Commit Graph

21 Commits

Author SHA1 Message Date
perkj@webrtc.org
36a992b030 Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection.
-Removed ssrc from tracks.
-Updated PeerConnectionMessage parsing and serialization.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@856 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 11:46:56 +00:00
mallinath@webrtc.org
c01c358f54 session/phone/channel.cc updates after new push of libjingle revision.
Review URL: http://webrtc-codereview.appspot.com/225003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@744 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 09:45:24 +00:00
henrike@webrtc.org
03a86998cd Fixes for build errors introduced most likely earlier today.
Review URL: http://webrtc-codereview.appspot.com/228003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@742 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 23:36:38 +00:00
wu@webrtc.org
6c2d7107ae * Update to use the new libjingle release.
* Stop using any local mods for the default build (non-dev).
Review URL: http://webrtc-codereview.appspot.com/224001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@737 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 16:58:50 +00:00
mallinath@webrtc.org
bafca109db Temp hook in WebRtcSession to VideoChannel.
Review URL: http://webrtc-codereview.appspot.com/195001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@689 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 17:45:21 +00:00
perkj@webrtc.org
2f56ff48a4 Implementation of PcSignaling. A Class to handle signaling between peerconnections.
Review URL: http://webrtc-codereview.appspot.com/149002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@657 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 20:35:37 +00:00
ronghuawu@google.com
c389aa2615 Fix the bad video issue on Window client by increasing the rtp recv buffer size.
Send buffer doesn't really matter, just to keep the same as talk does.

The same fix is submitted to libjingle for reivew. But I think it's worth to fix it here too as
it may take while for webrtc to get from libjingle. This patch is slightly different then that
one as I don't want to add the webrtcvideoengine.h back to webrtc.
Review URL: http://webrtc-codereview.appspot.com/166002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@634 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-21 16:53:45 +00:00
wu@webrtc.org
c49db5ea48 The files included in devicemanager.h/cc still have some conflict with chromium. Let's keep the devicemanager mods for now and I will see how can we solve this next.
Review URL: http://webrtc-codereview.appspot.com/166001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@626 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-20 00:40:52 +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
perkj@webrtc.org
0cc68dc38a Change Video capture module to be reference counting. Also prevent the module from beeing deleted using the interface.
Furthermore remove all static module creation and deletion functions.
Review URL: http://webrtc-codereview.appspot.com/133012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@580 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12 08:53:36 +00:00
henrika@google.com
73d65513f1 Adds reference counting to the ADM.
This CL modifies the ADM interface to ensure that an external ADM
can't call Create and Destroy any longer.

It also contains some minor style nits to conform better with
the Chromium style guide.
Review URL: http://webrtc-codereview.appspot.com/133014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
wu@webrtc.org
b15bfd32d7 * Add the time_stamp as one parameter to the ViE ExternalRenderer interface.
* Fix one issue in webrtcvideoengine where we should remove the renderer before adding a new one.
Review URL: http://webrtc-codereview.appspot.com/137011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@501 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-31 22:14:44 +00:00
perkj@google.com
4094c49ddf Temporarily use digital AGC in WebRTC since Chromium can't support analog AGC.
Fix suggested by henrika.
Review URL: http://webrtc-codereview.appspot.com/121001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@476 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-29 07:36:28 +00:00
mallinath@webrtc.org
6f555dcafe Review URL: http://webrtc-codereview.appspot.com/119002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@413 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-22 18:33:51 +00:00
wu@webrtc.org
eb29a9789d * Remove the previous renderer before set a new one.
* Allow to unregister a renderer by giving a NULL point.
Review URL: http://webrtc-codereview.appspot.com/123001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@412 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-22 15:58:03 +00:00
mallinath@webrtc.org
467b1a9e4a Review URL: http://webrtc-codereview.appspot.com/116007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@388 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-17 00:07:03 +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
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
ronghuawu@google.com
e8c5948b52 Revert back this change and wait when Tommi is only to submit the corresponding peerconnection test changes at the same time.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@32 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-01 17:14:19 +00:00
ronghuawu@google.com
7208ddddea Session layer update from p4 (cl37930)
Review URL: http://webrtc-codereview.appspot.com/29008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@30 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-01 17:00:36 +00:00
niklase@google.com
5c61233a88 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:41:01 +00:00