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How do we find fourier coefficients, ak, for any periodic signal?
Fourier analysis integral
Go from x(t) to ak
Defined over 1 period, T0

What are full-wave rectified sine waves (FWRS)?
Found in DC power supplies
Found in AC to DC power converters
Recharges batteries
What is the Fourier Analysis of a FWRS?
Use fourier integral for FWRS to find ak
Plug in FWRS signal
Plug in T0
Euler’s formula to write as sum of two integrals
Perform integral
Plug in upper and lower bounds
find e value
How do you determine the DC value of a Fourier Series
Set k = 0
Solve for ak

What is the spectrum of a FWRS?
Use ak values for the spectrum
Lines go at multiples for F0
Length of lines is ak
Spectrum goes both directions

How do finite and infinite synthesis compare?
Finite is bounded while infinite is not
Infinite has an infinite number of spectral lines and infinite bandwidth
Magnitude of the very high harmonics are extremely small
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What is the point of the Fourier Series Integral?
We can represent any periodic signal by using a sum of complex exponential signals
How can you derive Fourier Series Coefficients for pulse waves?
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Duty cycle
r = τ /T0
Ratio of time a signal or system spends active (on) compared to total duration of complete repeating period
How does the fourier series coefficients change with a square wave?
τ = T0/2 : Wave is square
ak = 0 when k is odd and phase is 0 because x(t) is centered at t=0
Fourier series coefficients are discrete frequencies kF0
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How can you derive Fourier Series Coefficients for triangular waves?
Continuous signals over one period
User fourier series integral to find ak
x(t) = |2t/To| for -T0/2 < t ≤ T0/2
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How can you derive Fourier Series Coefficients for half-wave recified sinusoids?
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How can you draw the spectrum from the Fourier Series Coefficients?
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How do we compute signal power?
A spectrum plot of ak is the distribution of signal power vs frequency
average power is the total energy of one period divided by the period
power depends only on amplitude of signal
typically measured in dB
How do we convert continuous-time signals to digital signals that can be processed by a computer?
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How do we compute the values of a discrete time signal directly from a formula?
we can use the formula
can also be obtained by sampling

How do we plot sequences?
we use a stem format

How do we sample sinusoidal signals?
discrete-time cosine signal
normalized radian frequency or digital frequency
What is aliasing?
alias: two different discrete-time sinusoid formulas can define the same signal values
What is the sampling theorem?
a continuous-time signal with frequencies no higher than fmax can be constructed from its samples if the samples are taken at a rate fs = 1/Ts, that is greater than 2fmax (Nyquist rate)
typically use the lowest possible sampling rate to minimize cost of storage and processing
if there are not enough samples, aliasing occurs and reconstruction becomes difficult
What happens when we over-sample?
when we sample at a rate of over 2fmax
use principal alias to reconstruct signal
What happens when we under-sample?
when fs < 2f0
aliasing occurs
What are the frequencies of physiological signals?
Nyquist rate sampling fs > 2fmax

What is the purpose of a D to C converter?
interpolates a smooth continuous-time function y(t) through the given discrete-time samples y[n]
How does a D to C converter work?
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What is a zero-order hold interpolation?
simplest pulse shape is a symmetric square pulse
What is first-order interpolation?
triangular pulse consists of first-order polynomial segments (straight lines)
What is cubic spline interpolation?
third-order polynomial segments
How does oversampling aid interpolation?
we get better reconstructions
What is band limited interpolation?
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What is a digital signal processor?
DSP
measure, filter, compress continuous real-world analog signals
What is a Finite Input Response (FIR) filter?
DSP whose impulse response settles to zero in a finite amount of time
What is a discrete-time system?
a sequence of data values defined only at specific points rather than continuously
What is the running average filter?
running or moving average of two or more consecutive values of the sequence
What are our options for indexing the output signal?
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What is the general FIR filter?
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What are the important properties of FIR filters?
modify signals usefully
length of average intervals has a big effect
running average filters introduce a shift of M/2 samples
What is the unit impulse sequence?
What is the unit impulse response?
What are the three ways to represent the FIR filter?
use shifted impulses to write h[n]
plot the values h[n]
list values bk
What is the convolution sum?
convolving the sequence h[n] and x[n] for finite length
h[n] = 0 for n<0 and n>M
What is the length of a convolution?
How do we use a FIR filter and convolution in MATLAB?
How do we filter the Unit-Step signal?
unit step turns on at n=0
3 cases for the output
What are the properties of convolution?
convolution is communicative
convolution is associative and distributive
What do we need to compute a FIR filter?
on a computer chip
multiplier
adder
unit delay operator
How do we represent systems with block diagrams?
nodes: junction points
branches: connections between nodes
feed forward difference equation
direct form: standard structure
How do we test for time invariance?
look at examples
How do we test for Linearity?
look at examples
What are cascaded LTI systems?
cascade connection of two systems means the output of the first is the input to the second system
for LTI, order of systems can be rearranged and provide the same output