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DSP Unit-I Part 2 Updated 20.7.2020

The document discusses the process of sampling and analog-to-digital conversion. It explains that sampling is the conversion of a continuous-time signal into a discrete-time signal by taking samples at regular time intervals. For unique reconstruction, the sampling frequency must be at least twice the maximum frequency of the sampled signal according to the Nyquist sampling theorem. It also describes the processes of quantization, coding, and the sources of quantization error.

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85 views57 pages

DSP Unit-I Part 2 Updated 20.7.2020

The document discusses the process of sampling and analog-to-digital conversion. It explains that sampling is the conversion of a continuous-time signal into a discrete-time signal by taking samples at regular time intervals. For unique reconstruction, the sampling frequency must be at least twice the maximum frequency of the sampled signal according to the Nyquist sampling theorem. It also describes the processes of quantization, coding, and the sources of quantization error.

Uploaded by

siva
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Sampling Process

• Most signals of practical interest, such as speech,


biological signals, seismic signals, radar signals,
sonar signals, and various communications
signals such as audio and video signals, are
analog.
• To process analog signals by digital means,
• It is first necessary to convert them into digital
form, that is, to convert them to a sequence of
numbers having finite precision. This procedure is
called analog-to-digital (A/D) conversion, and the
corresponding devices are called A/D converters
(ADCs).
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Cont…
• Sampling This is the conversion of a
continuous-time signal into a discrete-time
signal obtained by taking “samples” of the
continuous-time signal at discrete time
instants. Thus, if xa(t) is the input to the
sampler, the output is xa(nT ) = x(n), where T is
called the sampling interval.

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Cont…
• The sampling is the process of conversion of a
continuous time signal into a discrete time signal.
• The sampling is performed by taking samples of
continuous time signal at definite intervals of time.
• The time interval between two successive samples will
be same and such type of sampling is called periodic
or uniform sampling.
• The time interval between successive samples is called
sampling time (or sampling period or sampling
interval), and it is denoted by “T.” The unit of sampling
period is second (s).
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Cont…
• The inverse of sampling period is called sampling
frequency (or sampling rate), and it is denoted by Fs. The
unit of sampling frequency is hertz (Hz). ( kHz or MHz)
• Let, xa(t) = Analog / Continuous time signal.
x(n) = Discrete time signal obtained by sampling xa(t).
Mathematically, the relation between x(n) and xa(t) can be
expressed as,

where, T = Sampling period or interval in seconds Fs = 1 / T


= Sampling rate or sampling frequency in hertz
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Cont…
• The relation between frequency of analog and
discrete time signal is, f = F/Fs
• Where, f – continuous time signal frequency
F – discrete time signal frequency
Fs – Sampling rate or frequency
• The range of frequency of discrete time signal is,
1 1 Fs F
− f − F  s
2 2 2 2
• Infinite number of high frequency continuous
time signals will be represented by a single
discrete time signal. Such signals are called alias.
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Cont…
• Aliasing The phenomenon of high-frequency
component getting the identity of low-frequency
component during sampling is called aliasing
• Sampling an analog signal with frequency F by
choosing a sampling frequency Fs such that Fs/2 >
F will not result in alias.
• But sampling frequency is selected such that Fs/2
< F that the frequency above Fs/2 will have alias
with frequency below Fs/2.
• Folding frequency Hence the point of reflection is
Fs/2, and the frequency Fs/2 is called folding
frequency
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Cont…
• Let Fmax is the maximum value of analog
signal
F
• F =
max
s

2
Therefore Fs = 2Fmax
• To avoid aliasing Fs  Fmax
• When sampling frequency Fs is equal to 2Fmax
the sampling rate is called Nyquist rate.

