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lab4DSPBEE13 Sampling

This document provides instructions for a lab experiment on sampling and quantization of audio signals in MATLAB. The objectives are to get familiar with sampling, analyzing downsampled signals in the frequency domain, and quantizing signals. Students are asked to downsample an audio signal by factors of 2, 3, 5, and 10, with and without anti-aliasing filters, and compare the input and output spectra. They also quantize a signal for different numbers of quantization levels and plot the original and quantized signals. The report must include MATLAB codes, results with discussion, and a conclusion.

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22 views5 pages

lab4DSPBEE13 Sampling

This document provides instructions for a lab experiment on sampling and quantization of audio signals in MATLAB. The objectives are to get familiar with sampling, analyzing downsampled signals in the frequency domain, and quantizing signals. Students are asked to downsample an audio signal by factors of 2, 3, 5, and 10, with and without anti-aliasing filters, and compare the input and output spectra. They also quantize a signal for different numbers of quantization levels and plot the original and quantized signals. The report must include MATLAB codes, results with discussion, and a conclusion.

Uploaded by

Maryam
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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Department of Electrical Engineering

Faculty Member: __________________ Dated: ____________________

Course/Section: Semester:

EE-330 Digital Signal Processing

Lab 4: Sampling and Quantization of Audio Signal


PLO4-CLO4 PLO5- PLO8- PLO9-
CLO5 CLO6 CLO7
Name Reg. No Viva / Analysis Modern Ethics Individual
Quiz / Lab of data Tool and and Team
Performa in Lab Usage Safety Work
nce Report

5 Marks 5 Marks 5 Marks 5 Marks 5 Marks

Digital Signal Processing Page 1


Lab4: Sampling and Quantization of audio signal in
Matlab

Objectives

The objective in this lab is down sampled audio signal and its analysis in frequency domain.

 Familiarization with sampling

 Analysis of down sampled signal in frequency domain

 Quantization of signal

Lab Instructions
 The students should perform and demonstrate each lab task separately for step-wise evaluation
 Each group shall submit one lab report on LMS within 6 days after lab is conducted. Lab report
submitted via email will not be graded.
. Students are however encouraged to practice on their own in spare time for enhancing their
Lab Report Instructions
All questions should be answered precisely to get maximum credit. Lab report must ensure following
items:
 Lab objectives
 MATLAB codes
 Results (graphs/tables) duly commented and discussed
 Conclusion

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5 Sampling of audio signal in MATLAB
5.1 Sampling
In this lab you will gain some practical knowledge about how to handle real world signals. In any
modern signal processing system (e.g. in a telecommunication system), signal acquisition, its
processing and efficient storage/transmission are the critical steps. In the class lectures, you have
gained the knowledge about sampling of a continuous-time signal, change of sampling rates and
their hierarchical criteria. This lab covers the above mentioned tasks i.e., change of sampling
rate.

Change of Sampling Rate:

You are familiar with the above figure for downsampling. For a discrete-time signal to be
sampled by the factor of M, you need to first pass the signal from the low pass filter with a
certain cutoff frequency to avoid the frequency aliasing in the downsampled signal.

5.1.1 LAB TASK-1:


You are given a speech signal. Consider it a discrete-time signal with the sampling frequency
𝑓𝑠=16 𝑘𝐻𝑧.
1. Load the signal in Matlab using the function audioread. Listen to the signal using sound.
2. Design a 6th order low-pass butterworth filter. Hint: see Matlab help for butter and filter. The
butter command takes the normalized cutoff frequency (in the range 0-1) as an input argument
where the maximum 1 means 𝑓𝑆/2
3. Consider the maximum frequency of the speech signal 𝑓𝑁 = 𝑓𝑆/2. Apply the filter.
4. Now downsample the filtered signal by the factor of 2 i.e., M=2. Do this manually by picking
up every alternative sample and storing it in a different array.
5. See the Matlab help for the function downsample. Apply this function for downsampling the
signal by the factors M = 3,5,10. Listen to the output signal in every case and prepare your
conclusions. Also plot the spectrum of the input and output signal in a subplots for original and
three cases for different M.
6. For M = 10, avoid the anti-aliasing filter and directly downsample the speech to listen if there
is any difference. Also plot the spectrum using code given below for input and output signal .

Digital Signal Processing Page 3


Code to plot spectrum

% x is your input signal Fs is sampling frequencey


[x,Fs] = audioread('sample.wav');
L=length(x);
NFFT = 2^nextpow2(L);% Next power of 2 from length of y
Y = fft(x,NFFT)/L;
f = Fs/2*linspace(0,1,NFFT/2+1);
% Plot single-sided amplitude spectrum.
subplot(2,1,2)
plot(f,2*abs(Y(1:NFFT/2+1))),grid on
title('Single-Sided Amplitude Spectrum of filtered signal')
xlabel('Frequency (Hz)')

Quantization (self-study task)

The above-mentioned figure a quantizer function . Given below is the quantization function in
Matlab (algorithm)
X=(1:99)*(8/100)-4;
N = 8; %number of quantization levels.
% find the highest value point in the signal, round it to the upper limit.
% find the lowest value point in the signal, round it to the lower limit.
qstep = (high-low)/N;
Q = floor((X-low)/qstep);
low = low + qstep/2;
Y = low + qstep*Q;

Digital Signal Processing Page 4


5.1.2 TASK-2

1. Redo the above code in Matlab, plot the original and quantized signal in a
subplot for N = 8, 16, 32, 64.

2. Explain every step in the code.

Digital Signal Processing Page 5

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