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This document summarizes key concepts from a lecture on digital signal processing: 1) It discusses analog to digital conversion, where an analog signal is sampled and converted to digital values by an ADC. 2) It explains the sampling theorem - for perfect reconstruction, the sampling rate must be at least twice the highest frequency component of the analog signal. 3) It describes how the spectrum of a sampled signal consists of the original spectrum plus replicas, and discusses the conditions for applying a reconstruction filter to recover the original signal spectrum.

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30 views

Adsp - Lec 2

This document summarizes key concepts from a lecture on digital signal processing: 1) It discusses analog to digital conversion, where an analog signal is sampled and converted to digital values by an ADC. 2) It explains the sampling theorem - for perfect reconstruction, the sampling rate must be at least twice the highest frequency component of the analog signal. 3) It describes how the spectrum of a sampled signal consists of the original spectrum plus replicas, and discusses the conditions for applying a reconstruction filter to recover the original signal spectrum.

Uploaded by

amnapa
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PPT, PDF, TXT or read online on Scribd
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SP503

Advanced Digital Signal Processing

Course Instructor: Engr. Touseef A. Rajput

Week # 2
25-10-2010
Lecture 01 Review
• An analog signal is continuous in both time and amplitude. Analog signals in the real
world include current, voltage, temperature, pressure, light intensity, and so on.

• The digital signal is the digital values converted from the analog signal at the specified
time instants.

• Analog-to-digital signal conversion requires an ADC unit (hardware) and a low-pass filter
attached ahead of the ADC unit to block the high-frequency components that ADC cannot
handle.

• The digital signal can be manipulated using arithmetic. The manipulations may include
digital filtering, calculation of signal frequency content, and so on.

• The digital signal can be converted back to an analog signal by sending the digital values
to DAC to produce the corresponding voltage levels and applying a smooth filter
(reconstruction filter) to the DAC voltage steps.

• Digital signal processing finds many applications in the areas of digital speech and audio,
digital and cellular telephones, automobile controls, communications, biomedical imaging,
image/video processing, and multimedia.

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Signal Sampling and Quantization
• Below is an analog signal (solid line) defined at every point over the time
axis and amplitude axis. Hence, the analog signal contains an infinite
number of points.

Why Sampling?
• It is impossible to digitize an infinite number of points.
• Furthermore, the infinite points are not appropriate to be processed by the digital signal (DS) processor or computer, since they require infinite amount
of memory and infinite amount of processing power for computations. Sampling can solve such a problem by taking samples at the fixed time interval.

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• Each sample maintains its voltage level during the sampling interval T to give
the ADC enough time to convert it.
• This process is called ‘sample and hold’.
• Since there exists one amplitude level for each sampling interval, we can
sketch each sample amplitude level at its corresponding sampling time instant.
• For a given sampling interval T, which is defined as the time span between
two sample points, the sampling rate is therefore given by
fs = 1 / T samples per second (Hz).
• For example, if a sampling period is T = 125 microseconds, the sampling rate
is determined as fs = 1 / (125 sec) = 8,000 samples per second (Hz).

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• Next, we have to ensure that samples are collected at a rate high enough
that the original analog signal can be reconstructed or recovered later.

• If an analog signal is not appropriately sampled, aliasing will occur, which


causes unwanted signals in the desired frequency band.

• The sampling theorem guarantees that an analog signal can be in theory


perfectly recovered as long as the sampling rate is at least twice as large as
the highest-frequency component of the analog signal to be sampled. The
condition is described as fs >= fmax, where fmax is the maximum-frequency
component of the analog signal to be sampled.

For example
• The minimum sampling rate to sample a speech signal (which contains
frequencies up to 4 kHz) is at least 8 kHz, or 8,000 samples per second.

• To sample an audio signal, possessing frequencies up to 20 kHz, at least


40,000 samples per second, or 40 kHz, of the audio signal are required.

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We call the 10-Hz sine wave the aliasing noise in this case, since the sampled amplitudes actually come from sampling the 90-Hz sine wave

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Now let us develop the sampling theorem in frequency domain:
• Consider a sampled signal xs(t) obtained by sampling the continuous signal
x(t) at a sampling rate of fs samples per second.
• Mathematically, this process can be written as the product of the continuous
signal and the sampling pulses (pulse train):
xs(t) = x(t)p(t) (1)

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• From spectral analysis, the original spectrum X( f ) and the sampled signal
spectrum Xs( f ) in terms of Hz are related as

(2)

where X( f ) is assumed to be the original baseband spectrum, while Xs( f ) is


its sampled signal spectrum, consisting of the original baseband spectrum X( f )
and its replicas X( f ± nfs).

• Expanding Equation (2) leads to the sampled signal spectrum in Equation (3):

(3)

Equation (3) indicates that the sampled signal spectrum is the sum of the
scaled original spectrum and copies of its shifted versions, called replicas.

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From the above figure, it is clear that the sampled signal spectrum consists of the scaled baseband
spectrum centered at the origin and its replicas centered at the frequencies of ±nfs (multiples of the
sampling rate) for each of n =1,2,3, . . . .

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• If applying a low-pass reconstruction filter to obtain exact reconstruction
of the original signal spectrum, the following condition must be satisfied:

(4)

Solving Equation (4) gives


(5)

In terms of frequency in radians per second, Equation (5) is equivalent to


(6)

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which is a Fourier series expansion for a continuous periodic signal in terms
of the exponential form. We can identify the Fourier series coefficients as

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Signal Reconstruction
• Two simplified steps are involved:
– First, the digitally processed data y(n) are converted to the ideal impulse train
ys(t), in which each impulse has its amplitude proportional to digital output y(n),
and two consecutive impulses are separated by a sampling period of T
– second, the analog reconstruction filter is applied to the ideally recovered
sampled signal ys(t) to obtain the recovered analog signal.

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b. Based on the spectrum in (a), the sampling
theorem condition is satisfied; hence, we can recover
the original spectrum using a reconstruction low-pass
filter. The recovered spectrum is shown as:

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a. The spectrum for the sampled signal is sketched b. Since the maximum frequency of the analog
below: signal is larger than that of the Nyquist frequency,
the sampling theorem condition is violated.
The recovered spectrum is shown below, where
we see that aliasing noise occurs at 3 kHz.

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