Salve,
one year ago here was already a discussion about
a version of Ogg Vorbis, with low delay enough for
VoIP
http://www.xiph.org/archives/vorbis/200108/0106.html
>all transform codecs. They require a fairly large block of samples, i.e. in
>Vorbis you need to input 3072 samples (or is it 4096) from each channel
>before you can get any output. This could be reduced, trading off quality,
>but could not be reduced enough.
AAC Low Delay already reach < 20 ms
http://www.iis.fhg.de/amm/techinf/mpeg4/aac_ld.html
Ogg Vorbis is great, and I want to work the vision of ross
http://ross.sf.net
come true: a software-tool-box for radiostaion using GNU,
software only.
But for radiostations who want to interview
people sitting far away in an second studio, a low delay
of the audiocodec would be as important as for VoIP.
Is this feature possible to implement in Ogg Vorbis and
on which position is it on the to-do list?
When would it come true, 1-2 years?
Greetings form Aachen,
Rob
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