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Digital Signal Processing Overview

The document covers key concepts in Digital Signal Processing (DSP), including the sampling theorem, methods of sampling, quantization, and encoding. It explains how continuous-time analog signals are converted into discrete-time signals and the importance of choosing an appropriate sampling rate to avoid aliasing. Additionally, it discusses the process of quantization and encoding, which involves representing discrete values with binary codes.

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0% found this document useful (0 votes)
13 views24 pages

Digital Signal Processing Overview

The document covers key concepts in Digital Signal Processing (DSP), including the sampling theorem, methods of sampling, quantization, and encoding. It explains how continuous-time analog signals are converted into discrete-time signals and the importance of choosing an appropriate sampling rate to avoid aliasing. Additionally, it discusses the process of quantization and encoding, which involves representing discrete values with binary codes.

Uploaded by

sherifamr922
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd

Digital Signal Processing(DSP)

Dr. Shiref Aboelnour Abdalla


𝑛
Find 𝑦 𝑛 = 𝑥 3𝑛 𝑎𝑛𝑑 𝑦 𝑛 = 𝑥
3
where x(n)= {2,-1,3,5,7}
Compute the convolution 𝑦(𝑛) of the
signal

∝𝑛 , −3 ≤ 𝑛 ≤ 5
𝑥 𝑛 = ቊ
0, 𝑒𝑙𝑠𝑤ℎ𝑒𝑟𝑒
1, 0 ≤ 𝑛 ≤ 4
ℎ 𝑛 = ቊ
0, 𝑒𝑙𝑠𝑤ℎ𝑒𝑟𝑒
LECTURE(4)
SAMPLING THEOREM
 What will be covered today?
 What is a Sampling?
 Sampling Theorem
 Methods of Sampling
 Significance of Sampling Rate
 Anti-aliasing Filter
 What is a quantization?
 Quantization Error
 What is a coding?
INTRODUCTION

 A digital signal is superior to an analog signal because it is more

robust to noise and can easily be recovered, corrected and amplified.

 An analog signal: amplitude can take any value over a continuous

range.

 Digital signals: amplitude can take only discrete and finite values.
Analog to digital conversion
Analog to digital conversion
SAMPLING PROCESSS.
➢ The sampling process is a basic operation in the digital communication.
➢ Sampling is the processes of converting continuous-time analog signal,
S(t), into a discrete-time signal by taking the "samples" at discrete-time
intervals t = nT𝑠 , where T𝑠 , is the sampling period.
➢ It is necessary to choose the sampling rate properly, so the sequence of
samples uniquely defines the original analog signal.
SAMPLING THEOREM
 The sampling theorem states that a band-limited signal x(t) with a
bandwidth B (𝑓max is the highest frequency) can be reconstructed from
1
its sample values if the sampling rate (frequency) f𝑠 = , is greater than
𝑇𝑠
or equal to twice the max frequency of x(t)

𝑤𝑠 ≥ 2 𝐵 𝑓𝑠 ≥ 2𝑓𝑚𝑎𝑥

 The minimum sampling rate 𝑓𝑠 for an analog band-limited allowed for


reconstruction without error is called the Nyquist frequency and the
Nyquist Rate.
𝑤𝑠 = 2 𝐵 𝑓𝑠 = 2𝑓𝑚𝑖𝑛
Note that if 𝑤𝑠 ≥ 2 B, the replicas of X(e𝑗𝑤 ) do not overlap in the frequency
domain. We can recover the original signal exactly
RECOVERING THE ANALOG SIGNAL
➢ One way of recovering the original signal from sampled signal X.(f) is to pass it
through a Low Pass Filter (LPF) as shown below

➢ If 𝑓𝑠 > 2B then we recover x(t) exactly


➢ Else we run into some problems and signal is not fully
recovered
SIGNIFICANCE OF SAMPLING RATE
• If the waveform is under sampled (i.e. fs < 2B) then there will be spectral overlap in the
sampled signal

• The signal at the output of the filter will be different from the
original signal spectrum

This is the outcome of aliasing!

• This implies that whenever the sampling condition is not


met, an irreversible overlap of the spectral replicas is
produced
SAMPLING METHODS
There are 3 sampling methods:

• Ideal - an impulse at each sampling instant

• Natural - a pulse of short width with varying amplitude

• Flattop - sample and hold, like natural but with single amplitude value
QUANTIZATION
➢ The sampling results is a series of pulses of varying amplitude values ranging
between two limits: a min and a max.
➢ This is achieved by dividing the distance between min and max into L zones, each of
height ∆

(𝑚𝑎𝑥 − 𝑚𝑖𝑛)
∆=
𝑙
QUANTIZATION
➢ Assume we have a voltage signal with
amplitudes 𝑉𝑚𝑖𝑛 = −4𝑉 𝑎𝑛𝑑 𝑉𝑚𝑎𝑥 = 4
➢ We will use L=8 quantization levels.
4−−4
➢ Zone width ∆= =1
8

➢ The zones are :


➢ -4 to -3, -3 to -2,-2 to -1,-1 to 0, 0 to 1,
1 to 2,…..
➢ The difference between actual and midpoint
➢ The midpoints are : value is referred to as the quantization error.
➢ -3.5, -2.5, -1.5, -0.5, 1.5, 2.5, 3.5 ➢ The more zones, the smaller ∆ which results
in smaller errors.
ENCODING
 In combining the process of sampling and quantization, the specification of the

continuous-time analog signal becomes limited to a discrete set of values.

 Representing each of this discrete set of values as a code called encoding process.

 Code consists of a number of code elements called symbols.


ASSIGNING CODES TO ZONES
 In combining the process of sampling and quantization, the specification of the
continuous-time analog signal becomes limited to a discrete set of values.
 Representing each of this discrete set of values as a code called encoding
process.
 Each zone is assigned a binary code.
 The binary code consists of bits.
 The 8 zone (or level) codes are therefore: 000, 001, 010, 011, 100, 101, 110, and
111
 Assigning codes to zones:
• 000 will refer to zone -4 to -3
• 001 to zone -3 to -2, etc.
With an example explain the sampling process?
Assume the sampling theorem states that a band-limited signal x(t) with
a bandwidth B can be reconstructed from its sample values if the
sampling rate (frequency) 𝑓𝑠 ≥ 2𝑓𝑚𝑎𝑥

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