[speex-dev] Server based audio merge

Carsten Breuer CarstenBreuerSpeex at textwork.de
Fri Nov 21 11:02:06 PST 2003

Hi Ulrich,

<p>> 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. 

That's exactly what i have in mind.
Perhaps we should change that to C++ to get more
performance. I can also support you with coding because
that's the way i want to go. Please let me know which way you go.

> 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. 

Did i say that i love you ;-)))))).

> as said, speex in java is too slow for encoding more than 3
> participients, at least on the machines i tested. 

Let's start to port this to C++.
If this is also to slow, then we build smart multi core DSP boards,
running speex fast like hell and sell this for tons of dollars,
so that we can buy micrsosoft in some years (Sorry, i hav my kidding day 
today <vbg>).

Now, no kidding:

Have someone ever tried to run speex on DSP's.
What do you think is the adcantage?
How many chanels are posible?
Is CELP DSP fiendly?

<p><p>Best Regards,


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