but there is a pesky DC bias that keeps changing every now and then
Fix that, if you can. If it changes suddenly, then any filter you have is going to have odd results after the change, until it settles again.
So this is the algorithm you want (note the pseudo-C pseudo code -- data types are up to you):
static state_t state = 0;
static const ...
Also, an accelerometer has a very flat frequency response till its (high frequent) resonance point.
But as the name gives away; the sensor provides you with acceleration data. If you want to know velocity, you need to integrate the signal once, and if you want to know displacement, you need to integrate the signal twice.
Each integration step adds a pole at ...
In audio applications, this would be a low-cut filter. The term is often used synonymous with high pass, though that would not accurately describe general zero-mean filters.
A unit-mean filter meanwhile is indeed a lowpass filter.
In complement to Marcus, I have read the term "zero-sum": "zero-sum window", "zero-sum filter", "zero-sum kernel", the latter being more frequent. It is similar to "unit-sum windows", ie windows whose amplitudes sum to one. "Zero-average" can be found in image processing:
Further note that applying ...
"Zero-Mean" is the word that's commonly used to describe signals and signals with a zero average. "This is a zero-mean filter."
If you really mean a filter that is specifically meant to cancel the DC component, a "DC blocker" is a name for that.
You probably meant to reverse them: The LPF should be at 1100 Hz, and the HPF should be at 100 Hz. Then you're keeping everything between 100 Hz and 1100 Hz, and throwing away lower and higher frequencies.
will eliminates the low frequency samples
Also remember filters don't eliminate everything in the stopband, they drop off with frequency, so some of ...