| commit | b295a3f5920e2706d42c960c5180c7cc6e1f435e | [log] [tgz] |
|---|---|---|
| author | dwkang@webrtc.org <dwkang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Thu Aug 29 07:34:12 2013 |
| committer | dwkang@webrtc.org <dwkang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Thu Aug 29 07:34:12 2013 |
| tree | e042e42d412168dca55f371889ea9dfa58fa4312 | |
| parent | d7301775f54f2e681bdb5efe33636352d0bbd805 [diff] |
Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync. Background: Since we had http://review.webrtc.org/2048004, the SSRC value in RtpRtcp for audio hasn't been updated. Because this prevents NTP update in RtpRtcp, the sync logic in ViESyncModule::Process() does not work. BUG=b/10484087 TEST= pass 'git try' except tests already broken in http://build.chromium.org/p/tryserver.webrtc/console R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2131004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4638 4adac7df-926f-26a2-2b94-8c16560cd09d