Cyber Lect 1
Cyber Lect 1
Cyber Lect 1
http://www.engineering.uodiyala.edu.iq/
https://www.facebook.com/Engineer.College1
Course Overview
• Signals and Systems.
• Classification of Discrete Digital Systems.
• Time and Frequency Domains Analysis.
• Signal Transformation Methods: Fourier, Wavelet
and Z-transform.
• Digital Filter Types: FIR and IIR Filters.
• Digital Filter Design.
• Analog Filter Design.
• DSP Applications.
2 Dept. of Computer and Software Engineering
Books
• J.G. Proakis and D.G. Manolakis, Digital Signal Processing, 4rd edition,
Prentice-Hall , 2006.
• J.G. Proakis , Digital Signal Processing Using MATLAB, 3rd edition, Cengage
Learning , 2012.
Discussion:
Early education in engineering focuses on the use of calculus to analyze
various systems and processes at the analog level:
motivated by the prevalence of the analog signal model
e.g.: circuit analysis using differential equations
Yet, due to extraordinary advances made in micro-electronics, the most
common/powerful processing devices today are digital in nature.
Thus, there is a strong, practical motivation to carry out the processing of
analog real-world signals using such digital devices.
This has lead to the development of an engineering discipline know as
digital signal processing DSP.
Digital Signal Processing:
In its most general form, DSP refers to the processing of analog signals by
means of discrete-time operations implemented on digital hardware.
• In its most general form, a DSP system will consist of three main
components, as illustrated below:
(2) convert the digital information, after being processed back to an analog signal –
involves digital-to- analog conversion and reconstruction .
e.g. text-to-speech signal (characters are used to generate artificial sound)
3) convert analog signals into the digital information - sampling & involves analog-to-
digital conversion.
e.g. Touch-Tone system of telephone dialling (when button is pushed two sinusoid signals
are generated (tones) and transmitted, a digital system determines the frequencies and
uniquely identifies the button – digital (1 to 12) output
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Sampling Theorem
Fs 2fm
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Nyquist sampling rate for low-pass and bandpass signals
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Quantization
• Sampling results in a series of pulses of varying
amplitude values ranging between two limits: a
min and a max.
• The amplitude values are infinite between the
two limits.
• We need to map the infinite amplitude values
onto a finite set of known values.
• This is achieved by dividing the distance between
min and max into L zones, each of height
= (max - min)/L
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Quantization Levels
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Quantization Error
• When a signal is quantized, we introduce an error
- the coded signal is an approximation of the
actual amplitude value.
• The difference between actual and coded value
(midpoint) is referred to as the quantization error.
• The more zones, the smaller which results in
smaller errors.
• BUT, the more zones the more bits required to
encode the samples -> higher bit rate
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Analog-to-digital Conversion
Solution: If the input range is 10 volts then the analog voltage represented
by the LSB would be:
V max 10 10
VLSB = Nu bits = 12 = = .0024 volts
2 2 4096
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DSP Chips : Special Purpose Hardware
Advantages:
• Robustness:
• Signal levels can be regenerated.
• Precision not affected by external factors
• Storage capability:
• DSP system can be interfaced to low-cost devices for lasting storage
• allows for off-line computations
• Flexibility:
• Easy control of system accuracy via changes in sampling rate and number of representation
bits.
• Software programmable → reconfiguring the DSP operations simply by changing the program.
• Structure:
• Easy interconnection of DSP blocks (no loading problem)
• Possibility of sharing a processor between several tasks
Disadvantages:
• Cost/complexity added by A/D and D/A conversion.
• Input signal bandwidth is technology limited.
• Quantization effects.