Lab 07-Sampling Rate Conversion

Lab Engineer: Muhammad Hammad

1 Objectives:
. The goal of this laboratory is to perform sampling rate conversion for any given arbitrary sequence
(D.T) or signal (C.T) by interpolation, decimation, up sampling, down sampling and resampling (i.e.
fractional value)
2 Time Required: 3 hrs
.
3 Programming Language: MATLAB
.
4 Software Required:
. (a). MATLAB 7 or above

PRE - LAB
SAMPLING PROCESS:

It is a process by which a continuous time signal is converted into discrete time signal. X[n] is the
discrete time signal obtained by taking samples of the analog signal x(t) every T seconds, where T
is the sampling period.

X[n] = x (t) x p (t)

Where p(t) is impulse train; T – period of the train

SAMPLING THEOREM:

It states that the band limited signal x(t) having no frequency components above Fmax Hz is
specified by the samples that are taken at a uniform rate greater than 2 Fmax Hz (Nyquist rate), or
the frequency equal to twice the highest frequency of x(t).
Fs ≥ 2 Fmax

SAMPLING RATE CONVERSION:

Sampling rate conversion is employed to generate a new sequence with a sampling rate higher or
lower than that of a given sequence. If x[n] is a sequence with a sampling rate of F Hz and it is
used to generate another sequence y[n] with desired sampling rate F’ Hz, then the sampling rate
alteration is given by,
F’/F = R
If R > 1, the process is called interpolation and results in a sequence with higher sampling rate. If
R< 1, the process is called decimation and results in a sequence with lower sampling rate.

DOWNSAMPLE AND DECIMATION:

Down sampling operation by an integer factor M (M>1) on a sequence x[n] consists of keeping
every Mth sample of x[n] and removing M-1 in between samples, generating an output sequence
y[n] according to the relation

y [n] = x[nM]
y [n] – sampling rate is 1/M that of x[n]

If we reduce the sampling rate, the resulting signal will be an aliased version of x[n]. To avoid
aliasing, the bandwidth of x[n] must be reduced to Fmax =Fx/2 π or max = π /M. The input
sequence is passed through LPF or an antialiasing filter before down sampling.

x [n] - y[n]
ANTIALIASING M

FILTER H (Z)

UPSAMPLE AND INTERPOLATION:

Up sampling by an integer factor L (L > 1) on a sequence x[n] will insert (L–1) equidistant samples
between an output sequence y[n] according to the relation

x[n/L], n = 0, 1, 2 ….
y[n] = 0, otherwise

The sampling rate of y[n] is L times that of x[n]. The unwanted images in the spectra of sampled
signal must be removed by a LPF called anti-imaging filter. The input sequence is passed through
an anti-imaging filter after up sampling.

x[n] ANTI IMAGING y[n]

L
FILTER H (Z)
SAMPLING RATE CONVERSION BY A RATIONAL FACTOR I/O:

We achieve this conversion, by first performing interpolation by the factor I and then
decimating the output of interpolator by the factor D, interpolation has to be performed before
decimation to obtain the new rational sampling rate.

x[n] ANTI IMAGING ANTI ALIASING DOWN y[n]

FILTER FILTER SAMPLER 
UPSAMPLER 

Exercise 1: Define Aliasing Phenomena in 3 lines

Exercise 2: A sinusoidal, s having Fs=10Khz and F= 5.5Khz, how many samples per wave cycle
of s will be there. Sampled signal will be aliased or not, explain in 2 lines.