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Sampling Theorem
• A band limited continuous time signal with
highest frequency (bandwidth) Fm hertz can
be uniquely recovered from its samples
provided that the sampling rate Fs is greater
than or equal to 2Fm samples per second

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Cont…

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ADC

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Cont…

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Cont…

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Cont…
Normalized frequency

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Cont..
• Quantization This is the conversion of a discrete-
time continuous-valued signal into a discrete-
time, discrete-valued (digital) signal.
• The value of each signal sample is represented by
a value selected from a finite set of possible
values.
• Quantization error The difference between the
unquantized sample x(n) and the quantized
output xq(n) is called the quantization error or
noise

• Coding In the coding process, each discrete value


xq(n) is represented by a b-bit binary sequence.
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Cont…
• To represent a one significant digit by eliminating the
excess digits, so can either simply discard them
(truncation) or discard them by rounding the resulting
number (rounding).

• The values allowed in the digital signal are called the


quantization levels,

• The distance between two successive quantization


levels is called the quantization step size or resolution
().

15
Cont…
• The rounding quantizer assigns each sample of x(n) to
the nearest quantization level. The quantization error
eq(n) in rounding is limited to the range of −  /2 to  /2,

• If xmin and xmax represent the minimum and maximum


values of x(n) and L is the number of quantization levels

• The dynamic range of the signal as xmax − xmin .

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Cont…

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Problem
Prob1: Consider the analog signal represented by
xa(t) = 3 cos 100πt
(a) Determine the minimum sampling rate required to avoid
aliasing.
(b) Suppose that the signal is sampled at the rate Fs = 200
Hz. What is the discrete-time signal obtained after
sampling?
(c) Suppose that the signal is sampled at the rate Fs = 75
Hz. What is the discrete-time signal obtained after
sampling

18
Cont…
Solution
w k t  = 2F, Hence F =  /2  = 100  /2 
(a) The frequency of the analog signal is f = 50 Hz.
Hence the minimum sampling rate required to
avoid aliasing is Fs = 100 Hz. (i.e. Fs  2Fm )
(b) If the signal is sampled at Fs = 200 Hz, the
discrete-time signal is

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Cont…

• If the signal is sampled at Fs = 75 Hz, the


discrete-time signal is
put t = n/Fs

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Cont…
Prob.2 The analog signal is represented by

(a) What is the Nyquist rate for this signal?


(b) Assume that sample this signal using a sampling rate
Fs = 5000 samples/sec.
What is the discrete-time signal obtained after
sampling?
Solution
The frequencies existing in the analog signal are
F1 = 1 kHz, F2 = 3 kHz, F3 = 6 kHz
Thus Fmax = 6 kHz, and according to the sampling
theorem, Fs > 2Fmax = 12 kHz
The Nyquist rate is FN =12 kHz 21
Cont…
(b) Since we have chosen Fs = 5 kHz, the folding
frequency is Fs /2 = 2.5 kHz and this is the maximum
frequency that can be represented uniquely by the
sampled signal

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Problem

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Problem Cont…

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Problem
Determine the following systems are time invariant or variant
(i) y(n) = x(n) + x (n-1) (ii) y(n) = x (-n) (iii) y(n) = x (2n)

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Cont…

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Cont…

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Problem

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Cont…

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Cont…

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Cont…

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Cont…

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Cont…

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Cont…

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Cont…

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Cont…

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Cont…

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Cont…

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Quantization Noise Power

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Quantization Noise ratio (SQNR)

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Coding of Quantization Sample

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Digital Signal Processing Methods

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Convolution Integral
The convolution of two functions h(t) and x(t), denoted by h(t)*x(t), is
defined by:

The convolution performs integration on the product of the first function


and a shifted and reflected version of the second function.
Properties of the Convolution Integral
These properties are important in predicting the behavior of various
combinations of LTI systems
1. Commutative: h(t) *x(t) = x(t) *h(t)
2. Associative: h(t) *[x(t) *v(t)] = [h(t) *x(t)] * v(t)
3. Distributive: h(t) *[x(t) + v(t)] = h(t) *x(t) + h(t) * v(t).

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