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Re: "By moving the crossover point from 22.05 to 88.2kHz, 96kHz ...

But who processes digital without a buffer? So there is a half second delay between the time you hit play and the time you hear sound.

Clock cycles are getting cheaper every second and it is only a short span of time before this is a moot point, the batch of RCS's out now are showing what is around the corner.

But I agree that current implementations, though strong in theory are woefully inadequate. Designs likes yours are proving that but without the higher precision input receivers(and other components) now avaliable to use, your design would be no different from those back in '81. DSD is showing that we have come a ways and that higher sampling rates can fix the problems without the comprise of analog filters in the output of "redbook" DACs.

****************

Now back to the subject I asked you in this thread and your response,

The assumption that we can somehow know what the musical signal was "before" and then be able to correct (or should I say reinstate?) it "after" is just plain nonsense, there is no servo system fast enough to correct any of this within the real time time domain, no matter how much processing power is available.

Real "realtime" isn't even a possibility with analog components, there is always a lag time. At least with recorded digital streams, we can run data faster than 1x realtime and process it before its needed. Any kid with his computer does this while ripping CD into MPGs.

Also, where can I find a good plot of your DACs performance?


-CAL


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