IN- LAB

LIBRARY FUNCTIONS:

resample: Changes sampling rate by any rational factor.

y = resample (x,p,q) resamples the sequence in vector x at p/q times the original sampling rate,
using a polyphase filter implementation. p and q must be positive integers. The length of y is equal
to ceil (length(x)*p/q).

interp: Increases sampling rate by an integer factor (interpolation)

y = interp (x,r) increases the sampling rate of x by a factor of r. The interpolated vector y is r times
longer than the original input x. ‘interp’ performs low pass interpolation by inserting zeros into the
original sequence and then applying a special low pass filter.

upsample: Increases the sampling rate of the input signal

y = upsample(x,n) increases the sampling rate of x by inserting n-1 zeros between samples. The
upsampled y has length(x)*n samples

decimate: Decreases the sampling rate for a sequence (decimation).

y = decimate (x, r) reduces the sample rate of x by a factor r. The decimated vector y is r times
shorter in length than the input vector x. By default, decimate employs an eighth-order low pass
Chebyshev Type I filter. It filters the input sequence in both the forward and reverse directions to
remove all phase distortion, effectively doubling the filter order.
downsample: Decreases the sampling rate of the input signal
y = downsample(x,n) decreases the sampling rate of x by keeping every nth sample starting with
the first sample. The downsampled y has length(x)/n samples

ALGORITHM/PROCEDURE:
1. Generate a sinusoidal waveform
2. Using the appropriate library function for interpolation, decimation, up sampling,
3. down sampling and resampling, perform sampling rate conversion for the sinusoidal
waveform
4. Find the spectrum of all the signals and compare them in frequency domain.
5. Display the resultant waveforms

Source code :

clc;
clear all;
close all;
%continuous sinusoidal signal
a=input('Enter the amplitude:');
f=input('Enter the Frequency:');
t=-10:1:20;
x=a*sin(2*pi*f*t);
subplot(2,3,1);
plot(t,x);
xlabel('time');
ylabel('Amplitude');
title('Sinusoidal signal');
%decimating the signal
d=input('Enter the value by which the signal is to be decimated:');
y1=decimate(x,d);
subplot(2,3,2);
stem(y1);
xlabel('time');
ylabel('Amplitude');
title('Decimated signal');
%interpolating the signal
i=input('Enter the value by which the signal is to be interpolated:');
y2=interp(x,i);
subplot(2,3,3);
stem(y2);
xlabel('time');
ylabel('Amplitude');
title('Interpolated signal');
%resampling the signal
y3=resample(x,3,2);
subplot(2,3,4);
stem(y3);
xlabel('time');
ylabel('Amplitude');
title('Resampled signal');
%downsampling the signal
y4=downsample(x,2);
subplot(2,3,5);
stem(y4);
xlabel('time');
ylabel('Amplitude');
title('Downsampled signal');
%upsampling the signal
y5=upsample(x,3);
subplot(2,3,6);
stem(y5);
xlabel('time');
ylabel('Amplitude');
title('Upsampled signal');

Exercise 3: Decimate the Signal by 2 and Interpolate the Signal by the Factor of 2. Amplitude
should be 2 and F should be 0.05Hz.
Exercise 4: Plot (using stem) a sin wave, s having frequency of 3hz and sampling frequency of 60
Hz. How many samples /cycle will be plotted. Interpolate the s with L=3 and Decimate s with
M=2. Resample the s by the rate of 3/4.

POST-LAB

Exercise 5:

1. The sample rate used on compact discs is 44.1 kHz, while the sample rate used on digital
audio tape is 48 kHz. Create an input signal consisting of sum of sin waves sampled at
44.1 Khz. The sin waves should have frequencies of 2,4,8 Khz.
2. Change the sampling rate from 44.1 Khz to 48 Khz.
3. Append the following code to play both signals
 P44_1 = audioplayer(x,44100);
P48 = audioplayer(xnew,48000);
play(P44_1)
play(P48)

Exercise 6:

1. Change the sample rate of a speech sample from 7418 Hz to 8192 Hz. The speech signal
is a recording of a speaker saying "MATLAB®".
2. Load the speech sample by typing load mtlb
3. Use the rat command to determine the ratio and resample it to 8192 Hz.
4. Plot the 2 signals.
5. Play both signals by Sound command at their appropriate Fs.