On the digital filter issue here is what can happen:
lets say you are feeding the ADC in the recording studio an analog
waveform in which the peaks are exactly at the level to produce a Full
Scale (FS) output. What happens when a sharp peak (say a piano key
pressed hard) hits inbetween the sample points? The numbers that come
out of  the ADC are slightly less than FS, the sample time hits the
"skirts" rather than the peak. When these samples are fed to a DAC the
digital filter inside reconstructs the peak inbetween the original
sample points at exactly FS, the DAC handles that correctly and
everything is fine.  

But what frequently happens is the track gets processed and those
samples which were slightly below FS now get boosted so they are at
exactly FS. When THESE samples hit the digital filter in the DAC it now
produces a peak which is greater than FS. There are a few different ways
a DAC can deal with this. If it was not specifically designed to deal
with this issue that peak above FS will most likely wrap around and
become a FS peak of the opposite polarity, a short impulse as loud as
the DAC can  produce, a REALLY REALLY bad distortion! It can clip the
value, this is not  as bad since it is just chopping off the very peak
of the waveform, OR it can decrease the volume of the whole waveform
very slightly so it can properly reproduce these peaks above  FS. 

The last option seems like a good way to deal with it, BUT the chip
makers don't like doing this, it means everything is slightly lower in
amplitude which decreases the overall SN ratio of  the chip, and to a
chip maker that is  the single most important spec to get as good as
possible. Doing this to one of  their chips puts it at a disadvantage in
the marketplace. If every chip on the planet did this and did it by the
same amount, then it probably would not be an issue. But then you would
have massive creative specmanship going on.

John S.


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