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    Department of Computer Engineering

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    Neha
    The document describes a digital signal processing lab experiment conducted by a student to study classification and sampling of signals. It includes: 1) An introduction to the sampling theorem and Nyquist criterion which establishes the minimum required sample rate to reconstruct a signal without aliasing. 2) Code and output showing the classification of different types of signals like discrete, exponential, ramp, step, sine and cosine waves. 3) Code and output demonstrating upsampling of a sinusoidal signal by inserting zeros between samples to increase the sample rate without changing the bandwidth. 4) A conclusion stating the key concepts of signal classification and sampling were studied using programs in the Scilab tool.

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    Department of Computer Engineering

    Uploaded by

    Neha
    26 views9 pages
    The document describes a digital signal processing lab experiment conducted by a student to study classification and sampling of signals. It includes: 1) An introduction to the sampling theorem and Nyquist criterion which establishes the minimum required sample rate to reconstruct a signal without aliasing. 2) Code and output showing the classification of different types of signals like discrete, exponential, ramp, step, sine and cosine waves. 3) Code and output demonstrating upsampling of a sinusoidal signal by inserting zeros between samples to increase the sample rate without changing the bandwidth. 4) A conclusion stating the key concepts of signal classification and sampling were studied using programs in the Scilab tool.

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    The document describes a digital signal processing lab experiment conducted by a student to study classification and sampling of signals. It includes: 1) An introduction to the sampling theorem and Nyquist criterion which establishes the minimum required sample rate to reconstruct a signal without aliasing. 2) Code and output showing the classification of different types of signals like discrete, exponential, ramp, step, sine and cosine waves. 3) Code and output demonstrating upsampling of a sinusoidal signal by inserting zeros between samples to increase the sample rate without changing the bandwidth. 4) A conclusion stating the key concepts of signal classification and sampling were studied using programs in the Scilab tool.

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    © All Rights Reserved
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    Download as docx, pdf, or txt
    26 views9 pages

    Department of Computer Engineering

    Uploaded by

    Neha
    The document describes a digital signal processing lab experiment conducted by a student to study classification and sampling of signals. It includes: 1) An introduction to the sampling theorem and Nyquist criterion which establishes the minimum required sample rate to reconstruct a signal without aliasing. 2) Code and output showing the classification of different types of signals like discrete, exponential, ramp, step, sine and cosine waves. 3) Code and output demonstrating upsampling of a sinusoidal signal by inserting zeros between samples to increase the sample rate without changing the bandwidth. 4) A conclusion stating the key concepts of signal classification and sampling were studied using programs in the Scilab tool.

    Copyright:

    © All Rights Reserved
    Download as DOCX, PDF, TXT or read online from Scribd
    Download as docx, pdf, or txt
    of 9

    DEPARTMENT OF COMPUTER ENGINEERING

    CSL701 Digital Signal and Image Processing Lab


    Seventh Semester, 2021-2022 (Odd Semester)

    Nameof Student : Neha Yadav

    RollNo. : 73

    Division : BE –CMPN

    Day/Session : Monday/Afternoon

    Venue : GoogleMeet

    ExperimentNo. 1

    Title of Experiment : To study classification and sampling of signal

    Date of Conduction : 26/07/2021

    Date of Submission : 1/08/2021

    Particulars Max.Marks MarksObtained


    Preparedness and Efforts(PE) 3
    Knowledge of tools(KT) 3
    Debugging and results(DR) 3
    Documentation(DN) 3
    Punctuality & Lab Ethics(PL) 3
    Total 15
    Grades – Meet Expectations (3 Marks), Moderate Expectations (2 Marks), Below Expectations (1 Mark)

    Checked and Verified by


    Nameof Faculty : Prof. Neelam Phadnis
    Signature :
    Date :7/08/2021
    EXPERIMENTNO:1 ROLL NO:73

    SAMPLING OF SIGNAL

    AIM: To study classification and sampling of signal.

    THEORY:

    SAMPLING THEOREM

    In the field of digital signal processing, the sampling theorem is a fundamental bridge
    between continuous-time signals (often called "analog signals") and discrete-time signals (often called
    "digital signals"). It establishes a sufficient condition for a sample rate that permits a discrete sequence
    of samples to capture all the information from a continuous-time signal of finitebandwidth.

