[Freeswitch-users] Manipulating audio streams at sample level

[Oivvio Polite]

I have a scenario where I'd like to be able to manipulate audio streams
coming into FS on a sample level.

This is a 3-party SIP call where 2 parties (A and B) are in the same room but
using individual handsets. This creates an echo effect for the third
party (C) since the audio streams from A and B will arrive at her location slightly out
of sync.

So at the start of a call I want to examine the audio streams coming from A B as they come into
FS to see how out of sync they are, then add an appropriate delay to the
one that's ahead and put that delayed audio back into FS.

Oivvio