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Department of Electronics and Telecommunication Engineering

GOVERNMENT COLLEGE OF ENGINEERING,


KARAD

PROJECT REPORT

Project Name : To design RC Low Pass Filter using MATLAB Simulink.

Subject : Control System

Branch of Students : T.Y.ENTC

Name of Students : 1) Abhijeet Vijaykumar Patil (17151211)

2) Sumit Chandrakant Valsangkar (17151213)

YEAR :2019-2020
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INDEX

Sr. Details Page No.


No
1) Introduction 03
2) Circuit Diagram & Equation 04
3) Matlab Simulink Design 05
4) System Output 09
5) Conclusion 09
6) Applications 09
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Introduction

A low-pass filter (LPF) is a filter that passes signals with a frequency lower than a selected cutoff


frequency and attenuates signals with frequencies higher than the cutoff frequency. The exact frequency
response of the filter depends on the filter design. The filter is sometimes called a high-cut filter, or treble-
cut filter in audio applications. A low-pass filter is the complement of a high-pass filter.

Low-pass filters exist in many different forms, including electronic circuits such as a hiss filter used
in audio, anti-aliasing filters for conditioning signals prior to analog-to-digital conversion, digital filters for
smoothing sets of data, acoustic barriers, blurring of images, and so on. The moving average operation used
in fields such as finance is a particular kind of low-pass filter, and can be analyzed with the same signal
processing techniques as are used for other low-pass filters. Low-pass filters provide a smoother form of a
signal, removing the short-term fluctuations and leaving the longer-term trend.

Filter designers will often use the low-pass form as a prototype filter. That is, a filter with unity bandwidth
and impedance. The desired filter is obtained from the prototype by scaling for the desired bandwidth and
impedance and transforming into the desired bandform (that is low-pass, high-pass, band-pass or band-stop).

An ideal low-pass filter completely eliminates all frequencies above the cutoff frequency while passing
those below unchanged; its frequency response is a rectangular function and is a brick-wall filter. The
transition region present in practical filters does not exist in an ideal filter. An ideal low-pass filter can be
realized mathematically (theoretically) by multiplying a signal by the rectangular function in the frequency
domain or, equivalently, convolution with its impulse response, a sinc function, in the time domain.

However, the ideal filter is impossible to realize without also having signals of infinite extent in time, and so
generally needs to be approximated for real ongoing signals, because the sinc function's support region
extends to all past and future times. The filter would therefore need to have infinite delay, or knowledge of
the infinite future and past, in order to perform the convolution. It is effectively realizable for pre-recorded
digital signals by assuming extensions of zero into the past and future, or more typically by making the
signal repetitive and using Fourier analysis.

Real filters for real-time applications approximate the ideal filter by truncating and windowing the infinite
impulse response to make a finite impulse response; applying that filter requires delaying the signal for a
moderate period of time, allowing the computation to "see" a little bit into the future. This delay is
manifested as phase shift. Greater accuracy in approximation requires a longer delay.

An ideal low-pass filter results in ringing artifacts via the Gibbs phenomenon. These can be reduced or
worsened by choice of windowing function, and the design and choice of real filters involves understanding
and minimizing these artifacts. For example, "simple truncation [of sinc] causes severe ringing artifacts," in
signal reconstruction, and to reduce these artifacts one uses window functions "which drop off more
smoothly at the edges."

Circuit Diagram & Equation


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Matlab Simulink Design

Fig:-Simulink design for low pass filter

Fig:- Design in subsystem block by equation


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Blocks Used:-
1. Subsystem block:-
 Create a Subsystem Block
 Add a Subsystem block to the model, and then add the blocks that make up the subsystem.
 Create a Subsystem block from the Ports & Subsystems library.
 Double-click the block to open it.
 In the empty subsystem window, create the subsystem contents. Use Inport blocks to represent input from
outside the subsystem and Outport blocks to represent external output.
 For example, this subsystem includes a Sum block and Inport and Outport blocks to represent input to and
output from the subsystem.

 When you close the subsystem window, the Subsystem block


includes a port for each Inport and Outport block.

2. Discrete Derivative block:-

 The Discrete Derivative block computes an optionally scaled


discrete time derivative as follows
y(tn)=K(u(tn)−u(tn−1)Ts)
where,
 u(tn) and y(tn) are the block input and output at the current time step, respectively.
 u(tn−1) is the block input at the previous time step.
 K is an optional scaling factor, specified using the Gain value parameter.
 Ts is the simulation's discrete step size, which must be fixed.

3. Scope:-
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 The Scope block displays its input with respect to simulation time. The Scope block can have
multiple axes (one per port); all axes have a common time range with independent y-axes. The Scope
allows you to adjust the amount of time and the range of input values displayed. You can move and
resize the Scope window and you can modify the Scope's parameter values during the simulation.
 When you start a simulation, Simulink does not open Scope windows, although it does write data to
connected Scopes. As a result, if you open a Scope after a simulation, the Scope's input signal or
signals will be displayed.
 If the signal is continuous, the Scope produces a point-to-point plot. If the signal is discrete, the
Scope produces a stair-step plot.
 The Scope provides toolbar buttons that enable you to zoom in on displayed data, display all the data
input to the Scope, preserve axis settings from one simulation to the next, limit data displayed, and
save data to the workspace. The toolbar buttons are labeled in this figure, which shows the Scope
window as it appears when you open a Scope block.

Working Of Filter:-
 In these design,we are taking two sine waves as a sources.
 1st sine wave is with Amplitude 50 and freq of 1rad/sec.
 2nd sine wave is with Amplitude 5 and freq of 10rad/sec.
 Here, we are adding two sources to get the frequency signal at output.
 At the output,We get signal with both high and low frequencies mixed.So,to get only low frequency
signal we have attached a low frequency block at output of signal using subsystem block.
 In these discrete derivative block is used to get derived voltage and to be multiplied with RC values
i.e. gain values.Which are again substracted from Input.
 As these at Output of designed system,The Low pass frequency signal is obtained.

Below are some screenshots of filter block and sources:-


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System Output

Conclusion:-
As these,we got the low frequency signal as required.Here, provided low signal is 1st signal which are
obtaining as it is.

Applications of Low pass Filter:-


 In electronics these filters are widely used in many applications. These filters are used as hiss filters
in audio speakers to reduce the high frequency hiss produced in the system and these are used as
inputs for sub woofers.
 These are also used in equalizers and audio amplifiers. In analog to digital conversion these are used
as anti-aliasing filters to control signals. In digital filters these are used in blurring of images,
smoothing sets of data signals. In radio transmitters to block harmonic emissions.
 In acoustics these filters are used to filter the high frequency signals from the transmitting sound
which will cause echo at higher sound frequencies.

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