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    Activity No. 8 IIR Filter Design: Date Performed

    Uploaded by

    Fredo Ramirez
    The document describes designing IIR filters to filter audio signals. It involves: 1. Designing a 2nd order Butterworth bandpass filter with cutoffs of 500Hz and 1000Hz to filter an audio file, saving the filtered output. 2. Designing a 4th order Butterworth lowpass filter with a cutoff of 200Hz and highpass filter with a cutoff of 250Hz to separate the bass and treble parts of another audio file, then combining and saving the filtered stereo output. 3. Plotting the magnitude responses of the designed filters to verify their frequency responses.

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    © All Rights Reserved
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    Activity No. 8 IIR Filter Design: Date Performed

    Uploaded by

    Fredo Ramirez
    43 views7 pages
    The document describes designing IIR filters to filter audio signals. It involves: 1. Designing a 2nd order Butterworth bandpass filter with cutoffs of 500Hz and 1000Hz to filter an audio file, saving the filtered output. 2. Designing a 4th order Butterworth lowpass filter with a cutoff of 200Hz and highpass filter with a cutoff of 250Hz to separate the bass and treble parts of another audio file, then combining and saving the filtered stereo output. 3. Plotting the magnitude responses of the designed filters to verify their frequency responses.

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    The document describes designing IIR filters to filter audio signals. It involves: 1. Designing a 2nd order Butterworth bandpass filter with cutoffs of 500Hz and 1000Hz to filter an audio file, saving the filtered output. 2. Designing a 4th order Butterworth lowpass filter with a cutoff of 200Hz and highpass filter with a cutoff of 250Hz to separate the bass and treble parts of another audio file, then combining and saving the filtered stereo output. 3. Plotting the magnitude responses of the designed filters to verify their frequency responses.

    Copyright:

    © All Rights Reserved
    Download as DOCX, PDF, TXT or read online from Scribd
    Download as docx, pdf, or txt
    43 views7 pages

    Activity No. 8 IIR Filter Design: Date Performed

    Uploaded by

    Fredo Ramirez
    The document describes designing IIR filters to filter audio signals. It involves: 1. Designing a 2nd order Butterworth bandpass filter with cutoffs of 500Hz and 1000Hz to filter an audio file, saving the filtered output. 2. Designing a 4th order Butterworth lowpass filter with a cutoff of 200Hz and highpass filter with a cutoff of 250Hz to separate the bass and treble parts of another audio file, then combining and saving the filtered stereo output. 3. Plotting the magnitude responses of the designed filters to verify their frequency responses.

    Copyright:

    © All Rights Reserved
    Download as DOCX, PDF, TXT or read online from Scribd
    Download as docx, pdf, or txt
    You are on page 1of 7

    ___________________

    ___________________
    ECE515FL
    ___________________

    ACTIVITY NO. 8
    IIR Filter Design

    Date Performed:
    ______________
    Date Submitted:
    ______________

    Activity:
    1. Design a 2nd-order Butterworth bandpass digital filter with fc=500Hz when the
    sampling frequency is 22050Hz.
    (a) Plot the magnitude response over the frequency range 300Hz to 700Hz.
    Program:
    clc;clear;
    stacksize(10^8)
    s=poly(0,'s');
    //PRE-WARP THE CUT-OFF FREQUENCIES
    //Prewarping equation: wa=2*Fs*tan(wd/2/Fs)
    //wa - pre-warped analog filter cut-off frequencies, in radians/second
    //wd - digital filter cut-off frequencies, in radians/second
    Fs=22050; //sampling frequency
    fd1=500; //3dB cut-off
    fd2=1000; //3dB cut-off
    wd1=2*%pi*fd1;
    wd2=2*%pi*fd2;
    wa1=2*Fs*tan(wd1/2/Fs);
    wa2=2*Fs*tan(wd2/2/Fs);
    //GENERATE THE BUTTERWORTH DENOMINATOR POLYNOMIAL
    n=2; //number of poles
    Hs_den=2; //Initialize the denominator polynomial
    //nth-order
    for k=1:n
    Theta=(%pi/2)+((2*k-1)*%pi)/(2*n);
    pole=exp(Theta*%i);
    Hs_den=Hs_den*(s-pole);
    end;
    Hslp=1/real(Hs_den)
    //TRANSFORM THE LOWPASS TO BANDPASS
    Hs=horner(Hslp,(s^2+wa1*wa2)/((wa2-wa1)*s))
    //TRANSFORM H(s) TO H(z) USING BILINEAR TRANSFORMATION
    z=poly(0,'z');
    Hz=horner(Hs,2*Fs*(z-1)/(z+1))
    //PLOT THE FREQUENCY RESPONSE OF THE DIGITAL FILTER
    1

    ___________________
    ___________________
    ECE515FL
    ___________________

    ACTIVITY NO. 8
    IIR Filter Design

    Date Performed:
    ______________
    Date Submitted:
    ______________

    Fs=22050;
    f=300:700;
    W=(2*%pi*f)/Fs;
    HW=horner(Hz,exp(W*%i));
    plot2d(f,abs(HW)); xgrid();

    (b) Load the audio file dinar_16bit_22050Hz_stereo.wav and filter the audio signal.
    Filter each channel separately and then save the result as a stereo signal. Save the filtered
    audio signal as filter1.wav.

