are also called as discrete points. In
sampling theorem, the input signal is in an analog form of signal and the second
input signal is a sampling signal, which is a pulse train signal and each pulse is
equidistance with a period of “Ts”. This sampling signal frequency should be more
than twice of the input analog signal frequency.

2. **Discrete Sampling**: When we want to process or transmit

continuous signals usingdigital systems (computers, digital communication),
we need to convert them into discrete signals through a process called
sampling. Sampling involves measuring the amplitude of the continuous signal
at regular intervals of time. The result is a sequence of discrete values, or
samples, that represent the original continuous signal at specific points in time.

3. **Nyquist-Shannon Sampling Theorem**: This theorem states that in

order to accurately reconstruct a continuous signal from its samples, the
sampling frequency (the rate at which samples are taken) must be at least twice
the highest frequency present in the signal. Mathematically, this can be
expressed as:

\(f_s \geq 2f_{\text{max}}\)

Where:

- \(f_s\) is the sampling frequency (samples per second or Hertz).

- \(f_{\text{max}}\) is the highest frequency component present in the signal.

 This condition ensures that there is enough information in the samples to accurately
reconstruct the original continuous signal without aliasing or distortion.

4. **Aliasing**: If the sampling frequency is less than twice the highest

frequency component in the signal, aliasing occurs. Aliasing is a phenomenon
where highfrequency components in the original signal are incorrectly
represented as lower frequencies in the sampled signal. This leads to distortion
and loss of information during reconstruction.

5. **Applications**: The Sampling Theorem is crucial in various fields

such as telecommunications, audio processing, medical imaging, and many
others. It forms the basis for digital audio and video encoding, digital signal
processing algorithms, and the design of digital communication systems.

In summary, the Sampling Theorem establishes the minimum sampling rate

required to accurately capture and reconstruct a continuous signal into its discrete
representation, ensuring faithful reproduction of the original signal without
distortion.

RESULT AND CONCLUSION :

Result and Conclusion:

Write a MATLAB code for the generation of sampling in all different condition. observe the
output waveform. Paste the screenshot of output waveform with code. (Your name should be
reflected in the whole screenshot)
Note: Upload single pdf file for all above details

Experiment No.-8

Title: Write a MATLAB code for Generation of Pulse Width Modulation

Apparatus: Hardware/Software

Theory: Block / Circuit diagram explanation

1.Pulse-width modulation, commonly known as PWM, is a modulation method

that changes the pulse signal’s width in electrical systems to regulate the average
power supplied to a load. PWM is particularly helpful for effectively regulating
the output of audio amplifiers, the speed of motors, and the brightness of light.
PWM is an elegant way of referring to a particular kind of digital signal.
Applications for pulse width modulation are numerous and include complex
control circuit
2. **Duty Cycle**: The duty cycle of a PWM waveform is the ratio of the
pulse width (time spent in the ON state) to the total period of the waveform. It is
typically expressed as a percentage. For example, a duty cycle of 50% means the
pulse is ON half of the time and OFF half of the time.

3. **Generation Techniques**:

a. **Analog Comparator Method**: In this method, a sawtooth waveform

(triangle waveform) generated by a ramp generator is compared with an analog
signal. The output of the comparator goes high when the ramp voltage is less
than the analog signal and low otherwise. By adjusting the amplitude of the
analog signal, the width of the pulse can be varied, thus achieving PWM.

b. **Digital Counter Method**: Here, a digital counter counts up from

zero to a maximum value, generating a digital ramp waveform. This ramp
waveform is then compared with a reference value. When the ramp value is
less than the reference, the output is high, and when it exceeds the reference,
the output is low. By adjusting the reference value, the width of the pulse can
be controlled.

c. **Sigma-Delta Modulation**: This method involves comparing the

input analog signal with a high-frequency triangular waveform. The result of
the comparison is integrated and quantized to generate a digital output. By
varying the integration period, the pulse width can be controlled.

4. **Applications**: PWM is widely used in various applications such as motor

speed control, power regulation in inverters and amplifiers, LED dimming, and
audio signal processing. Its efficiency, simplicity, and ability to provide precise
control over output make it a preferred choice in many electronic systems.

In summary, PWM is a versatile technique for encoding analog information

into a digital signal by varying the width of pulses in a periodic waveform. Its
generation methods vary, but they all aim to control the duty cycle of the
waveform to achieve the desired output

Result and Conclusion:

Write a MATLAB Code for the generation of PWM. Observe the output waveform. Paste the
screenshot of all the waveform. (Your name should be reflected in the whole screenshot)

Note: Upload single pdf file for all above details

Note: Upload single pdf file for all above (1-8 Experiment) details