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Suppose I have a digital filter implemented in Direct Form II. How do I initialize the state of the filter as if the input $x[n]$ had a fixed value $x_0$ for all $n<0$?

Direct Form II Filter Topology

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  • $\begingroup$ normally we initialize states to zero, which corresponds to your input signal, $x[n]$ being prepended with an infinite number of samples that have zero value. $\endgroup$ – robert bristow-johnson Apr 5 at 22:42
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    $\begingroup$ solve this equation $$w[-1] = (-a_1) w[-1] + (-a_2) w[-1] + x_0$$ for $w[-1]$. $\endgroup$ – robert bristow-johnson Apr 5 at 22:47
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    $\begingroup$ I had asked a similar question: dsp.stackexchange.com/questions/50992/… a while back. I had never thought much of filter initialization until I had to implement a 0.01Hz low-pass filter. $\endgroup$ – Izzo May 6 at 0:16
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The difference equations for this filter are:

y[n] = b0 w[n] + b1 w[n-1] + b2 w[n-2]
w[n] =    x[n] - a1 w[n-1] - a2 w[n-2]

To achieve steady-state, w[n] == w[n-1] == w[n-2]. Call this value w. Solving the second difference equation, we find w = x/(1 + a1 + a2), where x is the steady-state input value.

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    $\begingroup$ you should swap the order of the two difference equations. w[n] is calculated first. $\endgroup$ – robert bristow-johnson Apr 5 at 22:40
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    $\begingroup$ BTW, i hope you're not using a DF2 filter with fixed-point arithmetic. DF2 and fixed-point don't like each other if the filter gets resonant. $\endgroup$ – robert bristow-johnson Apr 5 at 22:50

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