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Dsssssssss

Here are the key points about this problem: 1. The continuous-time (CT) signal x(t) is a sinusoid with frequency F0 = 2 kHz. 2. The sampling frequency Fs = 50 kHz. 3. By the Nyquist sampling theorem, the minimum sampling frequency required to avoid aliasing is Fs ≥ 2F0. Here, 2F0 = 4 kHz and Fs = 50 kHz, which satisfies the Nyquist criterion. 4. Since the sampling frequency Fs = 50 kHz is greater than the Nyquist rate of 2F0 = 4 kHz, there will be no aliasing when sampling this signal. 5. The equivalent discrete-time (DT) signal obtained after

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0% found this document useful (0 votes)
56 views41 pages

Dsssssssss

Here are the key points about this problem: 1. The continuous-time (CT) signal x(t) is a sinusoid with frequency F0 = 2 kHz. 2. The sampling frequency Fs = 50 kHz. 3. By the Nyquist sampling theorem, the minimum sampling frequency required to avoid aliasing is Fs ≥ 2F0. Here, 2F0 = 4 kHz and Fs = 50 kHz, which satisfies the Nyquist criterion. 4. Since the sampling frequency Fs = 50 kHz is greater than the Nyquist rate of 2F0 = 4 kHz, there will be no aliasing when sampling this signal. 5. The equivalent discrete-time (DT) signal obtained after

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Baymax Tadashi
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© © All Rights Reserved
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Lab No.

Continuous Time Signal

6.1. Objective:
Sampling of continuous time signals and its reconstruction.

6.2. Apparatus/equipment/software:

MATLAB

6.3. Theory:

6.3.1. Sampling Theorem:


A continuous time signal can be represented in its samples and can be recovered
back when sampling frequency fs is greater than or equal to the twice the highest
frequency component of message signal. i.e.

fs≥2fm.

Proof:
Consider a continuous time signal x(t). The spectrum of x(t) is a band limited to fm Hz
i.e the spectrum of x(t) is zero for |ω|>ωm.
Sampling of input signal x(t) can be obtained by multiplying x(t) with an impulse
train δ(t) of period Ts. The output of multiplier is a discrete signal called sampled
signal which is represented with y(t) in the following diagrams:

Figure 6.1
Here, you can observe that the sampled signal takes the period of impulse. The process of
sampling can be explained by the following mathematical expression:
Sampled signal y(t)=x(t).δ(t). (1)
The trigonometric Fourier series representation of δ(t) is given by
δ(t)=a0+Σ∞n=1(an cosn ωst + bn sinn ωst). (2)

Where
a0=1Ts∫T2−T2δ(t)dt=1Tsδ(0)=1Ts
an=2Ts∫T2−T2δ(t)cosnωsdt=2T2δ(0)cosnωs0=2T
bn=2Ts∫T2−T2δ(t)sinnωstdt=2Tsδ(0)sinnωs0=0

Substitute above values in equation 2.


δ(t)=1Ts+Σ∞n=1(2Tscosnωst+0)
Substitute δ(t) in equation 1. y(t)=x(t).δ(t)
x(t)[1Ts+Σ∞n=1(2Tscosnωst)]
1Ts[x(t)+2Σ∞n=1(cosnωst)x(t)]
y(t)=1Ts[x(t)+2cosωst.x(t)+2cos2ωst.x(t)+2cos3ωst.x(t)…..] Take
Fourier transform on both sides.

Y(ω)=1Ts[X(ω)+X(ω−ωs)+X(ω+ωs)+X(ω−2ωs)+X(ω+2ωs)+…..]
Y(ω)=1TsΣ∞n=−∞X(ω−nωs)
Were
n=0,±1,±2,...

To reconstruct x(t), you must recover input signal spectrum X(ω) from sampled signal
spectrum Y(ω), which is possible when there is no overlapping between the cycles of
Y(ω).
Possibility of sampled frequency spectrum with different conditions is given by the
following diagrams:
Figure 6.2

6.4. Continuous-to-Discrete Conversion:


By using a Continuous-to-Discrete (C-to-D) converter, we can take continuous-time signals and
form a discrete-time signal. • There are devices called Analog-to-Digital converters (A-to-D) •
The books chooses to distinguish an C-to-D converter from an A-to-D converter by defining a
C-to-D as an ideal device while A-to-D converters are practical devices where real world
problems are evident.

6.5. Discrete-Time Signals:


A discrete-time signal is a sequence of numbers and carry no information about the timesequence.

6.6. Discrete-Time Sinusoidal Signals:


Since a Fourier series can be written for any continuous-time signal, let’s concentrate on
sinusoids • We define a normalized frequency for the discrete sinusoidal signal.

Since we can have different signals with the same, then there can be an infinite number of
continuous-time signal which yield the same discrete-time sinusoid!
6.4. Aliasing Effect:
The overlapped region in case of under sampling represents aliasing effect, which
can be removed by Considering fs >2fm

Aliasing is when a continuous-time sinusoid appears as a discrete-time sinusoid with


multiple frequencies. The sampling theorem establishes conditions that prevent aliasing so
that a continuous-time signal can be uniquely reconstructed from its samples. The sampling
theorem is very important in signal processing. It tells us how fast we have to sample a signal
with a given bandwidth to guarantee a unique correspondence between continuous- and
discrete-time signals.

This lesson “simplifies” the sampling theorem because it only uses the non-uniqueness of
discrete-time sinusoids and the mapping between continuous- and discrete-time frequencies.
The next two lessons take a more conventional Fourier transform based approach to
developing the sampling theorem.

How to prevent aliasing?


