syllabus for a Digital Signal Processing
syllabus for a Digital Signal Processing (DSP) course:
Unit I: Introduction to DSP
- Discrete-Time Signals and Systems: Classification, properties, and representation
- Linear Time-Invariant (LTI) Systems: Convolution, difference equations, impulse response
Unit II: Z-Transform
- Definition and Properties: Region of convergence, inverse Z-transform
- Applications: Solving difference equations, system analysis
Unit III: Fourier Analysis
- Discrete-Time Fourier Transform (DTFT): Properties and applications
- Discrete Fourier Transform (DFT): Properties, computation using FFT algorithms
- Fast Fourier Transform (FFT): Radix-2 FFT algorithms, applications
Unit IV: Digital Filter Design
- FIR Filters: Design techniques, window method, frequency sampling method
- IIR Filters: Design techniques, bilinear transformation, impulse invariance
- Filter Realization: Direct form, cascade form, parallel form
Unit V: Sampling and Reconstruction
- Sampling Theorem: Nyquist rate, aliasing
- Reconstruction: Ideal and practical reconstruction methods
Unit VI: Multirate Digital Signal Processing
- Decimation and Interpolation: Concepts, applications, polyphase structures
- Multirate Filter Banks: Analysis and synthesis filter banks, applications
Unit VII: Applications of DSP
- Speech Processing: Speech analysis, synthesis, and recognition
- Image Processing: Image enhancement, compression, and reconstruction
- Biomedical Signal Processing: ECG, EEG signal analysis
Practical/Lab Work
- MATLAB Simulations: Signal generation, system analysis, filter design
- Experiments: Verification of theoretical concepts through practical experiments
- Projects: Real-world DSP applications and implementations
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