[vorbis] Low bitrate encoding

[Dr.Joerg Bergmann]

The question is related to the question "What kind of signal does
a sound card sample?". Mostly, one assumes the AD converter samples
at single points following each other with the sampling frequency.
But, this is impossible. This would include a more and more random
error due to very high frequency parts (electronic noise has a temperature
dependent lower limit, the random error enlarges with smaller 
sampling time). Therefore, for enlarging signal/noise ratio, the AD
converter will probably _integrate_ over the whole interval between
two sampling points. This probably is most equivalent to summation
of n samples and dividing by n for downsampling by the integer value n.
The result will be equivalent to the signal sampled by the AD
converter at the nth part of the original sampling rate.

Right?

J"org Bergmann, Dresden, Germany
email at jbergmann.de

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