[Asterisk-Users] Re: Audio filters (was: feature - VM gain adjust?)

[John Todd]

At 2:11 PM +0200 on 7/13/04, Holger Schurig wrote:
>  > liarliar.sourceforge.net gives you something, its still in development.
>
>Hehe, imagine a phone where you see a red LED flashing if the other person
>lies to you.
>
>
>
>When you thing about audio-plugins, you should think more into the
>direction of LADSPA, see http://www.ladspa.org/
>
>Amplifier: http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.5
>Compressor:
>http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.35
>
>There are possible others there as well.


LADSPA does indeed seem to be a good fit for such a project, though 
it is LGPL'ed and would thus be problematic to include into *'s main 
CVS tree.  However, perhaps some generic filter hooks could be built 
into * to allow insertion of LADSPA or LADSPA-like loops.

Even if LADSPA were to be integrated, the API would probably need to 
be modified such that it could be controlled by "external" programs.

Live manipulation of the stream parameters could be done in two ways 
(maybe more, haven't thought about it much):  Asterisk's RTP stack 
would have to intercept DTMF (because it's RFC2833, or INFO, or 
whatever) and then call out to some API that would permit 
manipulation of the stream.  That's difficult.  An alternate method 
would be to intercept DTMF within Asterisk, close the LADSPA stream 
down, and launch the API call again with different parameters (this 
would be easier, I suppose.)  This second method might even be 
transparent to the end user(s) if done fast enough.  This second 
method also assumes that Dial is capable of intercepting DTMF, doing 
some dialplan logic, and then re-connecting the two legs together 
without hanging up either leg (see my many, many previous discussions 
on this topic.)

All in all, not a simple project, but probably a good project for a 
graduate student who is working on practical applications of audio 
filtering programs.

JT