    Program:
    clc;clear;
    stacksize(10^8)
    s=poly(0,'s');
    //PRE-WARP THE CUT-OFF FREQUENCIES
    //Prewarping equation: wa=2*Fs*tan(wd/2/Fs)
    //wa - pre-warped analog filter cut-off frequencies, in radians/second
    //wd - digital filter cut-off frequencies, in radians/second
    Fs=22050; //sampling frequency
    fd1=500; //3dB cut-off
    fd2=1000; //3dB cut-off
    2

    ___________________
    ___________________
    ECE515FL
    ___________________

    ACTIVITY NO. 8
    IIR Filter Design

    Date Performed:
    ______________
    Date Submitted:
    ______________

    wd1=2*%pi*fd1;
    wd2=2*%pi*fd2;
    wa1=2*Fs*tan(wd1/2/Fs);
    wa2=2*Fs*tan(wd2/2/Fs);
    //GENERATE THE BUTTERWORTH DENOMINATOR POLYNOMIAL
    n=2; //number of poles
    Hs_den=2; //Initialize the denominator polynomial
    //nth-order
    for k=1:n
    Theta=(%pi/2)+((2*k-1)*%pi)/(2*n);
    pole=exp(Theta*%i);
    Hs_den=Hs_den*(s-pole);
    end;
    Hslp=1/real(Hs_den)
    //TRANSFORM THE LOWPASS TO BANDPASS
    Hs=horner(Hslp,(s^2+wa1*wa2)/((wa2-wa1)*s))

    //TRANSFORM H(s) TO H(z) USING BILINEAR TRANSFORMATION


    z=poly(0,'z');
    Hz=horner(Hs,2*Fs*(z-1)/(z+1))
    //FILTER AN AUDIO FILE
    stacksize(5*10^7);
    x=wavread('F:\DINAR_16bit_22050_stereo.wav');
    yleft=dsimul(tf2ss(Hz),x(1,:)); //Filter the left channel samples
    yright=dsimul(tf2ss(Hz),x(2,:)); //Filter the right channel samples
    yleft=yleft/max(abs(yleft)); //Normalize the filtered left channel samples
    yright=yright/max(abs(yright)); //Normalize the filtered right channel samples
    z=[yleft;yright];
    wavwrite(z, Fs, 'F:\filter1.wav');

    2. Filter the audio signal in sonatina.wav to separate its treble and bass parts. The
    bass part has a highest frequency of 196Hz while the treble part has a lowest
    frequency of 261.63Hz. Hence, to extract the bass part, use a lowpass filter with
    fc=200Hz. For the treble part, use a highpass filter with fc=250Hz. Use 4th-order
    Butterworth filters.
    3

    ___________________
    ___________________
    ECE515FL
    ___________________

    ACTIVITY NO. 8
    IIR Filter Design

    (a) Plot the magnitude response of the filters from 0 to 400Hz.


    Program:
    clc;clear;
    stacksize(10^8)
    s=poly(0,'s');
    //PRE-WARP THE CUT-OFF FREQUENCIES
    //Prewarping equation: wa=2*Fs*tan(wd/2/Fs)
    //wa - pre-warped analog filter cut-off frequencies, in radians/second
    //wd - digital filter cut-off frequencies, in radians/second
    Fs=11025; //sampling frequency
    fd1=200; //3dB cut-off lowpass
    fd2=250; //3dB cut-off highpass
    wd1=2*%pi*fd1;
    wd2=2*%pi*fd2;
    wa1=2*Fs*tan(wd1/2/Fs);
    wa2=2*Fs*tan(wd2/2/Fs);
    //GENERATE THE BUTTERWORTH DENOMINATOR POLYNOMIAL
    n=4; //number of poles
    Hs_den=4; //Initialize the denominator polynomial
    //nth-order
    for k=1:n
    Theta=(%pi/2)+((2*k-1)*%pi)/(2*n);
    pole=exp(Theta*%i);
    Hs_den=Hs_den*(s-pole);
    end;
    Hslp=1/real(Hs_den)
    //TRANSFORM THE LOWPASS TO LOWPASS
    Hs1=horner(Hslp,s/wa1);
    //TRANSFORM THE LOWPASS TO HIGHPASS
    Hs2=horner(Hslp,wa2/s);
    //PLOT THE FREQUENCY RESPONSE OF THE ANALOG FILTERS
    f=0:(Fs/2);
    w=2*%pi*f;
    Hw1=horner(Hs1,w*%i);
    subplot(2,1,1)
    plot2d(f,abs(Hw1),13); xgrid();
    Hw2=horner(Hs2,w*%i);
    4

