Digital Signal Processing Course Details

• Course Code : 0113620

Prof. Nizamettin AYDIN

[email protected] • Course Name: Digital Signal Processing

(Sayısal Đşaret Đşleme)
http://www.yildiz.edu.tr/~naydin
• Instructor : Nizamettin AYDIN

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Assesment Course Outline

1. Introduction.
Mathematical Representation of Signals. Mathematical Representation of Systems.
• Midterm 1 : 20% 2. Sinusoids.
Review of Sine and Cosine Functions. Sinusoidal Signals. Sampling and Plotting
Sinusoids. Complex Exponentials and Phasors. Phasor Addition. Time Signals.
3. Spectrum Representation.
The Spectrum of a Sum of Sinusoids. Beat Notes. Periodic Waveforms. Fourier
• Midterm 2 : 20% Series Analysis and Synthesis. Time-Frequency Spectrum. Frequency Modulation.
4. Sampling and Aliasing.
Sampling. Spectrum View of Sampling and Reconstruction. Discrete-to-Continuous
Conversion. The Sampling Theorem.
• Homework : 20% 5. FIR Filters.
Discrete-Time Systems. The Running Average Filter. The General FIR Filter.
Implementation of FIR Filters. Linear Time-Invariant (LTI) Systems. Convolution
and LTI Systems. Cascaded LTI Systems. Example of FIR Filtering.
6. Frequency Response of FIR Filters.
• Final : 40% Sinusoidal Response of FIR Systems. Superposition and the Frequency Response.
Steady State and Transient Response. Properties of the Frequency Response.
Graphical Representation of the Frequency Response. Cascaded LTI Systems.
Running-Average Filtering. Filtering Sampled Continuous-Time Signals.
7. z-Transforms.
Definition of the z-Transform. The z-Transform and Linear Systems. Properties of
the z-Transform. The z-Transform as an Operator. Convolution and the z-
3 Transform. Practical Bandpass Filter Design. Properties of Linear Phase Filters. 4

Course Outline Main course book

8. IIR Filters.
The General IIR Difference Equation. Time-Domain Response. System Function of
an IIR Filter. Poles and Zeros. Frequency Response of an IIR Filter. The Inverse z-
Transform and Some Applications. Second-Order Filters. Frequency Response of
Second-Order IIR Filter. Example of an IIR Lowpass Filter.
9. Continuous-Time Signals and LTI Systems. Signal Processing First
Continuous-Time Signals. The Unit Impulse. Continuous-Time Systems. Linear
Time-Invariant Systems. Impulse Responses of Basic LTI Systems. Convolution of
Impulses. Evaluating Convolution Integrals. Properties of LTI Systems.
by James H McClellan,
10. The Frequency Response. Ronald W. Schaffer
The Frequency Response Function for LTI Systems. Response to Real Sinusoidal
Signals. Ideal Filters. Application of Ideal Filters. Time-Domain or Frequency-
Domain?
and Mark A. Yoder.
11. Continuous-Time Fourier Transform.
Definition of the Fourier Transform. The Fourier Transform and the Spectrum.
Published by Prentice
Examples of Fourier Transform Pairs. Properties of Fourier Transform Pairs. The
Convolution Property. Basic LTI Systems. The Multiplication Property. Hall.
12. Filtering, Modulation, and Sampling.
Linear Time-Invariant Systems. Sinewave Amplitude Modulation. Sampling and
Isbn: 0-13-120265-0
Reconstruction.
13. Computing the Spectrum.
Finite Fourier Sum. Time-windowing. Analysis of a Sum of Sinusoids. Discrete
Fourier Transform. Spectrum Analysis of Finite-Length Signals. Spectrum Analysis
of Periodic Signals. The Spectrogram. The Fast Fourier Transform (FFT). 5 6

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 some recommended books Digital Signal Processing (DSP)
• Understanding Digital Signal Processing by Richard G. Lyons.
• The Scientist and Engineer's and Guide to Digital Signal Processing by
Steven W. Smith.
• Digital Signal Processing and the Microcontroller by Dale Grover and
John R. (Jack) Deller with illustrations by Jonathan Roth.
• Discrete-Time Signal Processing by A. V. Oppenheim and R. W. Schafer.
Basics:
• Digital Signal Processing: Principles, Algorithms, and Applications by
J. G. Proakis and D. G. Manolakis.
• Digital Signal Processing in Communication Systems by Marvin E.
What is DSP?
Frerking.
• Multirate Digital Signal Processing by R. E. Crochiere and L. R. Rabiner.
• Theory and Application of Digital Signal Processing by Rabiner and
Gold. A comprehensive, industrial-strength DSP reference book.
• Digital Signal Processing by Alan V. Oppenheim and Ronald W. Schafer.
Another industrial-strength reference.
• Disrete-Time Signal Processing by Alan V. Oppenheim and Ronald W.
Schafer Digital Signal Processing by William D. Stanley.

