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EEE-3217 - L5 - Basics of Sampling - I

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38 views28 pages

EEE-3217 - L5 - Basics of Sampling - I

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h01713717024
Copyright
© © All Rights Reserved
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Basics of Sampling_I

Digital Signal Processing I


Course ID: EEE 3217

Course Teacher: Prof. A K M Baki


Ahsanullah University of Science and Technology
Email: [email protected]
THE BASICS OF SAMPLING

Block diagram of a DSP system.

The lowpass/bandpass filter is required to remove unwanted


signals outside the bandwidth of interest and prevent aliasing.
2
Sampling Theorem

If a signal f(t) is sampled at regular intervals of time and at a rate higher


than twice the highest frequency, then the samples contain all the
information of the original signal.
The function f(t) may be reconstructed from these samples by the use of
a low pass filter.

According to the Nyquist theorem :


The sampling rate must be at least 2 times the highest frequency
contained in the signal.
Sampling Rate or Sampling Frequency : Number of samples per second

ts = Sampling interval
Sampling Frequency , fs=1/ts 3
DISCRETE TIME SAMPLING OF ANALOG SIGNALS

NYQUIST'S CRITERIA
 A signal with a bandwidth fa must be sampled at a rate fs > 2fa or
information about the signal will be lost
 Aliasing occurs whenever fs < 2fa
 The concept of aliasing is widely used in communications
applications such as direct IF-to-digital conversion

Simply stated, the Nyquist Criteria requires that the sampling frequency
be at least twice the signal bandwidth, or information about the signal will
be lost. If the sampling frequency is less than twice the analog signal
bandwidth, a phenomena known as aliasing will occur.
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ALIASING IN THE TIME DOMAIN

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ALIASING IN THE TIME DOMAIN

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ALIASING IN THE TIME DOMAIN
Example of aliasing: Frequency of the analog signal : 90 Hz , Sampling Frequency: 100 Hz
Frequency
Let of the aliased signal : 10 Hz (180 degree phase shift)

Exmple

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ALIASING IN THE TIME DOMAIN

Frequency of the analog signal : 90 Hz


Sampling Frequency: 100 Hz
Frequency of the aliased signal : 10 Hz (180 degree phase shift)
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ALIASING IN THE TIME DOMAIN

MATLAB CODE FOR THE EXAMPLE SHOWN ON LAST SLIDE:

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ALIASING IN THE TIME DOMAIN
Discussions:

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ALIASING IN THE TIME DOMAIN

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ALIASING IN THE TIME DOMAIN

ANALOG SIGNAL fa SAMPLED @ fs USING IDEAL SAMPLER


HAS IMAGES (ALIASES) AT |±Kfs ±fa|, K = 1, 2, 3, . . .

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ALIASING IN THE TIME DOMAIN

NYQUIST BANDWIDTH

The Nyquist bandwidth is defined to be the frequency spectrum from DC to


fs/2. The frequency spectrum is divided into an infinite number of Nyquist
zones, each having a width equal to 0.5fs as shown in Fig 3.4 . In practice,
the ideal sampler is replaced by an ADC followed by an FFT processor. The
FFT processor only provides an output from DC to fs/2, i.e., the signals or
aliases which appear in the first Nyquist zone.

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ALIASING IN THE TIME DOMAIN

Conversion of analog frequency into digital frequency during sampling:

When the continuous signal's frequency is above the Nyquist rate (fs/2),
aliasing changes the frequency into something that can be represented in
the sampled data.

As shown by the zigzagging line in Fig. 3-4 (next slide), every continuous
frequency above the Nyquist rate has a corresponding digital frequency
between zero and one-half the sampling rate. If there happens to be a
sinusoid already at this lower frequency, the aliased signal will add to it,
resulting in a loss of information.
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ALIASING IN THE TIME DOMAIN

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ALIASING IN THE TIME DOMAIN

Aliasing can also change the phase. Only two phase shifts are possible:
0 degree (no phase shift) and 180 degree (inversion). The zero phase
shift occurs for analog frequencies of 0 to 0.5, 1.0 to 1.5, 2.0 to 2.5,
etc. An inverted phase occurs for analog frequencies of 0.5 to 1.0, 1.5
to 2.0, 2.5 to 3.0, and so on.

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ALIASING IN THE TIME DOMAIN
Baseband Antialiasing Filters
Baseband sampling implies that the signal to be sampled lies in the
first Nyquist zone. It is important to note that with no input filtering at
the input of the ideal sampler, any frequency component (either signal
or noise) that falls outside the Nyquist bandwidth in any Nyquist zone
will be aliased back into the first Nyquist zone. For this reason, an
antialiasing filter is used in almost all sampling ADC applications to
remove these unwanted signals.

Properly specifying the antialiasing filter is important. The first step is


to know the characteristics of the signal being sampled. Assume that
the highest frequency of interest is fa. The antialiasing filter passes
signals from DC to fa while attenuating signals above fa.
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ALIASING IN THE TIME DOMAIN

Time domain signal x(t) Original signal spectrum X(f)

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ALIASING IN THE TIME DOMAIN
Pulse train p(t)

Time domain signal x(t) Sampled signal xs(t)

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ALIASING IN THE TIME DOMAIN

Sampled signal xs(t)

1 1 1
𝑋𝑠 𝑓 = … + 𝑋 𝑓 + 𝑓𝑠 + 𝑋 𝑓 + 𝑋 𝑓 − 𝑓𝑠 + ⋯ .
𝑇 𝑇 𝑇

Sampled signal spectrum for fs>2B 20


ALIASING IN THE TIME DOMAIN

Sampled signal xs(t)

𝑓𝑠 − 𝑓𝑚𝑎𝑥 ≥ 𝑓𝑚𝑎𝑥

Sampled signal spectrum for fs=2B 21


ALIASING IN THE TIME DOMAIN

Sampled signal xs(t)

Sampled signal spectrum for fs<2B 22


Digital-to-Analog Conversion

The simplest method for digital-to-analog conversion is to pull the samples


from memory and convert them into an impulse train. This is illustrated in
Fig. a on next slide, with the corresponding frequency spectrum in (b).

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Digital-to-Analog Conversion

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Digital-to-Analog Conversion

It is difficult to generate the required narrow pulses in electronics. To get


around this, nearly all DACs operate by holding the last value until
another sample is received. This is called a zeroth-order hold, the DAC
equivalent of the sample-and-hold used during ADC.

The analog filter used to convert the zeroth-order hold signal, (c), into the
reconstructed signal, (f) , shown on next slide.

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Digital-to-Analog Conversion

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Digital-to-Analog Conversion

Why an anti-imaging LPF needed at the DAC output?


What should be the highest cut-off frequency of the filter?
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Suggested reading:
Steven W. Smith, “The Scientist and Engineer's Guide to Digital Signal
Processing”, California Technical Publishing San Diego, California.
Li Tan.” DIGITAL SIGNAL PROCESSING”, Academic Press, Elsevier, 2011

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