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Contents

Objectives ..................................................................................................................................................... 1

List of Equipment and Parts ......................................................................................................................... 1

1 – Prelab - Fourier transform of a sampled filtered signal. ....................................................................... 2

2 - Simulink for Lab ....................................................................................................................................... 3

3 - Lab - Discrete-Time Interpolation ........................................................................................................... 4

3.1. Discrete Time Interpolator – interpolate factor=1........................................................................... 5

3.2. Discrete Time Interpolator – up sample by a factor of 4 ................................................................. 5

Objectives

1. The student will be able to understand the effects of analog to digital conversion (ADC) and

digital-to-analog (DAC) conversion on signals.

PC with Matlab

1

1 – Prelab - Fourier transform of a sampled filtered signal.

A signal 𝑥(𝑡) = 5 ∗ cos(2 ∗ 𝜋 ∗ 1000 ∗ 𝑡) is multiplied by an impulse train

∞

𝑛=−∞

1 1

the value of 𝑇𝑠 = = .

𝑓𝑠 10000

Let the product be xs(t).

The signal xs(t) is applied to an ideal low pass filter with cutoff frequency 5000 Hz.

Let the output of the filter be y(t).

1b. Find the Fourier transform X(f) of x(t) and plot X(f).

1c. Plot p(t).

1d. Find the exponential Fourier coefficient of p(t). Represent p(t) by its exponential Fourier series.

1e. Find the Fourier transform P(f) of p(t) and plot P(f).

1f. Plot xs(t).

1g. Find the Fourier transform Xs(f) of xs(t) and plot Xs(f).

1h. Plot y(t).

1i. Find the Fourier transform Y(f) of y(t) and plot Y(f).

2

2 - Simulink for Lab

In this lab, we will use Simulink to simulate the effects of the sampling and reconstruction processes.

Simulink treats all signals as continuous-time signals. This means that “sampled” signals are really just

continuous-time signals that contain a series of finite-width pulses. The height of each of these pulses is

the amplitude of the input signal at the beginning of the pulse. In other words, both the sampling action

and the zero-order-hold reconstruction are done at the same time; the discrete-time signal itself is

never generated. This means that the impulse-generator block is really a “pulse-generator”, or zero-

order-hold device. Remember that, in Simulink, frequency spectra are computed on continuous-time

signals. This is why many aliased components will appear in the spectra.

3

3 - Lab - Discrete-Time Interpolation

In the previous experiments, we saw that the frequency content of a signal must be limited to half the

sampling rate in order to avoid aliasing effects in the reconstructed signal.

However, reconstruction can be difficult if the sampling rate is just above the Nyquist frequency.

Reconstruction is much easier for a higher sampling rate because the sampled signal will better “track”

the original analog signal.

Accurately reconstructing a signal sampled just above the Nyquist rate requires an analog output filter

with a very sharp cutoff. Such filters are difficult and expensive to manufacture.

Alternatively, a higher sampling rate allows the use analog output filters that have a slow roll-off.

These filters are much less expensive.

However, a high sampling rate is not practical in most applications, as it results in unnecessary samples

and excessive storage requirements.

A solution is to interpolate the digital signal to create new (artificial) samples between existing samples.

This may be done by first up sampling the digital representation, and then filtering out unwanted

components using a discrete-time filter.

This discrete-time filter serves the same purpose as an analog filter with a sharp cutoff, but it is

generally simpler and more cost effective to implement.

Up sampling a signal by a factor of L is simply the process of inserting L−1 zeros in between each sample.

The frequency domain relationship between a signal x(n) and its up sampled version z(n) can be shown

to be the following: 𝑍(𝑒 𝑗𝑤 ) = 𝑋(𝑒 𝑗𝑤 ) (6)

Therefore the DTFT of z(n) is simply 𝑋(𝑒 𝑗𝑤 ) compressed in frequency by a factor of L. Since 𝑋(𝑒 𝑗𝑤 ) has

a period of 2π, 𝑍(𝑒 𝑗𝑤 ) will have a period of 2π /L.

All of the original information of x(n) will be contained in the interval [− π /L, π /L] of 𝑍(𝑒 𝑗𝑤 ), and the

new aliases that are created in the interval [− π , π ] are the unwanted components that need to be

filtered out.

In the time domain, this filtering has the effect of changing the inserted zeros into artificial samples of

x(n), commonly known as interpolated samples.

Figure 6 shows such a system for performing discrete-time interpolation.

The interpolating system contains three main components:

b. A discrete-time low pass filter to remove aliased signal components in the interpolated signal,

c. A gain block to correct the magnitude of the final signal.

Notice that signal a is the input discrete-time signal while c is the final interpolated discrete-time signal.

4

3.1. Discrete Time Interpolator – interpolate factor=1

The components of the system are initially set to interpolate by a factor of 1.

This means that the input and output signals will be the same except for a delay.

1. Open the experiment by double clicking on the icon labeled Discrete Time Interpolator.

2. Run this model with the initial settings, and observe the signals on the Scope.

3. Submit your plot of signal c and its frequency spectrum.

4. Circle the aliased components in your plot.

Simulink represents any discrete-time signal by holding the sample value over the discrete time period.

This representation is equivalent to a sample-and-hold reconstruction of the underlying discrete-time

signal.

Therefore, a continuous-time Spectrum Analyzer may be used to view the frequency content of the

output signal c.

The Zero-Order Hold at the Gain output is required as a buffer for the Spectrum Analyzer in order to set

its internal sampling period.

The lowest frequency component in the spectrum corresponds to the frequency content of the original

input signal, while the higher frequencies are aliased components resulting from the sample-and-hold

reconstruction.

Notice that the aliased components of signal c appear at multiples of the sampling frequency of 1 Hz.

Print the output of the Spectrum Analyzer.

1. Modify the system to up sample by a factor of 4 by setting this parameter in the Up sampler.

2. Set the Sample time of the DT filter to 0.25.

This effectively increases the sampling frequency of the system to 4 Hz.

3. Run the simulation again and observe the behavior of the system.

Notice that zeros have been inserted between samples of the input signal.

4. Print the output of the Spectrum Analyzer.

5. Submit your plot of signal c and its frequency spectrum.

6. On your frequency plot, circle the first aliased component and label the value of its center

frequency.

7. Comment on the shape of the envelope of the spectrum

Some of these spectral components lie between the frequency of the original signal and the new

sampling frequency, 4 Hz.

these aliases are due to the zeros that are inserted by the up sampler.

Notice in the previous Scope output that the process of up sampling causes a decrease in the energy of

the sample-and-hold representation by a factor of 4.

This is the reason for using the Gain block.

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