    If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its
    ordinates at a series of points spaced 1/(2B) seconds apart.

    A sufficient sample-rate is therefore 2B samples/second, or anything larger. Equivalently, for a given


    sample rate fs, perfect reconstruction is guaranteed possible for a bandlimitB < fs/2.

    When the bandlimit is too high (or there is no bandlimit), the reconstruction exhibits imperfections
    known as aliasing. Modern statements of the theorem are sometimes careful to explicitly state that x(t)
    must contain no sinusoidal component at exactly frequency B, or that B must be strictly less than ½ the
    sample rate. The two thresholds, 2B and fs/2 are respectively called the Nyquist rate and Nyquist
    frequency. And respectively, they are attributes of x(t) and of the sampling equipment. The condition
    described by these inequalities is called the Nyquist criterion, or sometimes the Raabe condition. The
    theorem is also applicable to functions of other domains, such as space, in the case of a digitized
    image. The only change, in the case of other domains, is the units of measure applied to t, fs, andB.
    The symbol T = 1/fs is customarily used to represent the interval between samples and is called
    the sample period or sampling interval. And the samples of function x(t) are commonly denoted
    by x[n] = x(nT) (alternatively "xn" in older signal processing literature), for all integer values of n. A
    mathematically ideal way to interpolate the sequence involves the use of sinc functions. Each sample in
    the sequence is replaced by a sinc function, centered on the time axis at the original location of the
    sample, nT, with the amplitude of the sinc function scaled to the sample value, x[n]. Subsequently, the
    sinc functions are summed into a continuous function. A mathematically equivalent method is to
    convolve one sinc function with a series of Dirac delta pulses, weighted by the sample values. Neither
    method is numerically practical. Instead, some type of approximation of the sinc functions, finite in
    length, is used. The imperfections attributable to the approximation are known as interpolationerror.

    Practical digital-to-analog converters produce neither scaled and delayed sinc functions, nor ideal Dirac
    pulses. Instead they produce a piecewise-constant sequence of scaled and delayed rectangular
    pulses (the zero-order hold), usually followed by an "anti-imaging filter" to clean up spurious high-
    frequencycontent.
    CLASSIFICATION OF SIGNALS:

    CODE
    clear all;
    clf();
    clc;
    x1=0:5;
    y1=[ones(1,4),zeros(1,2)];
    y2=[ones(1,6)];
    y3=exp(x1);
    y4=x1;

    subplot(2,3,1);
    plot2d3(x1,y1);
    xlabel('Time');
    ylabel('Amplitude');
    title('Sample Discrete Signal');

    subplot(2,3,2);
    plot2d3(x1,y3);
    xlabel('Time');
    ylabel('Amplitude');
    title('Exponential Signal');

    subplot(2,3,3);
    plot2d3(x1,y4);
    xlabel('Time');
    ylabel('Amplitude');
    title('Unit Ramp Signal');

    subplot(2,3,4);
    plot2d3(x1,y2);
    xlabel('Time');
    ylabel('Amplitude');
    title('Unit Step Discrete Signal');

    subplot(2,3,5);
    sx=[0:0.1:2*%pi];
    plot2d(sin(sx));
    title('Sine Wave');

    subplot(2,3,6);
    cx=[0:0.1:2*%pi];
    plot2d(cos(cx));
    title('Cosine Wave');
    OUTPUT
    UP-SCALING:

    CODE
    clc;
    clf();
    clearall;
    n = input('Enter Length of Sinuisoidal Sequence ');
    l = input('Enter Length of Sampling Factor ');
    n = 0:0.01:n-1;
    x = sin(2*%pi*n);
    y = zeros(1,l*length(x));
    y([1:l:length(y)])=x;
    subplot(2,2,1);
    plot2d3(n,x);
    xlabel('Time');
    ylabel('Amplitude');
    title('Input Signals');
    subplot(2,2,2);
    plot2d3(n,y(1:length(x)));
    xlabel('Time');
    ylabel('Amplitude');
    title('Upsampling');

    OUTPUT
    Enter Length of Sinuisoidal Sequence = 10
    Enter Length of Sampling Factor = 5
    CONCLUSION:

    Thus we have studied the classification and sampling of signals using the Scilab Tool to execute the
    programs.

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