DSP Exam Aid
DSP Exam Aid
. To determine the unit step response of a system, multiply its system function H(z) with the z transform
of u[n]. This will be a convolution in the discrete time domain.
. If you have access to the system function, H(z), and you want to figure out the output in response to a
cosine or sine based signal, you must evaluate H(z) at z = e^jw.
- x(n) = Acos(wt+phi)
- y(n) = A|H(e^jw)|cos(wt+phi+<H(e^jw)).
- If it's a DC signal you are dealing with, use an omega or w of 0.
. High pass filters have the following z transforms: 1/(1+az^-1) where the value of a signifies how
strong/good the filter is.
. Low pass filters have the following z transforms: 1/(1-az^-1) where the value of a signifies how
strong/good the filter is.
. z^i - 1 = 0 yields i number of roots or angular frequencies of i/2*pi.
. What does the p/z plot and z transform look like for the different kinds of filters?
HOURLY 2
To Do List:
. What is leakage? Leakage occurs for peaks at non-integer values.
. Convolution
- Length of linear convolution = L + M - 1 where L and M are the discrete bandwidths of the signal.
- Length of circular convolution = 0 to N-1 where N is the number of samples in a linear convolution.
. DTFT and DFT of discrete time plots.
- DFT: L is the number of samples and N is basically the index number associated with the period. Refere
to Proakis Page 253.
. DFT
- Simple cosine/exponential based signals.
-- Simple cosine/exponential based signals in the continuous time domain in the case of the fourier
series would always produce peaks at 1 because their frequencies corresponded to the fundamental
frequency. In the case of the discrete time domain, the discrete fourier series is a sampled version of the
DTFT at a frequency of N. We calculate peak location on the basis of N.
- For simple cosine based signals, make sure you make adjustments to your signal according to the value
of N i.e. the frequency term needs to be in the form of 2pi/N and the amplitude at k_0 needs to be
equal to N.
- Complicated signals that rely on the use of the table of DFT pairs.
. DTFT
- Simple cosine/exponential based signals.
- Complicated signals that rely on the use of the table of DTFT pairs.
. Sampling
- A-to-D conversion etc.
- Aliasing; how to figure out the different aliases for a given signal.
- Downsampling.
- Upsampling.
. Circular Convlution:
- Do linear convolution on your signals.
- Linear convolution -> just shift the origin of your signal.
- N - 1 is the length of a single period of the convolved signal.
- Samples at n = N + all subsequent samples start over again at n = 0 i.e. they add to the samples at n = 0
and onwards.
Questions?
. Why is the phase delay exponential always decaying?
. How do you do the DFT or DFS for infinite sequences?
- The DFT and IDFT yield one period of the infinite sequences.
- It does not matter if the sequence is finite or infinite, we only use one period or N.
- For a finite length sequence, the value of N-1 in the sigma must correspond to the number of samples
in your signal.
. For a finite length sequence, does the number of k values also correspond to the number of samples in
your signal?
. Example 5.15, Signals and Systems, Oppenheim.
. pset05_3, McClellan.
. DFTs of real sequences and symmetry?
. The spectrum repeats itself for a DFT. Does the property of symmetry involve conjugates? Yes. For real
valued sequences. X_k = *X_N-k.
. N = Fs/F where F is the frequency resolution.
. Is L the number of samples for a DFT? No. It is the upper limit to your discrete signal.
DSP Preexam
. Revise Z-transforms + filters.
. Revise filters and their pz-plots and z-transforms.
. Revise all the transforms.
. Low priority: revise continuous convolution + look s09, Q2.a.
. Medium priority: revise the hourlies etc.
. Revise circular convolution.
. Look up the filter command in MATLAB.
. The multiplication property of the discrete fourier series.
. L = N for discrete time signals with no jumps.
. DFT needs some work.
. Revise freq. domain characteristics of lti systems.
. The length of continuous linear convolution = the two bandwidths starting from 0.
. The trick for cosine based functions in the case of the z transform which involves evaluating the system
function at z = e^jw also works for the continuous time fourier transform. Works for deltas. The output
is the basically the input with a phase change and a different amplitude.
. Remember there is always symmetry for filters.
. The cut-off frequency for a discretized and filtered signal should be handled with care. You need to
take into account the sampling rate.
. The range for a digital filter is between -fs/2 and fs/2.
. The combined system function is the product of the cascaded system functions and gives you the total
impulse response.
. The frequency response of a system or the magnitude of the system function is zero at the zeros. So
rearrange your z-transform or e^jw based expression to figure out the poles and zeros.
DSP Exam
. Nulling frequencies in the time domain.
- You sample them at frequencies where the undesired components overlap.
. DC component in the discrete time domain: signals between 0 and up or down.
. The z transform of a constant, A, is A/(1-z^-1).
. The DFT of a period impulse train produces a plot that has a DC component and a couple of
components like a cosine plot.
. Figuring out the DC gain from a PZ Plot?
. Filters and impulse response?
. For a signal to be periodic, it must contain frequency components that are multiples of the
fundamental frequency only. All other components MUST GO AWAY.
. The maximum frequency out of an ideal D to C converter is equal to Fs/2.
. The time delay property of the fourier transform: x(t-a) <=> X(F)e^-jwa.