Digital Signal Processing
|Lecture hours||3 hours|
|Lab hours||2 hours|
|Digital resources||View on Evdoxos (Open e-Class)|
The aim of the course is to introduce students to the theory of time and spectral field by which continuous and discrete time systems are analyzed and designed. Based on this theory, students will be able to design analogue and digital filters based on frequency response specifications.
Upon successful completion of the course the student will be able to:
- Be familiar with FIR and IIR filter design algorithms
- create transfer function of generic analog filters
- Handle IIR and FIR filter implementation methods: serial and parallel structures.
- Be able to design IIR and FIR filters using Matlab software tool.
- Discrete time convolution, Z transform, frequency response of discrete time signals and systems.
- Prototypes of analogue lowpass filters: Butterworth polynomials and Chebyshev polynomials.
- Frequency translation of normalized analogue filters, general algorithm for creating arbitrary analogue filters.
- Bilinear transformation.
- Design of digital infinite impulse response (IIR) filters using bilinear transformation.
- Frequency transformation of digital filters.
- Digital finite impulse response (FIR) filters with linear phase.
- FIR filter design using frequency sampling.
- FIR filter design using optimal method.
- Implementation issues and techniques for IIR and FIR digital filters.
- Proakis J. & Manolakis D. (2007): Digital Signal Processing: Principles, Algorithms and Applications, 4th Edition, Prentice Hall.
- Ingle V. & Proakis J. (2000): Digital Signal Processing Using Matlab, Brooks/Cole Publishing.