[net.audio] Philips 14-bit system

jlg@lanl.ARPA (03/28/85)

> Let's follow a signal through the Philips system.  At the beginning, the 2
> LSBs are truncated and applied to the digital filter.  The filter
> interpolates 3 new data points between samples, `recovering' the lost 2
> bits if 16 were used at the output, which they are not.  Thus the base S/N
> of the Philips system is 84dB.  Since the final bandwidth of the system is
> 1/4 the effective bandwidth of the filter output(i.e., the maximum
> bandwidth of an input signal sampled at the filter clock frequency) we gain
> 6dB in S/N.  The roundoff feedback further averages the quantization error
> to return to a 96dB S/N.

This is not what my article on the Philips system says.  It claims that
full 16-bit data items are fed to the digital filter with three words of
zero between each.  The digital filter is a 96 stage discrete convolution
integral with multipliers of 12-bits each.  The filter keeps all
intermediate values to the full 28-bits (product of 16-bit data with 12-bit
multiplier).  After the digital filter, the 28-bit result is truncated
to feed through the 14-bit DAC (This 14-bit result is dithered so that the
average of four 14-bit values equals the same result as a single 16-bit
filtered value would have been).  This is the reason that Philips claims
that the 14-bit system looses no information.

J. Giles