[speex-dev] Server based audio merge
dj at tenet.res.in
Fri Nov 21 18:30:12 PST 2003
CELP based coders perform poorly when you mix streams. Perhaps, we need
a transform code or a subband code at somewhat higher rate 12-16 kbps
that will retain the quality after transcoding and mixing. Is the speex
ultra-wideband based on subband coding?
<p>On Fri, 21 Nov 2003, Ulrich B. Staudinger wrote:
> been silent in this discussion for quite long ...
> Am Fr, 2003-11-21 um 08.07 schrieb Jean-Marc Valin:
> > There's no perfect solution to the multiple client problem. Each
> > approach has advantages and drawbacks:
> > 1) Mixing at the server
> > - Allows a constant bandwidth for every client
> > - Allows compatibility with regular VoIP prones
> > - Requires transcoding, even when only on person is talking
> > - Higher bit-rate required for the general case (one speaker is talking)
> > 2) Sending multiple streams
> > - Possible to do without a server at all
> > - Best quality (no transcoding)
> > - Non-constant bandwidth
> > Jean-Marc
> i think 1) is definitely the way to go. 2) uses up to much bandwidth ...
> what i did when writing an avrealy in java (man speex is too expensive
> in java): i used 1) and mixed everything on the server by doing a simple
> addition of the various streams on raw level and then encoding after the
> prior decoding of course. it works very good, although it could all be
> improved a lot.
> what i do : i have a session frame and all connections are bundled in a
> session, all connections have buffers again. whenever there is data in
> the buffer of the connection, the connection get's added to the session
> buffer, which after mixing is wiped again, as is the connection buffer.
> as said, speex in java is too slow for encoding more than 3
> participients, at least on the machines i tested.
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