We can simply avoid aliasing by sampling the signal at a higher rate than the Nyquist rate
(Fs>Fm). Or, we can use anti-aliasing filters. These are special low-pass filters that are usually
found in the initial stages of any digital signal processing operation. The anti-aliasing filters
attenuate the unnecessary high-frequency components of a signal. They band-limit the input
signal by removing all frequencies higher than the signal frequencies. As a result, they help
preserve a lot of information that is needed and remove unnecessary information.

6.5. The Sampling Process in the Time Domain:

The purpose of this section is to study the relation in the time domain between a continuous time
signalxa(t) and the discrete-time signal x[n] generated by a periodic sampling of xa(t).
Task 6.1 Code:

Output:
Task 6.2 Code:

Output:
Task 6.3 Code:
Task 6.4 Code:
Output:
Task 6.4 Code:

Output:
Task 6.5
Sampling of a Sinusoidal Signal

In this project you will investigate the sampling of a continuous-time sinusoidal signal xa(t) at various
sampling rates. Since MATLAB cannot strictly generate a continuous-time signal, you will generate a
sequence {xa(nTH)} from xa(t) by sampling it at a very high rate, 1/TH, such that the samples are very
close to each other. A plot of xa(nTH) using the plot command will then look like a continuous-time
signal. Code:
Output:
Continuous-time signal

Questions and Answers

1. What is the frequency in Hz of the sinusoidal signal? What is the sampling period in second?

Ans. The frequency of the sinusoidal signal is 13KHz and the sampling period is 0.1sec.
Task 6.6

Run Program 6.5 for four other values of the sampling period with two lower and two higher than that
listed in Program P6.5. Comment on your results.
Code:
For T = 0.01

Output:
Task 6.7 Code:
For T = 0.05
Task 6.8

Output:

Code:
For T = 0.2
Output:
Task 6.10

Code:
For T = 0.5

Output:
Repeat Pervious Program by changing the frequency of the sinusoidal signal to 3 KHz and 7 KHz,
respectively. Is there any difference between the corresponding equivalent discrete-time signals and
the one generated in Question Q6.5? If not, why not? Code:

For f = 3
Task 6.12

Output:
Task 6.11 Code:

For f = 7

Output:
Task 6.12
Aliasing Effect in the Time Domain
In this exercise I will generate a continuous-time equivalent ya(t) of the discrete-time signal x[n]
generated in Program P6.5 to investigate the relation between the frequency of the sinusoidal signal
xa(t) and the sampling period. To generate the reconstructed signal ya(t) from x[n] , we pass x[n]
through an ideal low-pass filter that in turn can be implemented. Code:

Output:
2. Explain working of line 10 of program which reads:

ya = sinc((1/T)*t(:,ones(size(n))) - (1/T)*n(:,ones(size(t)))')*xs;
Ans. ya(t) is calculated from thje sampling period of the program.

Task 6.13

Repeat Pervious Program by changing the frequency of the sinusoidal signal to 3 KHz and 7 KHz,
respectively. Is there any difference between the corresponding equivalent discrete-time signals and the
one generated in Q 6.5? If not, why not?
Code:

For f = 3
Output:

Task 6.14

Repeat Pervious Program by changing the frequency of the sinusoidal signal to 3 KHz and 7 KHz,
respectively. Is there any difference between the corresponding equivalent discrete-time signals and the
one generated in Q 6.5? If not, why not? Code:
For f = 7
Output:

There is no changing in the signal

Aliasing Effect in the Frequency Domain

The relation between the continuous-time Fourier transform (CTFT) of an arbitrary bandlimited
continuous-time signal and the discrete-time Fourier transform (DTFT) of the discrete-time signal is
investigated next in this project. In order to convert a continuous-time signal xa(t) into an equivalent
discrete-time signal x[n], the former must be band-limited in the frequency domain (see R6.2). To
illustrate the effect of sampling in the frequency domain we choose an exponentially decaying
continuous-time signal with a CTFT that is approximately bandlimited. Task 6.15

Code:

Output:
There is an effect of aliasing in signal.

3. What is the continuous-time function xa(t) in Program P6.15? How is the CTFT of xa(t) being
computed?
Ans. The continuous-time function xa(t) in this program is 2*t*e-t and the CTFT is compute by using
MATLAB command freqs ().
Task 6.16

Repeat Program P6.15 by increasing the sampling period to 1.5. Is there any visible effect of aliasing?

Code:
There is a difference in the signal due to effect of aliasing.
Output:

Task 6.17
−πt2
Modify Program P6.3 for the case of xa(t) = e

Code:
The signal shape is completely changed.
Output:

Exercise 1
Consider the given CT signal: x(t) = sin (2 pi F0 t). Suppose that F0 = 2 kHz and Fs = 50 kHz.
Plot the signal x(n). What will be the discrete frequency fd of the signal x(n)? ii) Plot the signal
y(n) created by taking the even numbered samples of x(n). Is this a sinusoidal signal? Why? If
so, what is the frequency?

Code:
Exercise 2 Code:
For f=5000
Output:
Output:

Exercise 3
Consider the following CT signal: x(t) = sin (2 pi F0 t). The sampled version will be: x(n) =sin
(2 pi F0/Fs n), where n is a set of integers and sampling interval Ts=1/Fs. Plot the signal x(n) for
n = 0 to 99 for Fs = 5 kHz and F1 = 0.5, 2, 3 and 4.5 kHz. Explain the similarities and differences
among various plots. Also mention that whether aliasing occurs or not.

Code:
Output:
When the f1=2 Code:

Output:
When the f=3 Code:

Output:
When the f=4 Code:

Output:

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