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CONVERGING FIELDS COURSE OBJECTIVE

• Students will be able to:
Math Physics
• Understand mathematical descriptions of
signal processing algorithms
EE
CmpE
Computer
• Express those algorithms as computer
Science Applications implementations (MATLAB)

BIO
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Digital Signal Processing (DSP) WHY USE DSP ?

Dictionary definitions of the words in DSP:
• Digital • Versatility:
– operating by the use of discrete signals to represent data in the form of – digital systems can be reprogrammed for other applications
numbers
• Signal – digital systems can be ported to different hardware
– a variable parameter by which information is conveyed through an electronic
circuit • Repeatability:
• Processing
– to perform operations on data according to programmed instructions – digital systems can be easily duplicated
• So a simple definition of DSP could be: – digital systems do not depend on strict component
– changing or analysing information which is measured as discrete sequences
of numbers tolerances
• Unique features of DSP as opposed to ordinary digital processing: – digital system responses do not drift with temperature
– signals come from the real world
• this intimate connection with the real world leads to many unique needs
such as the need to react in real time and a need to measure signals and
• Simplicity:
convert them to digital numbers – some things can be done more easily digitally than with
– signals are discrete
• which means the information in between discrete samples is lost analogue systems

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DSP is used in a very wide variety of applications Fundamental concepts in DSP
• Radar, sonar, telephony, audio, multimedia, • DSP applications deal with analogue signals
communications, ultrasound, process control, digital
– the analogue signal has to be converted to digital form
camera, digital tv, Telecommunications, Sound &
Music, Fourier Optics, X-ray Crystallography, Protein
Structure & DNA, Computerized Tomography, Reconst.
Nuclear Magnetic Resonance: MRI,Radioastronomy I
Anti-alias. A/D DSP D/A
&
Anti-
F
Filter image
• All these applications share some common features: (Signal
Processing
Filter

Reconst.
– they use a lot of maths (multiplying and adding signals) Q
Anti-alias. A/D
Algorithm)
D/A
&
Anti-
R
Filter image
– they deal with signals that come from the real world Filter

– they require a response in a certain time

• Where general purpose DSP processors are
concerned, most applications deal with signal
frequencies that are in the audio range

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• The analogue signal - a continuous variable • The continuous analogue signal has to be held before
defined with infinite precision - is converted to it can be sampled
a discrete sequence of measured values which
are represented digitally
• Information is lost in converting from analogue
to digital, due to: • Otherwise, the signal would be changing during the
– inaccuracies in the measurement measurement
– uncertainty in timing • Only after it has been held can the signal be measured,
– limits on the duration of the measurement and the measurement converted to a digital value
• These effects are called quantisation errors

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Sampling related concepts Steps for digitization/reconstruction of a signal

• Over/exact/under sampling
• Regular/irregular sampling • Band limiting (LPF) • D/A converter
• Linear/Logarithmic sampling • Sampling / Holding • Sampling / Holding
• Aliasing • Quantization • Image rejection
• Anti-aliasing filter • Coding
• Image These are basic steps for These are basic steps for
A/D conversion reconstructing a
• Anti-image filter
sampled digital signal

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Digital data: end product of A/D conversion and related
concepts Sampling
• Bit: least digital information, binary 1 or 0 • The sampling results in a discrete set of digital
numbers that represent measurements of the signal
• Nibble: 4 bits
– usually taken at equal intervals of time
• Byte: 8 bits, 2 nibbles • Sampling takes place after the hold
• Word: 16 bits, 2 bytes, 4 nibbles – The hold circuit must be fast enough that the signal is not
changing during the time the circuit is acquiring the signal
• Some jargon: value
– integer, signed integer, long integer, 2s • We don't know what we don't measure
complement, hexadecimal, octal, floating point,
• In the process of measuring the signal, some
etc.
information is lost

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