Channel equalization affect on input signal
This appears to be a homework problem so I will avoid providing the complete solution here, but the following should suggest a path toward a solution:
Typically this problem is done when the channel is NOT known and results in a least squared approach given the effects of the added noise $v(n)$. Solving for the equalizer coefficients when the channel is a known quantity results in a similar process to solving for inverse convolution using a matrix equation approach as is linked here: Compensating Loudspeaker frequency response in an audio signal with a convolution equation of the form:
$$h[n] * w[n] = \delta[0]$$
Where $h[n]$ represents the non-causal channel response and $w[n]$ represents the non-causal equalizer coefficients. I use &quot;non-causal&quot; to mean each is centered about $n=0$ which simplifies the equation to remove any delay in the result, as in $\delta[0]$ vs $\delta[n-m]$. The result in turn for the causal solution simply changes the indexing back such that the first sample starts at $n=0$ rather than the middle sample.
In basic English this is saying knowing $h[n]$, solve for $w[n]$ that would result in an impulse response when the two systems are cascaded, since the output and input of a system whose impulse response is an impulse would be identical.
This would result in $y[n] = s[n]$ when the noise is not added, and since the noise is white, this would also be the solution that any least squares approach using actual signals $s[n]$ and $x[n]$ would converge to. As constructed I do not see this as a least squares problem but simply an inverse convolution problem. This is similar to the use of least-squares versus zero-forcing equalizers and has similar construction: the least-squares solution converges to the zero-forcing solution as SNR increases.
In this case we solve for $w[n]$ using inverse convolution.