Location via proxy:   [ UP ]  
[Report a bug]   [Manage cookies]                
Scribd Logo
Upload

Welcome to Scribd!

  • 31 views

    Lab S: Sampling and Aliasing: 0.1 Simple Examples

    Uploaded by

    Andrea Fields
    This document discusses sampling and aliasing of signals in Matlab. It includes examples of chirp signals with different sampling frequencies, sampling sinusoidal signals at different rates, and subsampling an audio file to demonstrate how high notes become low notes due to folding. It also describes a method to adjust the pitch of a sound by low-pass filtering, downsampling, modulating with a cosine, and reconstructing to produce a shifted signal. Code examples are provided to generate signals, apply the pitch shifting method, and create a function to shift the frequency of an audio signal.

    Copyright:

    © All Rights Reserved
    Download as PDF, TXT or read online from Scribd

    Lab S: Sampling and Aliasing: 0.1 Simple Examples

    Uploaded by

    Andrea Fields
    31 views4 pages
    This document discusses sampling and aliasing of signals in Matlab. It includes examples of chirp signals with different sampling frequencies, sampling sinusoidal signals at different rates, and subsampling an audio file to demonstrate how high notes become low notes due to folding. It also describes a method to adjust the pitch of a sound by low-pass filtering, downsampling, modulating with a cosine, and reconstructing to produce a shifted signal. Code examples are provided to generate signals, apply the pitch shifting method, and create a function to shift the frequency of an audio signal.

    Original Description:

    hjk

    Original Title

    Sampling

    Copyright

    © © All Rights Reserved

    Available Formats

    PDF, TXT or read online from Scribd

    Did you find this document useful?

    Is this content inappropriate?

    This document discusses sampling and aliasing of signals in Matlab. It includes examples of chirp signals with different sampling frequencies, sampling sinusoidal signals at different rates, and subsampling an audio file to demonstrate how high notes become low notes due to folding. It also describes a method to adjust the pitch of a sound by low-pass filtering, downsampling, modulating with a cosine, and reconstructing to produce a shifted signal. Code examples are provided to generate signals, apply the pitch shifting method, and create a function to shift the frequency of an audio signal.

    Copyright:

    © All Rights Reserved
    Download as PDF, TXT or read online from Scribd
    Download as pdf or txt
    31 views4 pages

    Lab S: Sampling and Aliasing: 0.1 Simple Examples

    Uploaded by

    Andrea Fields
    This document discusses sampling and aliasing of signals in Matlab. It includes examples of chirp signals with different sampling frequencies, sampling sinusoidal signals at different rates, and subsampling an audio file to demonstrate how high notes become low notes due to folding. It also describes a method to adjust the pitch of a sound by low-pass filtering, downsampling, modulating with a cosine, and reconstructing to produce a shifted signal. Code examples are provided to generate signals, apply the pitch shifting method, and create a function to shift the frequency of an audio signal.

    Copyright:

    © All Rights Reserved
    Download as PDF, TXT or read online from Scribd
    Download as pdf or txt
    of 4

    Lab S: Sampling and Aliasing

    The domain of Matlab is discrete time signals, and without delving into the world of hardware, it is impossible for us to truly consider the problems of sampling and reconstructing.
    We can fake it though, by using over-sampled signals or computed-on-demand functions and
    concentrating on the boundary where discrete-time effects get strange.

    0.1
    0.1.1

    Simple Examples
    Chirp Folding

    Remember that chirps are signals with linearly increasing frequency. That is x(t) = cos((t+
    0 )t + ).
    1. Using a sampling frequency of 8000 Hz, generate a chirp signal that goes from 200 Hz
    to 800 Hz. Listen to it. Make a spectrogram of it.
    2. Now make a chirp that goes from 200 Hz to 10200 Hz. Listen and make a spectrogram.

    0.1.2

    Sinusoid Sampling

    1. Generate about 20 periods of a sinusoid of your choice. Choose a sampling frequency


    to get at least 20 samples per period.
    2. On the same plot or in subplots, generate 4 more sinusoid plots by using different
    sampling rates, covering a range of apparent frequencies, including some sinusoids
    that appear to have a higher frequency than others for which the sampling frequency
    was higher. That is, by choosing different, slower, sampling rates, you should be
    able to get very low apparent frequencies. If you continue to decrease your sampling
    frequency, you should then higher apparent frequencies (though not as high as your
    original signal).

    0.1.3

    Folding in Music

    1. Find a .wav file of a simple song (preferably one with distinct notes and minimal
    singing). Read it into Matlab using wavread. Note that Matlab cannot read all kinds
    of .wav files and you may have to try more than one to find one for which the wavread
    1

    command is successful. http://www.chivalry.com/cantaria/ has a collection of files


    that might be suitable. Your lab 3 songs are another good choice.
    2. If the song is longer than 30 seconds, take a subsection of it. Also, if the song is in stereo
    (it has two columns), isolate one of them with xx = song(:, 1) or xx = song(1,
    :). Make a spectrogram of that subsection. Look at how high up the energy in the
    spectrum goes.
    3. Subsample the data (with a command like yy = xx(1:N:length(xx)) and play it
    with the new sampling frequency to see when you can notice the distortion. Can you
    subsample the signal enough to hear the high notes become low notes?

    0.2

    Adjusting Pitch

    Using these methods, you can adjust the pitch of a sound without adjust its length, or visa
    versa. We will concentrate on the former since it is less programmatically intensive. See the
    figure for a graphical representation of our method. This method allows us to finely adjust
    the pitch within a limited range, so long as we are willing to down-sample the original data
    by 2.

    1. Generate your low-pass filter using the equation X() =


    x[n] =

    1, 0 || W
    0, W < ||

    sin(W n)
    n

    Where W is 2(.25 fx /fs ) and fx is the desired frequency shift. Use the equation on
    the right to get the frequency-domain representation on the left. n should go from 0
    to between 50 and 500.
    Plot the result. Now plot(abs(fftshift(fft(xx)))) where xx is your x[n] above to
    see the spectrum. Comment on any abnormalities.
    2. Using your low-pass filter, perform the sequence of functions shown in the figure.
    You can use commands like plot(abs(fftshift(fft(xx)))) to confirm that your
    method is going according to plan. You can apply the low-pass filter with the command
    conv(xx, lowpass) (where xx is the input and lowpass is the filter). conv will return
    vectors longer than its inputs, so you may want to generate your t vectors for the cosines
    on-the-fly with (0:1/fs:((length(uu) - 1) / fs)). Note that the f s used in the
    second cosine-modulation should be 1/2 of the original f s.
    3. Listen to and make a wavwrite of the result. Comment on its effectiveness. Try doubly
    applying the lowpass filter everywhere that you apply it.
    4. Make your code a function which takes sound data, a sampling frequency, and a shift
    frequency and returns subsampled data with the appropriate shift.

    You might also like