Commit Graph

21 Commits

Author SHA1 Message Date
perkj@webrtc.org
99239d5a41 First compiling version of peerconnection_client_dev using the new Peerconnection API.
Links but does not work since the new peerconnection is under development.
I would like to commit a version with as few changes as possible to the old peerconnection_client but using the new PeerConnection API.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/183003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@677 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 15:59:40 +00:00
wu@webrtc.org
cb99f78653 * Update to use libjingle r85.
* 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
2011-09-19 21:59:33 +00:00
tommi@webrtc.org
a027bed377 Handle a null local renderer for times when there's no local cam.
Review URL: http://webrtc-codereview.appspot.com/138015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@545 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 09:52:45 +00:00
andrew@webrtc.org
e4c4d4f0e9 Fix "unused variable" warning in release mode.
Review URL: http://webrtc-codereview.appspot.com/131015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@537 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-06 16:31:55 +00:00
andrew@webrtc.org
49e58da5b1 Fix release mode "unused variable" warnings in peerconnection.
Review URL: http://webrtc-codereview.appspot.com/133010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@510 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-01 17:29:43 +00:00
tommi@webrtc.org
c6e54a97a7 Update to the peerconnection sample app.
* 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
2011-09-01 08:37:05 +00:00
mallinath@webrtc.org
f990eb3e88 Hi,
Removed OnLocalStreamInitialized callback from the PeerConnection callback list. After adding OnAddStream trigger at the originator this callback was redundant. Also other modification is to provide same stream label in OnAddStream callback at the originator which provided in AddStream API.
Review URL: http://webrtc-codereview.appspot.com/138002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@490 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-30 17:16:35 +00:00
tommi@webrtc.org
8811e5af02 Switch to a smoother stretch algorithm on Windows and delete buffers from previous conversations on linux when switching back to peer list.
Review URL: http://webrtc-codereview.appspot.com/135003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@488 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-30 08:39:04 +00:00
tommi@webrtc.org
350d091e0e Send the hangup message when asked to disconnect from a peer.
Review URL: http://webrtc-codereview.appspot.com/131006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@459 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-26 13:20:41 +00:00
tommi@webrtc.org
102b2270c7 First version of the peerconnection client application for Linux.
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
2011-08-25 15:03:52 +00:00
henrikg@webrtc.org
a2de6060b7 Review URL: http://webrtc-codereview.appspot.com/108007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@400 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-18 11:11:04 +00:00
tommi@google.com
b0d7a87bb0 Mock implementation for the UI of the linux version of the peerconnection client.
At this point, there's not a lot too it as it only shows what the UI will look like and basically mimics what the Windows version does presently.
Review URL: http://webrtc-codereview.appspot.com/92018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@344 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-10 09:03:29 +00:00
tommi@google.com
d15afa86c2 fix build warnings on linux.
Review URL: http://webrtc-codereview.appspot.com/99003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@335 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-09 13:40:24 +00:00
ronghuawu@google.com
e256187f8b * Push the //depotGoogle/chrome/third_party/libjingle/...@38654 to svn third_party_mods\libjingle.
* Update the peerconnection sample client accordingly.
Review URL: http://webrtc-codereview.appspot.com/60008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@302 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-04 17:44:30 +00:00
tommi@google.com
53af7595d1 Switch the sample client back to render the videos in the main window
instead of two popup windows.  This also demonstrates one way of
implementing the VideoRenderer interface.
Review URL: http://webrtc-codereview.appspot.com/51004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@143 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-04 12:47:37 +00:00
tommi@google.com
b2e56b9816 Switch use of wsprintfW out for the libjingle equivalent.
Review URL: http://webrtc-codereview.appspot.com/55001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@135 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-30 07:30:13 +00:00
niklase@google.com
b808501c30 If this gives you problems, delete the third_party/libjingle directory and sync again
Review URL: http://webrtc-codereview.appspot.com/22023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@57 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-08 11:24:32 +00:00
niklase@google.com
0839478fa7 git-svn-id: http://webrtc.googlecode.com/svn/trunk@45 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-06-07 09:00:54 +00:00
ronghuawu@google.com
e6988b9de5 * Update the session layer to p4 37930
* Update the peerconnection_client in sync with updates on the libjingle side.
Review URL: http://webrtc-codereview.appspot.com/29008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@34 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-01 18:50:40 +00:00
niklase@google.com
dbad7582d5 git-svn-id: http://webrtc.googlecode.com/svn/trunk@12 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 12:15:30 +00:00
niklase@google.com
278733b2d9 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:39:02 +00:00