Re: Great ! Speex codec impl

On Mon, Oct 27, 2003 at 09:39:30AM +1100, Soren H wrote:
 
> I am interested in continuing to maintain Speakfreely.  Thomas has
> been helping me sort out the bugs in my recent patches, and I'm
> reasonably positive about that and the other plans I have in mind for
> mods.  I've only recently become involved with the mailing list
> though, so I'm wondering who has been working on speakfreely recently?
> I know at least Thomas and myself have web pages with patches up.  Is
> anyone else actively doing things that could be brought together into
> a common update tree? 

Anshuman Aggarwal <[email protected]> has registered the
name speak-freely as sourceforge project.  

	http://speak-freely.sourceforge.net/

This site would be the logical place where developers meet and set up
their CVS server. You can already submit patches using the "tracker"
feature:

	http://sourceforge.net/tracker/?group_id=87107

I applied and received developer status. I am sure he would like others
to get registered as developers. 

> So Johannes, do you have any patches up somewhere?  Otherwise Bill
> must be talking about mine.  What I've done doesn't implement SIP at
> all.  It's still using the same protocol as before, but just adding a
> codec and better received packet handling.

I was only quoting Johns (your ;-?) mail to praise your/his efforts for
speak freely.  The patches are yours/his not mine.

Cheers 

Johannes

> I would be interested in adding a SIP library, or perhaps going with
> Thomas' idea of using the jabber network for session initiation.
> Jabber could be a nice option to replace the lwl server.  We just need
> to check if it can give the firewall information that would be useful
> for NAT piercing. 
 
> Over the weekend I modified the windows version of SF to be NAT
> friendly.  It just means using common sockets for TX and RX.  It makes
> an almost trivial patch.  The way I've done it still allows multiple
> receipients at once, but there may be a slight increase in system load
> because it can't use connect() to set the default address for the
> sendto() calls.  I don't know about that yet though - there may be no
> penalty at all. 
> 
> My future ideas are:
> - echo cancellation (for speaker-phone)  using libspeex.  This should
> be fairly easy on windows, but the split design of the unix version
> may need a rework.
> - adaptive jitter delays
> - adaptive audio buffering (I know how to do this with ALSA, but need
> to find out about underrun detection for other platforms) 
> 
> A this stage, it would be good to have some unix (non-linux) testers
> of my packeting and speex patches, so I could get some more confidence
> in that side of things.
> 
> My patches are at: http://www.2pi.info/software/sf_speex/index.html
> 
> I don't have the latest changes from the weekend up there yet
> though. (Windows NAT, unix ogg/speex logging to come soon)
> 
> 
> Soren
> (previously posting as John H from forum page)
> 
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