    Date Performed:
    ______________
    Date Submitted:
    ______________

    ___________________
    ___________________
    ECE515FL
    ___________________

    ACTIVITY NO. 8
    IIR Filter Design

    Date Performed:
    ______________
    Date Submitted:
    ______________

    subplot(2,1,2)
    plot2d(f,abs(Hw2),13); xgrid();
    //TRANSFORM H(s) TO H(z) USING BILINEAR TRANSFORMATION
    z=poly(0,'z');
    Hz1=horner(Hs1,2*Fs*(z-1)/(z+1));
    Hz2=horner(Hs2,2*Fs*(z-1)/(z+1));

    (b) Save the filtered signal in stereo with the bass part on the left channel and the
    treble part on the right channel. Save as sonatina_stereo.wav .

    Program:
    clc;clear;
    stacksize(10^8)
    s=poly(0,'s');
    5

    ___________________
    ___________________
    ECE515FL
    ___________________

    ACTIVITY NO. 8
    IIR Filter Design

    //PRE-WARP THE CUT-OFF FREQUENCIES


    //Prewarping equation: wa=2*Fs*tan(wd/2/Fs)
    //wa - pre-warped analog filter cut-off frequencies, in radians/second
    //wd - digital filter cut-off frequencies, in radians/second
    Fs=11025; //sampling frequency
    fd1=200; //3dB cut-off lowpass
    fd2=250; //3dB cut-off highpass
    wd1=2*%pi*fd1;
    wd2=2*%pi*fd2;
    wa1=2*Fs*tan(wd1/2/Fs);
    wa2=2*Fs*tan(wd2/2/Fs);
    //GENERATE THE BUTTERWORTH DENOMINATOR POLYNOMIAL
    n=4; //number of poles
    Hs_den=4; //Initialize the denominator polynomial
    //nth-order
    for k=1:n
    Theta=(%pi/2)+((2*k-1)*%pi)/(2*n);
    pole=exp(Theta*%i);
    Hs_den=Hs_den*(s-pole);
    end;
    Hslp=1/real(Hs_den)
    //TRANSFORM THE LOWPASS TO LOWPASS
    Hs1=horner(Hslp,s/wa1);
    //TRANSFORM THE LOWPASS TO HIGHPASS
    Hs2=horner(Hslp,wa2/s);
    //PLOT THE FREQUENCY RESPONSE OF THE ANALOG FILTERS
    f=0:(Fs/2);
    w=2*%pi*f;
    Hw1=horner(Hs1,w*%i);
    subplot(2,2,1)
    plot2d(f,abs(Hw1)); xgrid();
    Hw2=horner(Hs2,w*%i);
    subplot(2,2,2)
    plot2d(f,abs(Hw2)); xgrid();
    //TRANSFORM H(s) TO H(z) USING BILINEAR TRANSFORMATION
    z=poly(0,'z');
    6

    Date Performed:
    ______________
    Date Submitted:
    ______________

    ___________________
    ___________________
    ECE515FL
    ___________________

    ACTIVITY NO. 8
    IIR Filter Design

    Date Performed:
    ______________
    Date Submitted:
    ______________

    Hz1=horner(Hs1,2*Fs*(z-1)/(z+1));
    Hz2=horner(Hs2,2*Fs*(z-1)/(z+1));
    //PLOT THE FREQUENCY RESPONSE OF THE DIGITAL FILTER
    W=0.001:0.001:%pi;
    f=W*Fs/2/%pi;
    HW1=horner(Hz1,exp(W*%i));
    subplot(2,2,3)
    plot2d(f,abs(HW1)); xgrid();
    HW2=horner(Hz2,exp(W*%i));
    subplot(2,2,4)
    plot2d(f,abs(HW2)); xgrid();
    //Normalize transfer function
    Hz1=Hz1/(max(abs(HW1)));
    Hz2=Hz2/(max(abs(HW2)));
    //FILTER AN AUDIO FILE
    stacksize(5*10^7);
    x=wavread('F:\sonatina.wav');
    yleft=dsimul(tf2ss(Hz1),x); //Filter the left channel samples
    yright=dsimul(tf2ss(Hz2),x); //Filter the right channel samples
    yleft=yleft/max(abs(yleft)); //Normalize the filtered left channel samples
    yright=yright/max(abs(yright)); //Normalize the filtered right channel samples
    z=[yleft;yright];
    wavwrite(z, Fs, 'F:\sonatina_stereo.wav');

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