add beginning of lab6-filter

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Klagarge authored and Rémi Heredero committed 2024-02-03 23:35:38 +01:00
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<header><h1>
Lab 06 - Filters
</h1></header>
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In this laboratory on digital filters, the main objective is to understand how digital filters work.
We will use MATLAB to simulate and implement these filters on a digital signal processor (DSP).
# Analog filter design
We must design Butterworth low-pass filter that cuts at 1kHz as shown with the following equation :
$$
F(s) = \frac{1}{\frac{s^{2}}{\omega_c^{2}} + s\cdot\frac{1.414}{\omega_c} + 1}
$$
In according to the sample theory, we have to choose at least 2 times the maximum frequency of the signal.
In this case, the maximum frequency is 1 kHz, so we have to choose a sample frequency of 2 kHz.
## Transfer function
For calculate the transfer function, we have to calculate the poles and zeros of the filter.
$$
p_{di} = e^{p_{ai} \cdot T} \qquad z_{di} = e^{z_{ai} \cdot T}
$$
- $k_a$, $p_{ai}$ and $z_{ai}$ are the gain, poles and zeros of the analog filter that the representation `zpk`.
Be careful, $k_a$ is not the DC gain.
- $k_d$, $p_{di}$ and $z_{di}$ are the gain, poles and zeros of the digital filter.
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