After wondering about the compression procedure of the audio
DSP, I got to thinking "Eh ?". If each block of 8 bits of data is
expanded to 10 bits by left shifting and 0 padding, then are the
functions of the Estimate and Syllabic Integrators to sense trends
in the waveform and replace the LSB "00" bits with fill-in values ?
It's the only way I can see to get 10-bit quality sound from 8 bits
of data. 8 bit audio does not sound very nice at 16kHz (which is
originally sampled at 64kHz), and the DSP's output is definitely OK

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