[vorbis] Low bitrate encoding

[Dr.Joerg Bergmann]

On Tue, 9 Jan 2001, Marshall Eubanks wrote:

> 
> This is how the samplers that I am familar with work - it's
> called accumulation rather than integration, frequently.
> 
> One detail is what time is reported - the end time, or the middle
> of the sample.

This is _no_ question, an integral (or accumulation) is one identic
value for the whole interval. An average is an average, there are no
special averages for the middle and the end.

> 
> The other detail is what _analogue_ averaging (AKA low pass filtering) is
> applied. Generally (but not always) frequencies that are above the
> Nyquist frequency are low pass filtered.

Thats OK. But sampling itself adds some kind of low pass, an additional
analogue low pass will only reduce signal/noise ratio. An analogue
low pass may be (after fourier transformation) described by some
weighted averaging (weighting function is zero for negative times due to
signal theory, non-zero for positive values, in general none-zero for all
future times!)

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

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