[vorbis] Low bitrate encoding

[Marshall Eubanks]

"Dr.Joerg Bergmann" wrote:

> 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
>

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.

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.


--
                                 Regards
                                 Marshall Eubanks

T.M. Eubanks
Multicast Technologies, Inc
10301 Democracy Lane, Suite 410
Fairfax, Virginia 22030
Phone : 703-293-9624
Fax     : 703-293-9609
e-mail : tme at on-the-i.com     tme at multicasttech.com

http://www.on-the-i.com http://www.buzzwaves.com

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