[vorbis] Low bitrate encoding

[Gregory Maxwell]

On Tue, Jan 09, 2001 at 10:12:07AM -0500, Gregory Maxwell wrote:
> On Tue, Jan 09, 2001 at 03:41:58PM +0100, Dr.Joerg Bergmann wrote:
> > > 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!)
> 
> No. If you sample a signal with signifant components above the nyquest
> components these cofeeicents will be aliased and reflected back into the
> bandpass of your sampled data. Not cool.
> 
> To get good quality downsampling (or A/Ding) you must filter first. This is
> non-trivial especially when the rates in noninteger.

To get a mental handle on this: Imagine sampling a 15KHz signal at 8KHz.
Would you get all zeros?

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