Audio quality improvement for 16-bit fixed point SBC encoder

Multiplying the first part of the analysis filter constant tables
by some coefficients and dividing the second part by the same
coefficients is a transformation which should produce the same
results if rounding errors are not taken into account. These
additional C0/C1/... coefficients can be varied in a certain
range (the requirement is that we still do not get overflows).
The 'magic' values for these coefficients are selected in such
a way that the rounding errors are minimized (rounding errors
are unavoidable when putting all the floating constants into
16-bit tables and losing some of the fractional part).

Also non-SIMD variant of the analysis filter is dropped because
keeping it would require applying a similar change to its tables,
which is a bit tricky and just increases maintenance overhead.

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