[Freeswitch-users] implementing stereo (e.g. portaudio to test it easily)

[Tim St. Pierre]

Here's an idea  - it might be a little easier to make interoperate with 
other SIP devices.

What if you built a conferencing endpoint that mixed all the audio 
locally into a 3D matrix?

Each participant in the call could use any of the common codecs, but 
would have an extra SIP header than indicated their position in the 
sound field.

Here's a bit of an overview:

A regular Freeswitch system takes care of registering endpoints, 
routing, etc.  Just like a normal PBX.  The only thing special it would 
do is set add a custom SIP header with a person's sound field position, 
based on a directory variable.

Each conference unit would be a stripped down freeswitch system with a 
custom modified mod_conference.  In addition to mixing audio to mono 
like the regular mod_conference does, it would also generate a stereo 
mix, based on the position information of each participant.  It could 
dump out the stereo audio directly to a named pipe, mod_shout, or 
directly to audio hardware.  You could probably shoehorn a stereo input 
from hardware as well.  Essentially, you are building a stereo 
conference phone.

If you want to join up multiple conference units, just establish two 
sessions between each conference - one panned hard left, the other hard 
right.  They can then use G722 without any modification.

This probably isn't quite as nice as having true stereo codecs and media 
handling, but it may be easier to implement.  After all, telecom is very 
much a mono world.

-Tim
> The main reason we do not have stereo completely implemented is there
> was not really any codec or devices that could handle it to bother
> testing with.
>