EE432 Fall 2013: Instructional Objectives by Chapter
Text: Introduction to Digital Signal Processing 1/e by Blanding
Chapter 1: Introduction and Overview
1. State the three reasons to be concerned with frequency filtering.
2. Define the term finite precision.
Chapter 2: Discrete-Time Signals
1. Describe the steps necessary to get from a continuous-time signal x(t) to discrete signal x[n].
2. Define the unit impulse, unit step and general exponential functions.
3. Write any discrete signal as a sum of amplitude scaled and time-shifted impulses.
4. Determine if a given function is even, odd, or neither; determine the even and odd parts of a function.
5. For a discrete sinusoid, determine whether it is periodic or not, and if so, determine its period.
6. Determine the following properties of a discrete-time system: memory, invertibility, causality, stability, linearity and time-invariance.
7. Perform discrete-time convolution of two signals.
8. Calculate the impulse response of a system, given its difference equation.
9. Explain the difference between a finite impulse response (FIR) filter and an infinite impulse response (IIR) filter.
10. Determine the step response of a discrete system.
11. Calculate the overall impulse response of series and parallel systems.
12. Using impulse response, determine the power gain of a system.
Chapter 3: Frequency Domain Concepts
1. Calculate the Fourier transform of a continuous-time signal using the integral equation or a table of transform pairs.
2. Use the properties of the Fourier transform to solve related problems (e.g, time-shifting, etc).
3. Explain how the Discrete-Time Fourier Transform (DTFT) is created.
4. Describe the relation between frequency in Hz, rad/sec, and rad/sample.
5. Explain why the DTFT is periodic with period 2p rad/sample.
6. Calculate the DTFT for various discrete signals; plot its magnitude and phase.
7. Calculate frequency response of a system from its difference equation or impulse response.
8. Given samples of a discrete signal, calculate the Discrete Fourier Transform (DFT).
9. Given the DFT of a signal, calculate the signal.
10. Explain the relationship between the Fourier Transform, the DTFT and the DFT.
11. Use the Fast Fourier Transform (FFT) to calculate the DFT of a signal, and analyze its frequency content.
12. Describe the limitations of the FFT in terms of frequency resolution, and how resolution can be improved.
13. Calculate the Z-Transform (ZT) for various discrete signals; plot its magnitude, phase and region of convergence (ROC).
14. State the nature of the ROC of causal signals, anti-causal signals and non-causal signals.
15. Using the table of ZT pairs, partial fraction expansion or long division, calculate a discrete-time signal given its ZT.
16. Derive a system transfer function using the difference equation, and vice versa.
17. Given a system transfer function, determine the stability and causality of the system.
Chapter 4: Sampling and Reconstruction
1. Sketch the frequency spectrum of an impulse sampled signal, given the signal’s frequency spectrum and sample rate.
2. State Shannon’s Sampling Theorem, and define Nyquist rate, Nyquist frequency and aliasing.
3. Given a sinusoidal signal and sample rate, determine the frequency components that are produced from the sampling process; determine if aliasing occurs.
4. Describe how and why an anti-aliasing filter is used.
5. Calculate max quantization error and the quantization SNR (QSNR), given applicable A/D parameters.
6. Sketch the time signals and frequency spectrum at various points throughout the A/D and D/A process (see Figure 4.1 of text).
7. State how dithering is used to improve the A/D process.
8. Design a weighted resistor D/A, given the # of bits and desired resolution.
9. Describe the purpose of an anti-imaging filter.
Chapter 5: FIR Filter Design and Analysis
1. Given specifications or a plot of frequency response, determine ripple in the pass band and minimum stop band attenuation in dB.
2. Plot the magnitude and phase of the frequency response of an FIR filter, along with its pole-zero plot.
3. Describe the process needed to create an approximation to an ideal LPF using a digital sinc function (truncation, time-shift), and the reasons for these.
4. Explain the purpose of windows, and the relative advantages/disadvantages of the rectangular, Hanning, Hamming and Blackman windows.
5. Given a set of filter specifications, design a FIR LPF using the procedure outlined in the course handout. Using MATLAB, plot the frequency response and evaluate how actual frequency response compares to specifications.
6. Define the term linear phase and state how it affects frequency response.
7. Given a set of filter specifications, design a FIR BPF or HPF using the procedure outlined in the course handout. Using MATLAB, plot the frequency response and evaluate how actual frequency response compares to specifications.
8. Use MATLAB filter design tools to design FIR filters, and compare actual frequency response to specifications.
Chapter 6: Analysis and Design of IIR Filters
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Chapter 7: Sample Rate Conversion
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Chapter 8: Realization and Implementation of Digital Filters
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Chapter 9: Audio File Coding
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