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    Low Pass Filter Report

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    sumit valsangkar
    The document describes designing a low pass filter using MATLAB Simulink. Key steps include: 1) Creating a subsystem block with an equation for a low pass filter. 2) Adding blocks like a discrete derivative block and scope block. 3) Testing the filter by passing in signals with high and low frequencies and observing that only the low frequency signal passes through. Applications of low pass filters include reducing hiss in audio, smoothing data signals, and preventing aliasing in analog to digital conversion.

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    Low Pass Filter Report

    Uploaded by

    sumit valsangkar
    789 views9 pages
    The document describes designing a low pass filter using MATLAB Simulink. Key steps include: 1) Creating a subsystem block with an equation for a low pass filter. 2) Adding blocks like a discrete derivative block and scope block. 3) Testing the filter by passing in signals with high and low frequencies and observing that only the low frequency signal passes through. Applications of low pass filters include reducing hiss in audio, smoothing data signals, and preventing aliasing in analog to digital conversion.

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    The document describes designing a low pass filter using MATLAB Simulink. Key steps include: 1) Creating a subsystem block with an equation for a low pass filter. 2) Adding blocks like a discrete derivative block and scope block. 3) Testing the filter by passing in signals with high and low frequencies and observing that only the low frequency signal passes through. Applications of low pass filters include reducing hiss in audio, smoothing data signals, and preventing aliasing in analog to digital conversion.

    Copyright:

    © All Rights Reserved
    Download as DOCX, PDF, TXT or read online from Scribd
    Download as docx, pdf, or txt
    789 views9 pages

    Low Pass Filter Report

    Uploaded by

    sumit valsangkar
    The document describes designing a low pass filter using MATLAB Simulink. Key steps include: 1) Creating a subsystem block with an equation for a low pass filter. 2) Adding blocks like a discrete derivative block and scope block. 3) Testing the filter by passing in signals with high and low frequencies and observing that only the low frequency signal passes through. Applications of low pass filters include reducing hiss in audio, smoothing data signals, and preventing aliasing in analog to digital conversion.

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    1

    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
    2

    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
    3

    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


    4
    5

    Matlab Simulink Design

    Fig:-Simulink design for low pass filter

    Fig:- Design in subsystem block by equation


    6

    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:-
    7

     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:-


    8
    9

    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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