vorbis_analysis(&vblock, &ogg_pkt). Then, your ogg_pkt contains one ogg
packet - that's just the raw vorbis data. ogg_pkt.packet, of length
ogg_pkt.bytes.
You'll need to recreate some of the framing information in ogg - but not all
of it. See the vorbis-over-RTP drafts for one way to do this.
>
> > You might also be able to get away with it (again, this depends: you
> > haven't given much detail, so I can't say for sure) by just
ensuring you
> > always write blocks of audio of some minimum size (for VBR modes, this
> > size needn't be all that large. It depends on sampling rate, so
give more
> > details if you want recommendations on this), and using a low-latency
> > encapsulation layer (i.e. probably not ogg).
>
> Im using a sample rate of 8000, quality 0.0 and mono audio, just to ensure
> that im using the minimal bandwidth.
This seems rather at odds with your previous assertion that you were doing
this for streaming over a LAN. Surely using such low bandwidth isn't really
neccesary there? And your whole approach is doomed to failure
Anyway, at this low a bitrate, you'll find that your current use of ogg is
probably more than doubling your bitrate.
You'll also find that libvorbis cannot get particularly low-latency at this
sample rate. Around one second is likely to be the best you can do.
Mike
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