* Remove (most of) local libjingle mods. Only webrtcvideoengine and webrtcvoiceengine are left now, because the refcounted module has not yet been released to libjingle, so I can't submit the changes to libjingle at the moment.
* Update the peerconnection client sample app.
Review URL: http://webrtc-codereview.appspot.com/151004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@625 4adac7df-926f-26a2-2b94-8c16560cd09d
* Fixes bug where remote video wasn't renderered.
* Update the Conductor class in accordance to the latest changes in the API.
We now process the stream add/remove callbacks asynchronously.
* When a remote peer connects to us, we now call AddStream for our local streams
to share with the peer if we haven't already done so. To do that, we maintain
a set of streams we have already shared.
BUG=11
Review URL: http://webrtc-codereview.appspot.com/131011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@506 4adac7df-926f-26a2-2b94-8c16560cd09d
I made several updates to the Windows version as well so that both
implementations share
a big portion of the code.
The underlying PeerConnection notifications have changed a bit since the last
update
so that there's still a known issue that I plan to fix in my next change:
// TODO(tommi): There's a problem now with terminating connections:
// When ending a conversation, both peers now send a signaling message
// that indicates that their ports are closed (port=0). The trouble this
// causes us here is that we can interpret such a message as an invite
// to a new conversation. So, currently there is a bug that ending
// a conversation can immediately start a new one.
// To fix this I plan to change how conversations start and have a special
// notification message via the server that prepares a client for a
// conversation instead of automatically recognizing the first signaling
// message as an invite.
Review URL: http://webrtc-codereview.appspot.com/112008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@446 4adac7df-926f-26a2-2b94-8c16560cd09d
