[speex-dev] Server based audio merge

Carsten Breuer CarstenBreuerSpeex at textwork.de
Fri Nov 21 10:44:21 PST 2003

Hi Allen,

> I tend to disagree.  It normal human conversation it wouldn't make much
> sense to have 2 people talking over each other at the same time.  

One of the problem is, that if the server doesn't distribute the stuff,
then one entity must send the stream to every other entity. That could 
work fine with fast connections, but doesn't work with a modem connection.

My mother have onl< a 33K6 connection. I dont want here to use this 
bandwith ONLY to send identical audio packages to different entitys.

BTW: The server should only merge stuff, if it is necessary.
Some people need fideback like hmmm, aaah, oooh etc ;-)).
Others say allways : ja .... ja....jo.....yes.....ja...oooh ;-))
And The others say ... yes, but .... yes, but... no, because....

You need this feedback. People are iritated if the call someone how
have a phone that mutes in nobody is speaking. Youz have the feeling 
that you speak with a wall. That's it.

 > Thus,
> it most scenarios you would have only one talker anyway.  Additionally,
> encode->decode/mix/encode->decode isn't a very efficient CPU process for
> a server, it's complicated to keep timing correct and it has a negative
> impact on total latency.

True, but unless you change The internet, there are not much posibilities.

> The overhead required to mix merge and re-encode is usually not worth
> the benefit as in most situations you are not really saving any
> bandwidth.

Sure you do and you must do, because elsewhere it doesn't go through the 

<p>Best Regards,


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