So, this boils down instead of 12 k linearly spaced, to create just 5-7 FIR filters in parametric mode, when you can adjust freq. response, and tune each filter to exact freq. of interests.
The problem is the low end, for a LPF at 50Hz, you need at least 50Hz resolution (and use the lowest bin), and FIR is inherently linear, not logarithmic. That means to get 50Hz to work, I have to anyway build 3.84k bins at 192ksps (3.84k bins means 12k taps for Hanning window).
I can get around with a few short IIRs, but then how the phase delay will react is open to debate.
This can be compensated for by just using the appropriate delay factor when summing your filtered signal with the source audio, so long as your filter doesn't have any positive amplified gain within it. Post mixing a +/-% of the filter's output with the delayed source audio doesn't count. However, such a 50hz filter will not be narrow band. If you want anything with FIR performance, your stuck with 3.84k bins.
Simple sum-averaging the source audio into a 3.84k buffer, mixing a % of that result with the center sample in the buffer at the 1.92k point will give you an in-phase audiophile grade 0-50hz bin which you can add or subtract basically what you would call very low bass.
Within that 3.84k buffer, keep track of the center 1.92k samples as well making a second average would give you a 0-100hz bin. Subtract out the 0-50hz bin from 0-100hz bin, and you will have a 100hz +/- 50hz bin. This is an all integer equalizer which can be done in a small DSPIC with ease.
With this, you now have the processing power to create a 25hz filter as well as any other odd low frequency like 35hz, 36hz, 37hz, ect. You are no longer tied to exact multiple of your sample rate except for higher frequencies...
Sort of like the opposite problem to a FIR filter...
Now, when averaging, remember that 3840 * 2^16bit fits within a 32 bit integer.
Now, when averaging, remember that 3840 * 2^24 is more than a 32 bit integer.
Processing these averages in floating point solves this issue but introduces floating rounding errors.
This is how to make a super low processing power EQ, but, be warned that when applying a +%, you don't exceed the upper and lower sample limits of your variables bit depth.
Also, applying too much negative % (IE, lowering the BASS), you will go past lowering the bass to inverting the bass, then negatively amplifying it at -180 degrees to a n out of control level.
If you are creating a high pass filter, EG - DC removal/correction, you don't need the delay, but tune your filter for VLF.