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Lab7 Sampling

This document outlines a lab focused on sampling in digital signal processing, covering the sampling theorem, Nyquist rate, aliasing, and signal reconstruction. It includes objectives for understanding these concepts through practical experiments using MATLAB, where students will sample and reconstruct a sinusoidal signal at various frequencies. Additionally, it poses related questions about the effects of undersampling and oversampling on signal reconstruction.

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6 views2 pages

Lab7 Sampling

This document outlines a lab focused on sampling in digital signal processing, covering the sampling theorem, Nyquist rate, aliasing, and signal reconstruction. It includes objectives for understanding these concepts through practical experiments using MATLAB, where students will sample and reconstruct a sinusoidal signal at various frequencies. Additionally, it poses related questions about the effects of undersampling and oversampling on signal reconstruction.

Uploaded by

lijian041119
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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SIGNALS AND SYSTEM LAB #7

TOPIC: Sampling

Shien-Ming Wu School of Intelligent Engineering, SCUT


Spring 2025

SYLLABUS
• The sampling theorem and its significance in digital signal processing.
• Definition of the Nyquist rate.
• Effect of undersampling - aliasing.
• Reconstruct the original continuous-time signals through interpolation.
• Visualization of sampling, aliasing, and reconstruction processes

OBJECTIVES
• Understand the fundamental concepts of the sampling theorem, including the
Nyquist rate and aliasing.
• Learn how to sample continuous-time signals at different sampling rates and
observe the effects of sampling.
• Explore methods for reconstructing continuous-time signals from their samples.
• Use MATLAB to implement sampling and reconstruction processes and visualize
the results.

THEORY
1. Sampling theorem
The sampling theorem, also known as the Nyquist-Shannon sampling theorem, states
that a continuous-time signal 𝑥(𝑡) with a maximum frequency 𝜔𝑀 (band-limited
signal) can be perfectly reconstructed from its samples 𝑥[𝑛] = 𝑥(𝑛𝑇) if the sampling
frequency 𝜔𝑠 is greater than twice the maximum frequency of the signal, i.e., 𝜔𝑠 >
2𝜔𝑀 . The frequency 2𝜔𝑀 is called the Nyquist rate, and 𝜔𝑀 is the Nyquist frequency.
2. Aliasing
If the sampling frequency 𝜔𝑠 < 2𝜔𝑀 , aliasing occurs. Aliasing is the phenomenon
where different frequencies in the continuous-time signal are mapped to the same
frequency in the discrete-time domain, causing distortion in the reconstructed signal.
3. Reconstruction
To reconstruct a continuous-time signal 𝑥(𝑡) from its samples 𝑥[𝑛], an ideal lowpass
filter with a cutoff frequency 𝜔𝑐 = 𝜔𝑠 /2 can be used. The reconstructed signal can be
obtained through interpolation
+∞
𝑥𝑟 (𝑡) = ∑ 𝑥(𝑛𝑇)ℎ(𝑡 − 𝑛𝑇),
𝑛=−∞
where
𝜔𝑐 𝑇 sin(𝜔𝑐 𝑡)
ℎ(𝑡) = .
𝜋𝜔𝑐 𝑡
In practice, this interpolating function can be approximated by the zero-order hold
interpolating function, linear interpolating function and others.

EXPERIMENT 1:
Consider the following continuous-time sinusoidal signal
𝜋 𝜋
𝑥(𝑡) = cos (2𝜋 ⋅ 50𝑡 + ) + cos (2𝜋 ⋅ 100𝑡 + ).
4 2

Sample the signal 𝑥(𝑡) with sampling frequency 𝑓𝑠 = 150Hz, 200Hz, 250Hz and
300Hz . Then reconstruct the original signal based on the samples with different
sampling frequencies.
Plot the original signal 𝑥(𝑡) , the sampled version signal 𝑥𝑝 (𝑡) , and the
reconstructed signal 𝑥𝑟 (𝑡) for each sampling frequency. Compare the original signal
and the reconstructed signal.

EXPERIMENT 2:
Consider the same signal 𝑥(𝑡) in EXPERIMENT 1. Sample the signal 𝑥(𝑡)
with sampling frequency 𝑓𝑠 = 600Hz, 800Hz and 1000Hz Reconstruct the original
signal 𝑥(𝑡) with the samples using zero-order hold interpolation and linear
interpolation. Show the interpolating functions for zero-order hold interpolation and
linear interpolation. Plot the original signal 𝑥(𝑡), the sampled version signal 𝑥𝑝 (𝑡),
and the reconstructed signal 𝑥𝑟 (𝑡). Compare the results with EXPERIMENT 1.

RELATED QUESTIONS
1. When the sampling frequency 𝜔𝑠 = 2𝜔𝑀 , can we still exactly reconstruct the
original signal with the ideal lowpass filter?
2. What is the effect or consequence of undersampling (𝜔𝑠 < 2𝜔𝑀 )?
3. What is the effect or consequence of oversampling (𝜔𝑠 > 2𝜔𝑀 )?

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