Sir Syed University Of Engineering and Technology, Karachi

Telecommunication Engineering Department

Course Title: Digital Signal Processing
Course Code: TE321
Credit Hours: Three
Semester: 5th
Course Instructor: Irfan Ahmed Usmani

Aims & Objectives:

This course introduces digital signal processing, providing the theoretical and practical
framework for understanding its algorithms, architectures, and applications. The course
covers the concepts and techniques of modern digital signal processing. The course starts
with an overview of signals and systems. The sampling theory of continuous-time signals
is explained next. The notion of discrete Fourier transform is introduced, followed by an
overview of fast algorithms for its computation. The methods for spectral analysis of
discrete-time signals are discussed next. Finally, principal methods for design of FIR and
IIR filters are covered, followed by a discussion of their applications.

Course Outline:
1. Introduction (2 weeks)
1.1. Review of Signals & Systems
1.2. Introduction to Digital Signal Processing
1.3. The Concept of frequency in Continuous-Time & Discrete-Time Signals
1.4. Analog to Digital & Digital to Analog Conversion
1.4.1. Sampling of Analog Signals
1.4.2. Sampling Theorem
1.4.3. Quantization of Continuous – Amplitude Signals
1.4.4. Quantization of Sinusoidal Signals
1.4.5. Coding of Quantized samples
1.4.6. Digital to Analog Conversion

2. Discrete Transforms (3 weeks)

2.1. Discrete-Time Fourier Transform (DTFT)
2.2. Discrete Fourier Transform (DFT)
2.2.1. Properties of DFT
2.2.1.1. Linearity
2.2.1.2. Circular shift of a sequence
2.2.1.3. Duality
2.2.1.4. Symmetry Properties
2.2.1.5. Circular Convolution
2.2.2. Linear Convolution using DFT
2.3. Efficient Computation of DFT
2.3.1. FFT Algorithm
2.3.1.1. Divide-and-Conqueror Approach to computation of DFT
2.3.1.2. Radix-2 FFT Algorithm: 2, 4 and 8 point FFT
2.3.2. Decimation-in-Frequency (DIF) FFT Algorithm

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 3. Review of Z-Transform (1 week)
3.1. The Z-Transform
3.1.1. Region of Convergence (ROC)
3.1.2. Properties of Z-Transform
3.2. Inverse Z-Transform
3.3. Solution of Difference equations using Z-Transform

4. A framework for Digital Filter Design (2 weeks)

4.1. Introduction to Digital Filters
4.2. The Difference Equation and Digital Filtering
4.3. Difference Equation and Transfer Function
4.3.1. Impulse Response, Step Response and System Response
4.4. Pole-Zero Plot and Stability
4.5. Digital Filter Frequency Response
4.6. Basic Types of Filtering
4.7. Filter Design Steps
4.7.1. Specification of filter requirements
4.7.2. Coefficient Calculation
4.7.3. Representation of a filter by a suitable structure (realization)
4.7.4. Analysis of finite wordlength effects
4.7.5. Implementation of filter
4.8. Realization of Digital Filters
4.8.1. Direct-Form I
4.8.2. Direct-Form II
4.8.3. Cascade (Series) Realization
4.8.4. Parallel Realization

5. Finite Impulse Response (FIR) Filter Design (3 weeks)

5.1. Introduction
5.2. FIR filter specifications
5.3. FIR Coefficient Calculation methods
5.3.1. Window method
5.3.1.1. Some common window functions
5.3.2. The optimal method
5.4. Realization structure of FIR: Transversal structure

6. Infinite Impulse Response (IIR) Filter Design (2 weeks)

6.1. IIR filter format
6.2. Coefficient Calculation methods of IIR filter
6.2.1. Bilinear transformation method of coefficient calculation
6.2.1.1. Analog Filter using lowpass prototype Transformation
6.2.1.2. Bilinear transformation and frequency warping
6.2.1.3. Bilinear transformation design procedure
6.2.2. Impulse Invariant design method

7. Optimal & Adaptive Filtering (2 weeks)

7.1. Wiener filtering
7.2. LMS & array processing

Books and References:

1. Digital Signal Processing (3rd Edition) John G. Proakis
2. Discrete-Time Signal Processing (2nd Edition) V. Oppenheim & W. Schafer
2. Digital Signal Processing Emmanuel I Feacshor
3. Digital Signal Processing Li Tan