gecko/media/webrtc/trunk
[:ng] 575aa2d4f2 bug 1241064 - updating stats filter SSRC when audio channel SSRC changes; r=jib
bug: getStats was returning statistics for the shortlived, initial SSRC
now updating SSRC filter on statistics update callback to match audio channel ssrc
getStats API now returns statistics for correct SSRC: jitter, packets lost, etc.

MozReview-Commit-ID: WCd71WMkUW
2016-03-03 08:03:06 -08:00
..
build Bug 1198458: Rollup of changes previously applied to media/webrtc/trunk/webrtc and fixes to those rs=jesup r=froyd,jib,bwc,jesup,gcp,sotaro,pkerr,pehrsons 2015-11-18 15:03:25 -05:00
chromium_deps
google_apis/build
net
supplement
testing
third_party/opus
tools
webrtc bug 1241064 - updating stats filter SSRC when audio channel SSRC changes; r=jib 2016-03-03 08:03:06 -08:00
AUTHORS Bug 1198458: Webrtc updated to branch 43; pull made 2015-09-29 09:00AM PDT rs=jesup 2015-11-18 15:03:22 -05:00
DEPS Bug 1198458: Webrtc updated to branch 43; pull made 2015-09-29 09:00AM PDT rs=jesup 2015-11-18 15:03:22 -05:00
dummy_file.txt
Makefile.old
OWNERS Bug 1198458: Webrtc updated to branch 43; pull made 2015-09-29 09:00AM PDT rs=jesup 2015-11-18 15:03:22 -05:00
peerconnection_client.target.mk
peerconnection.gyp Bug 1198458: Rollup of changes previously applied to media/webrtc/trunk/webrtc and fixes to those rs=jesup r=froyd,jib,bwc,jesup,gcp,sotaro,pkerr,pehrsons 2015-11-18 15:03:25 -05:00
peerconnection.Makefile
README

This folder can be used to pull together the chromium version of webrtc
and libjingle, and build the peerconnection sample client and server. This will
check out a new repository in which you can build peerconnection_server.

Steps:
1) Create a new directory for the new repository (outside the webrtc repo):
   mkdir peerconnection
   cd peerconnection
2) gclient config --name trunk http://webrtc.googlecode.com/svn/trunk/peerconnection
3) gclient sync
4) cd trunk
5) make peerconnection_server peerconnection_client