Dynamically changing cut-off and sampling frequency of a digital filter I have designed a low pass filter that smooths the output coming from an accelerometer attached to a vibrating machine. I designed this assuming cut-off frequency $f_{c1}$, and sampling frequency $f_s$. Now the issue is when I change the machine , I have to change the cut-off frequency and $f_s$ manually, else the output is not as smooth as expected. That is every machine is actually vibrating with slightly changed frequency. 

Is there a way the filter learns the appropriate Cut-off frequency $f_{c2}$ and $f_{s2}$, itself, without me to change the code  and perform filtering based on this $f_{c2}$ and $f_{s2}$?       

I shall appreciate if some one can provide some web-links so I can explore this area further.
There are a few ways that you could do run-time dynamic filters.


With a program like Matlab or Octave, but I'm guessing that they aren't available to your application.  There might be DSP libraries that have functions for generating filters though.
Calculating from scratch a windowed sinc function.
You could create somewhat crude low-pass filters based off of your starting filter.  For instance, if you wanted to halve the cutoff frequency you would halve the filter sample time.  You would thus create a new filter that had all of the original filter's samples and a sample in between those samples.  You could create the extra samples through splines.  This same approach could be used to create any fractional cutoff frequency change.  A more complicated example would be if you wanted the cutoff frequency to be $\frac{9}{8}f_c$.  If we say that your original filter's samples are at time 0, 1, 2, ..., $n$, then the new filter's samples would be at time 0, $\frac{9}{8}$, $\frac{18}{8}$, ...